diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-09-11 20:19:35 +0400 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-09-11 20:19:35 +0400 |
| commit | a9c86d42599519f3d83b5f46bdab25046fe47b84 (patch) | |
| tree | 9b269e3162e5cc0c1a8dfc3349303c902718a9a9 | |
| parent | a12e4d304ce701844c639541d90df86e165d03f9 (diff) | |
| parent | 1110afbe728838ac7ce973c37af9e11385dbaef9 (diff) | |
| download | linux-a9c86d42599519f3d83b5f46bdab25046fe47b84.tar.xz | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (377 commits)
ASoC: au1x: PSC-AC97 bugfixes
ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
ALSA: dummy - Add debug proc file
ALSA: Add const prefix to proc helper functions
ALSA: Re-export snd_pcm_format_name() function
ALSA: hda - Use auto model for HP laptops with ALC268 codec
ALSA: cs46xx - Fix minimum period size
ASoC: Fix WM835x Out4 capture enumeration
ALSA: Remove unneeded ifdef from sound/core.h
ALSA: Remove struct snd_monitor_file from public sound/core.h
ASoC: Remove unuused hw_read_t
sound: oxygen: work around MCE when changing volume
ALSA: dummy - Fake buffer allocations
ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
ASoC: fix pxa2xx-ac97.c breakage
ALSA: dummy - Fix the timer calculation in systimer mode
ALSA: dummy - Add more description
ALSA: dummy - Better jiffies handling
ALSA: dummy - Support high-res timer mode
ALSA: Release v1.0.21
...
221 files changed, 30709 insertions, 6969 deletions
diff --git a/Documentation/feature-removal-schedule.txt b/Documentation/feature-removal-schedule.txt index 09e031c55887..f0690bbbd73c 100644 --- a/Documentation/feature-removal-schedule.txt +++ b/Documentation/feature-removal-schedule.txt @@ -468,3 +468,27 @@ Why: cpu_policy_rwsem has a new cleaner definition making it local to cpufreq core and contained inside cpufreq.c. Other dependent drivers should not use it in order to safely avoid lockdep issues. Who: Venkatesh Pallipadi <venkatesh.pallipadi@intel.com> + +---------------------------- + +What: sound-slot/service-* module aliases and related clutters in + sound/sound_core.c +When: August 2010 +Why: OSS sound_core grabs all legacy minors (0-255) of SOUND_MAJOR + (14) and requests modules using custom sound-slot/service-* + module aliases. The only benefit of doing this is allowing + use of custom module aliases which might as well be considered + a bug at this point. This preemptive claiming prevents + alternative OSS implementations. + + Till the feature is removed, the kernel will be requesting + both sound-slot/service-* and the standard char-major-* module + aliases and allow turning off the pre-claiming selectively via + CONFIG_SOUND_OSS_CORE_PRECLAIM and soundcore.preclaim_oss + kernel parameter. + + After the transition phase is complete, both the custom module + aliases and switches to disable it will go away. This removal + will also allow making ALSA OSS emulation independent of + sound_core. The dependency will be broken then too. +Who: Tejun Heo <tj@kernel.org> diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4252697a95d6..1c8eb4518ce0 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -60,6 +60,12 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. slots - Reserve the slot index for the given driver. This option takes multiple strings. See "Module Autoloading Support" section for details. + debug - Specifies the debug message level + (0 = disable debug prints, 1 = normal debug messages, + 2 = verbose debug messages) + This option appears only when CONFIG_SND_DEBUG=y. + This option can be dynamically changed via sysfs + /sys/modules/snd/parameters/debug file. Module snd-pcm-oss ------------------ @@ -513,6 +519,26 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. or input, but you may use this module for any application which requires a sound card (like RealPlayer). + pcm_devs - Number of PCM devices assigned to each card + (default = 1, up to 4) + pcm_substreams - Number of PCM substreams assigned to each PCM + (default = 8, up to 16) + hrtimer - Use hrtimer (=1, default) or system timer (=0) + fake_buffer - Fake buffer allocations (default = 1) + + When multiple PCM devices are created, snd-dummy gives different + behavior to each PCM device: + 0 = interleaved with mmap support + 1 = non-interleaved with mmap support + 2 = interleaved without mmap + 3 = non-interleaved without mmap + + As default, snd-dummy drivers doesn't allocate the real buffers + but either ignores read/write or mmap a single dummy page to all + buffer pages, in order to save the resouces. If your apps need + the read/ written buffer data to be consistent, pass fake_buffer=0 + option. + The power-management is supported. Module snd-echo3g @@ -768,6 +794,10 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. bdl_pos_adj - Specifies the DMA IRQ timing delay in samples. Passing -1 will make the driver to choose the appropriate value based on the controller chip. + patch - Specifies the early "patch" files to modify the HD-audio + setup before initializing the codecs. This option is + available only when CONFIG_SND_HDA_PATCH_LOADER=y is set. + See HD-Audio.txt for details. [Single (global) options] single_cmd - Use single immediate commands to communicate with diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 939a3dd58148..97eebd63bedc 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -114,8 +114,8 @@ ALC662/663/272 samsung-nc10 Samsung NC10 mini notebook auto auto-config reading BIOS (default) -ALC882/885 -========== +ALC882/883/885/888/889 +====================== 3stack-dig 3-jack with SPDIF I/O 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 @@ -127,12 +127,8 @@ ALC882/885 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection w2jc ASUS W2JC - auto auto-config reading BIOS (default) - -ALC883/888 -========== - 3stack-dig 3-jack with SPDIF I/O - 6stack-dig 6-jack digital with SPDIF I/O + 3stack-2ch-dig 3-jack with SPDIF I/O (ALC883) + alc883-6stack-dig 6-jack digital with SPDIF I/O (ALC883) 3stack-6ch 3-jack 6-channel 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O 6stack-dig-demo 6-jack digital for Intel demo board @@ -140,6 +136,7 @@ ALC883/888 acer-aspire Acer Aspire 9810 acer-aspire-4930g Acer Aspire 4930G acer-aspire-6530g Acer Aspire 6530G + acer-aspire-7730g Acer Aspire 7730G acer-aspire-8930g Acer Aspire 8930G medion Medion Laptops medion-md2 Medion MD2 @@ -155,10 +152,13 @@ ALC883/888 3stack-hp HP machines with 3stack (Lucknow, Samba boards) 6stack-dell Dell machines with 6stack (Inspiron 530) mitac Mitac 8252D + clevo-m540r Clevo M540R (6ch + digital) clevo-m720 Clevo M720 laptop series fujitsu-pi2515 Fujitsu AMILO Pi2515 fujitsu-xa3530 Fujitsu AMILO XA3530 3stack-6ch-intel Intel DG33* boards + intel-alc889a Intel IbexPeak with ALC889A + intel-x58 Intel DX58 with ALC889 asus-p5q ASUS P5Q-EM boards mb31 MacBook 3,1 sony-vaio-tt Sony VAIO TT @@ -229,7 +229,7 @@ AD1984 ====== basic default configuration thinkpad Lenovo Thinkpad T61/X61 - dell Dell T3400 + dell_desktop Dell T3400 AD1986A ======= @@ -258,6 +258,7 @@ Conexant 5045 laptop-micsense Laptop with Mic sense (old model fujitsu) laptop-hpmicsense Laptop with HP and Mic senses benq Benq R55E + laptop-hp530 HP 530 laptop test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y @@ -278,9 +279,16 @@ Conexant 5051 hp-dv6736 HP dv6736 lenovo-x200 Lenovo X200 laptop +Conexant 5066 +============= + laptop Basic Laptop config (default) + dell-laptop Dell laptops + olpc-xo-1_5 OLPC XO 1.5 + STAC9200 ======== ref Reference board + oqo OQO Model 2 dell-d21 Dell (unknown) dell-d22 Dell (unknown) dell-d23 Dell (unknown) @@ -368,10 +376,12 @@ STAC92HD73* =========== ref Reference board no-jd BIOS setup but without jack-detection + intel Intel DG45* mobos dell-m6-amic Dell desktops/laptops with analog mics dell-m6-dmic Dell desktops/laptops with digital mics dell-m6 Dell desktops/laptops with both type of mics dell-eq Dell desktops/laptops + alienware Alienware M17x auto BIOS setup (default) STAC92HD83* @@ -385,3 +395,8 @@ STAC9872 ======== vaio VAIO laptop without SPDIF auto BIOS setup (default) + +Cirrus Logic CS4206/4207 +======================== + mbp55 MacBook Pro 5,5 + auto BIOS setup (default) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 71ac995b1915..7b8a5f947d1d 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -139,6 +139,10 @@ The driver checks PCI SSID and looks through the static configuration table until any matching entry is found. If you have a new machine, you may see a message like below: ------------------------------------------------------------------------ + hda_codec: ALC880: BIOS auto-probing. +------------------------------------------------------------------------ +Meanwhile, in the earlier versions, you would see a message like: +------------------------------------------------------------------------ hda_codec: Unknown model for ALC880, trying auto-probe from BIOS... ------------------------------------------------------------------------ Even if you see such a message, DON'T PANIC. Take a deep breath and @@ -403,6 +407,66 @@ re-configure based on that state, run like below: ------------------------------------------------------------------------ +Early Patching +~~~~~~~~~~~~~~ +When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a +firmware file for modifying the HD-audio setup before initializing the +codec. This can work basically like the reconfiguration via sysfs in +the above, but it does it before the first codec configuration. + +A patch file is a plain text file which looks like below: + +------------------------------------------------------------------------ + [codec] + 0x12345678 0xabcd1234 2 + + [model] + auto + + [pincfg] + 0x12 0x411111f0 + + [verb] + 0x20 0x500 0x03 + 0x20 0x400 0xff + + [hint] + hp_detect = yes +------------------------------------------------------------------------ + +The file needs to have a line `[codec]`. The next line should contain +three numbers indicating the codec vendor-id (0x12345678 in the +example), the codec subsystem-id (0xabcd1234) and the address (2) of +the codec. The rest patch entries are applied to this specified codec +until another codec entry is given. + +The `[model]` line allows to change the model name of the each codec. +In the example above, it will be changed to model=auto. +Note that this overrides the module option. + +After the `[pincfg]` line, the contents are parsed as the initial +default pin-configurations just like `user_pin_configs` sysfs above. +The values can be shown in user_pin_configs sysfs file, too. + +Similarly, the lines after `[verb]` are parsed as `init_verbs` +sysfs entries, and the lines after `[hint]` are parsed as `hints` +sysfs entries, respectively. + +The hd-audio driver reads the file via request_firmware(). Thus, +a patch file has to be located on the appropriate firmware path, +typically, /lib/firmware. For example, when you pass the option +`patch=hda-init.fw`, the file /lib/firmware/hda-init-fw must be +present. + +The patch module option is specific to each card instance, and you +need to give one file name for each instance, separated by commas. +For example, if you have two cards, one for an on-board analog and one +for an HDMI video board, you may pass patch option like below: +------------------------------------------------------------------------ + options snd-hda-intel patch=on-board-patch,hdmi-patch +------------------------------------------------------------------------ + + Power-Saving ~~~~~~~~~~~~ The power-saving is a kind of auto-suspend of the device. When the diff --git a/arch/arm/mach-omap2/mcbsp.c b/arch/arm/mach-omap2/mcbsp.c index 99b6e1546311..0447d26d454b 100644 --- a/arch/arm/mach-omap2/mcbsp.c +++ b/arch/arm/mach-omap2/mcbsp.c @@ -128,6 +128,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = { .rx_irq = INT_24XX_MCBSP1_IRQ_RX, .tx_irq = INT_24XX_MCBSP1_IRQ_TX, .ops = &omap2_mcbsp_ops, + .buffer_size = 0x6F, }, { .phys_base = OMAP34XX_MCBSP2_BASE, @@ -136,6 +137,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = { .rx_irq = INT_24XX_MCBSP2_IRQ_RX, .tx_irq = INT_24XX_MCBSP2_IRQ_TX, .ops = &omap2_mcbsp_ops, + .buffer_size = 0x3FF, }, { .phys_base = OMAP34XX_MCBSP3_BASE, @@ -144,6 +146,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = { .rx_irq = INT_24XX_MCBSP3_IRQ_RX, .tx_irq = INT_24XX_MCBSP3_IRQ_TX, .ops = &omap2_mcbsp_ops, + .buffer_size = 0x6F, }, { .phys_base = OMAP34XX_MCBSP4_BASE, @@ -152,6 +155,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = { .rx_irq = INT_24XX_MCBSP4_IRQ_RX, .tx_irq = INT_24XX_MCBSP4_IRQ_TX, .ops = &omap2_mcbsp_ops, + .buffer_size = 0x6F, }, { .phys_base = OMAP34XX_MCBSP5_BASE, @@ -160,6 +164,7 @@ static struct omap_mcbsp_platform_data omap34xx_mcbsp_pdata[] = { .rx_irq = INT_24XX_MCBSP5_IRQ_RX, .tx_irq = INT_24XX_MCBSP5_IRQ_TX, .ops = &omap2_mcbsp_ops, + .buffer_size = 0x6F, }, }; #define OMAP34XX_MCBSP_PDATA_SZ ARRAY_SIZE(omap34xx_mcbsp_pdata) diff --git a/arch/arm/mach-pxa/include/mach/audio.h b/arch/arm/mach-pxa/include/mach/audio.h index 16eb02552d5d..a3449e35a6f5 100644 --- a/arch/arm/mach-pxa/include/mach/audio.h +++ b/arch/arm/mach-pxa/include/mach/audio.h @@ -3,10 +3,12 @@ #include <sound/core.h> #include <sound/pcm.h> +#include <sound/ac97_codec.h> /* * @reset_gpio: AC97 reset gpio (normally gpio113 or gpio95) * a -1 value means no gpio will be used for reset + * @codec_pdata: AC97 codec platform_data * reset_gpio should only be specified for pxa27x CPUs where a silicon * bug prevents correct operation of the reset line. If not specified, @@ -20,6 +22,7 @@ typedef struct { void (*resume)(void *); void *priv; int reset_gpio; + void *codec_pdata[AC97_BUS_MAX_DEVICES]; } pxa2xx_audio_ops_t; extern void pxa_set_ac97_info(pxa2xx_audio_ops_t *ops); diff --git a/arch/arm/plat-omap/dma.c b/arch/arm/plat-omap/dma.c index e3ac94f09006..9b00f4cbc903 100644 --- a/arch/arm/plat-omap/dma.c +++ b/arch/arm/plat-omap/dma.c @@ -1127,6 +1127,11 @@ int omap_dma_running(void) void omap_dma_link_lch(int lch_head, int lch_queue) { if (omap_dma_in_1510_mode()) { + if (lch_head == lch_queue) { + dma_write(dma_read(CCR(lch_head)) | (3 << 8), + CCR(lch_head)); + return; + } printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; @@ -1149,6 +1154,11 @@ EXPORT_SYMBOL(omap_dma_link_lch); void omap_dma_unlink_lch(int lch_head, int lch_queue) { if (omap_dma_in_1510_mode()) { + if (lch_head == lch_queue) { + dma_write(dma_read(CCR(lch_head)) & ~(3 << 8), + CCR(lch_head)); + return; + } printk(KERN_ERR "DMA linking is not supported in 1510 mode\n"); BUG(); return; diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index bb154ea76769..63a3f254af7b 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -134,6 +134,11 @@ #define OMAP_MCBSP_REG_XCERG 0x74 #define OMAP_MCBSP_REG_XCERH 0x78 #define OMAP_MCBSP_REG_SYSCON 0x8C +#define OMAP_MCBSP_REG_THRSH2 0x90 +#define OMAP_MCBSP_REG_THRSH1 0x94 +#define OMAP_MCBSP_REG_IRQST 0xA0 +#define OMAP_MCBSP_REG_IRQEN 0xA4 +#define OMAP_MCBSP_REG_WAKEUPEN 0xA8 #define OMAP_MCBSP_REG_XCCR 0xAC #define OMAP_MCBSP_REG_RCCR 0xB0 @@ -249,8 +254,27 @@ #define RDISABLE 0x0001 /********************** McBSP SYSCONFIG bit definitions ********************/ +#define CLOCKACTIVITY(value) ((value)<<8) +#define SIDLEMODE(value) ((value)<<3) +#define ENAWAKEUP 0x0004 #define SOFTRST 0x0002 +/********************** McBSP DMA operating modes **************************/ +#define MCBSP_DMA_MODE_ELEMENT 0 +#define MCBSP_DMA_MODE_THRESHOLD 1 +#define MCBSP_DMA_MODE_FRAME 2 + +/********************** McBSP WAKEUPEN bit definitions *********************/ +#define XEMPTYEOFEN 0x4000 +#define XRDYEN 0x0400 +#define XEOFEN 0x0200 +#define XFSXEN 0x0100 +#define XSYNCERREN 0x0080 +#define RRDYEN 0x0008 +#define REOFEN 0x0004 +#define RFSREN 0x0002 +#define RSYNCERREN 0x0001 + /* we don't do multichannel for now */ struct omap_mcbsp_reg_cfg { u16 spcr2; @@ -344,6 +368,9 @@ struct omap_mcbsp_platform_data { u8 dma_rx_sync, dma_tx_sync; u16 rx_irq, tx_irq; struct omap_mcbsp_ops *ops; +#ifdef CONFIG_ARCH_OMAP34XX + u16 buffer_size; +#endif }; struct omap_mcbsp { @@ -377,6 +404,11 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct clk *iclk; struct clk *fclk; +#ifdef CONFIG_ARCH_OMAP34XX + int dma_op_mode; + u16 max_tx_thres; + u16 max_rx_thres; +#endif }; extern struct omap_mcbsp **mcbsp_ptr; extern int omap_mcbsp_count; @@ -385,10 +417,25 @@ int omap_mcbsp_init(void); void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, int size); void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config); +#ifdef CONFIG_ARCH_OMAP34XX +void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold); +void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold); +u16 omap_mcbsp_get_max_tx_threshold(unsigned int id); +u16 omap_mcbsp_get_max_rx_threshold(unsigned int id); +int omap_mcbsp_get_dma_op_mode(unsigned int id); +#else +static inline void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold) +{ } +static inline void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold) +{ } +static inline u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) { return 0; } +static inline u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) { return 0; } +static inline int omap_mcbsp_get_dma_op_mode(unsigned int id) { return 0; } +#endif int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); -void omap_mcbsp_start(unsigned int id); -void omap_mcbsp_stop(unsigned int id); +void omap_mcbsp_start(unsigned int id, int tx, int rx); +void omap_mcbsp_stop(unsigned int id, int tx, int rx); void omap_mcbsp_xmit_word(unsigned int id, u32 word); u32 omap_mcbsp_recv_word(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index efa0e0111f38..8dc7927906f1 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -198,6 +198,170 @@ void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) } EXPORT_SYMBOL(omap_mcbsp_config); +#ifdef CONFIG_ARCH_OMAP34XX +/* + * omap_mcbsp_set_tx_threshold configures how to deal + * with transmit threshold. the threshold value and handler can be + * configure in here. + */ +void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold) +{ + struct omap_mcbsp *mcbsp; + void __iomem *io_base; + + if (!cpu_is_omap34xx()) + return; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + io_base = mcbsp->io_base; + + OMAP_MCBSP_WRITE(io_base, THRSH2, threshold); +} +EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold); + +/* + * omap_mcbsp_set_rx_threshold configures how to deal + * with receive threshold. the threshold value and handler can be + * configure in here. + */ +void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold) +{ + struct omap_mcbsp *mcbsp; + void __iomem *io_base; + + if (!cpu_is_omap34xx()) + return; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + io_base = mcbsp->io_base; + + OMAP_MCBSP_WRITE(io_base, THRSH1, threshold); +} +EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold); + +/* + * omap_mcbsp_get_max_tx_thres just return the current configured + * maximum threshold for transmission + */ +u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->max_tx_thres; +} +EXPORT_SYMBOL(omap_mcbsp_get_max_tx_threshold); + +/* + * omap_mcbsp_get_max_rx_thres just return the current configured + * maximum threshold for reception + */ +u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->max_rx_thres; +} +EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold); + +/* + * omap_mcbsp_get_dma_op_mode just return the current configured + * operating mode for the mcbsp channel + */ +int omap_mcbsp_get_dma_op_mode(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + int dma_op_mode; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%u)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + spin_lock_irq(&mcbsp->lock); + dma_op_mode = mcbsp->dma_op_mode; + spin_unlock_irq(&mcbsp->lock); + + return dma_op_mode; +} +EXPORT_SYMBOL(omap_mcbsp_get_dma_op_mode); + +static inline void omap34xx_mcbsp_request(struct omap_mcbsp *mcbsp) +{ + /* + * Enable wakup behavior, smart idle and all wakeups + * REVISIT: some wakeups may be unnecessary + */ + if (cpu_is_omap34xx()) { + u16 syscon; + + syscon = OMAP_MCBSP_READ(mcbsp->io_base, SYSCON); + syscon &= ~(ENAWAKEUP | SIDLEMODE(0x03) | CLOCKACTIVITY(0x03)); + + spin_lock_irq(&mcbsp->lock); + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { + syscon |= (ENAWAKEUP | SIDLEMODE(0x02) | + CLOCKACTIVITY(0x02)); + OMAP_MCBSP_WRITE(mcbsp->io_base, WAKEUPEN, + XRDYEN | RRDYEN); + } else { + syscon |= SIDLEMODE(0x01); + } + spin_unlock_irq(&mcbsp->lock); + + OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon); + } +} + +static inline void omap34xx_mcbsp_free(struct omap_mcbsp *mcbsp) +{ + /* + * Disable wakup behavior, smart idle and all wakeups + */ + if (cpu_is_omap34xx()) { + u16 syscon; + + syscon = OMAP_MCBSP_READ(mcbsp->io_base, SYSCON); + syscon &= ~(ENAWAKEUP | SIDLEMODE(0x03) | CLOCKACTIVITY(0x03)); + /* + * HW bug workaround - If no_idle mode is taken, we need to + * go to smart_idle before going to always_idle, or the + * device will not hit retention anymore. + */ + syscon |= SIDLEMODE(0x02); + OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon); + + syscon &= ~(SIDLEMODE(0x03)); + OMAP_MCBSP_WRITE(mcbsp->io_base, SYSCON, syscon); + + OMAP_MCBSP_WRITE(mcbsp->io_base, WAKEUPEN, 0); + } +} +#else +static inline void omap34xx_mcbsp_request(struct omap_mcbsp *mcbsp) {} +static inline void omap34xx_mcbsp_free(struct omap_mcbsp *mcbsp) {} +#endif + /* * We can choose between IRQ based or polled IO. * This needs to be called before omap_mcbsp_request(). @@ -257,6 +421,9 @@ int omap_mcbsp_request(unsigned int id) clk_enable(mcbsp->iclk); clk_enable(mcbsp->fclk); + /* Do procedure specific to omap34xx arch, if applicable */ + omap34xx_mcbsp_request(mcbsp); + /* * Make sure that transmitter, receiver and sample-rate generator are * not running before activating IRQs. @@ -305,6 +472,9 @@ void omap_mcbsp_free(unsigned int id) if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(id); + /* Do procedure specific to omap34xx arch, if applicable */ + omap34xx_mcbsp_free(mcbsp); + clk_disable(mcbsp->fclk); clk_disable(mcbsp->iclk); @@ -328,14 +498,15 @@ void omap_mcbsp_free(unsigned int id) EXPORT_SYMBOL(omap_mcbsp_free); /* - * Here we start the McBSP, by enabling the sample - * generator, both transmitter and receivers, - * and the frame sync. + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. */ -void omap_mcbsp_start(unsigned int id) +void omap_mcbsp_start(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -348,32 +519,58 @@ void omap_mcbsp_start(unsigned int id) mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7; mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7; - /* Start the sample generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Start the sample generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + } /* Enable transmitter and receiver */ + tx &= 1; w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | tx); + rx &= 1; w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR1, w | rx); - udelay(100); + /* + * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec + * REVISIT: 100us may give enough time for two CLKSRG, however + * due to some unknown PM related, clock gating etc. reason it + * is now at 500us. + */ + udelay(500); - /* Start frame sync */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + if (idle) { + /* Start frame sync */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + } + + if (cpu_is_omap2430() || cpu_is_omap34xx()) { + /* Release the transmitter and receiver */ + w = OMAP_MCBSP_READ(io_base, XCCR); + w &= ~(tx ? XDISABLE : 0); + OMAP_MCBSP_WRITE(io_base, XCCR, w); + w = OMAP_MCBSP_READ(io_base, RCCR); + w &= ~(rx ? RDISABLE : 0); + OMAP_MCBSP_WRITE(io_base, RCCR, w); + } /* Dump McBSP Regs */ omap_mcbsp_dump_reg(id); } EXPORT_SYMBOL(omap_mcbsp_start); -void omap_mcbsp_stop(unsigned int id) +void omap_mcbsp_stop(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -385,16 +582,33 @@ void omap_mcbsp_stop(unsigned int id) io_base = mcbsp->io_base; /* Reset transmitter */ + tx &= 1; + if (cpu_is_omap2430() || cpu_is_omap34xx()) { + w = OMAP_MCBSP_READ(io_base, XCCR); + w |= (tx ? XDISABLE : 0); + OMAP_MCBSP_WRITE(io_base, XCCR, w); + } w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~tx); /* Reset receiver */ + rx &= 1; + if (cpu_is_omap2430() || cpu_is_omap34xx()) { + w = OMAP_MCBSP_READ(io_base, RCCR); + w |= (tx ? RDISABLE : 0); + OMAP_MCBSP_WRITE(io_base, RCCR, w); + } w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~rx); - /* Reset the sample rate generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + } } EXPORT_SYMBOL(omap_mcbsp_stop); @@ -883,6 +1097,149 @@ void omap_mcbsp_set_spi_mode(unsigned int id, } EXPORT_SYMBOL(omap_mcbsp_set_spi_mode); +#ifdef CONFIG_ARCH_OMAP34XX +#define max_thres(m) (mcbsp->pdata->buffer_size) +#define valid_threshold(m, val) ((val) <= max_thres(m)) +#define THRESHOLD_PROP_BUILDER(prop) \ +static ssize_t prop##_show(struct device *dev, \ + struct device_attribute *attr, char *buf) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + \ + return sprintf(buf, "%u\n", mcbsp->prop); \ +} \ + \ +static ssize_t prop##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t size) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + unsigned long val; \ + int status; \ + \ + status = strict_strtoul(buf, 0, &val); \ + if (status) \ + return status; \ + \ + if (!valid_threshold(mcbsp, val)) \ + return -EDOM; \ + \ + mcbsp->prop = val; \ + return size; \ +} \ + \ +static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store); + +THRESHOLD_PROP_BUILDER(max_tx_thres); +THRESHOLD_PROP_BUILDER(max_rx_thres); + +static const char *dma_op_modes[] = { + "element", "threshold", "frame", +}; + +static ssize_t dma_op_mode_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int dma_op_mode, i = 0; + ssize_t len = 0; + const char * const *s; + + spin_lock_irq(&mcbsp->lock); + dma_op_mode = mcbsp->dma_op_mode; + spin_unlock_irq(&mcbsp->lock); + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { + if (dma_op_mode == i) + len += sprintf(buf + len, "[%s] ", *s); + else + len += sprintf(buf + len, "%s ", *s); + } + len += sprintf(buf + len, "\n"); + + return len; +} + +static ssize_t dma_op_mode_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + const char * const *s; + int i = 0; + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) + if (sysfs_streq(buf, *s)) + break; + + if (i == ARRAY_SIZE(dma_op_modes)) + return -EINVAL; + + spin_lock_irq(&mcbsp->lock); + if (!mcbsp->free) { + size = -EBUSY; + goto unlock; + } + mcbsp->dma_op_mode = i; + +unlock: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR(dma_op_mode, 0644, dma_op_mode_show, dma_op_mode_store); + +static const struct attribute *additional_attrs[] = { + &dev_attr_max_tx_thres.attr, + &dev_attr_max_rx_thres.attr, + &dev_attr_dma_op_mode.attr, + NULL, +}; + +static const struct attribute_group additional_attr_group = { + .attrs = (struct attribute **)additional_attrs, +}; + +static inline int __devinit omap_additional_add(struct device *dev) +{ + return sysfs_create_group(&dev->kobj, &additional_attr_group); +} + +static inline void __devexit omap_additional_remove(struct device *dev) +{ + sysfs_remove_group(&dev->kobj, &additional_attr_group); +} + +static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp) +{ + mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; + if (cpu_is_omap34xx()) { + mcbsp->max_tx_thres = max_thres(mcbsp); + mcbsp->max_rx_thres = max_thres(mcbsp); + /* + * REVISIT: Set dmap_op_mode to THRESHOLD as default + * for mcbsp2 instances. + */ + if (omap_additional_add(mcbsp->dev)) + dev_warn(mcbsp->dev, + "Unable to create additional controls\n"); + } else { + mcbsp->max_tx_thres = -EINVAL; + mcbsp->max_rx_thres = -EINVAL; + } +} + +static inline void __devexit omap34xx_device_exit(struct omap_mcbsp *mcbsp) +{ + if (cpu_is_omap34xx()) + omap_additional_remove(mcbsp->dev); +} +#else +static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp) {} +static inline void __devexit omap34xx_device_exit(struct omap_mcbsp *mcbsp) {} +#endif /* CONFIG_ARCH_OMAP34XX */ + /* * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. * 730 has only 2 McBSP, and both of them are MPU peripherals. @@ -953,6 +1310,10 @@ static int __devinit omap_mcbsp_probe(struct platform_device *pdev) mcbsp->dev = &pdev->dev; mcbsp_ptr[id] = mcbsp; platform_set_drvdata(pdev, mcbsp); + + /* Initialize mcbsp properties for OMAP34XX if needed / applicable */ + omap34xx_device_init(mcbsp); + return 0; err_fclk: @@ -976,6 +1337,8 @@ static int __devexit omap_mcbsp_remove(struct platform_device *pdev) mcbsp->pdata->ops->free) mcbsp->pdata->ops->free(mcbsp->id); + omap34xx_device_exit(mcbsp); + clk_disable(mcbsp->fclk); clk_disable(mcbsp->iclk); clk_put(mcbsp->fclk); diff --git a/arch/arm/plat-s3c/include/plat/audio-simtec.h b/arch/arm/plat-s3c/include/plat/audio-simtec.h new file mode 100644 index 000000000000..0f440b9168db --- /dev/null +++ b/arch/arm/plat-s3c/include/plat/audio-simtec.h @@ -0,0 +1,37 @@ +/* arch/arm/plat-s3c/include/plat/audio-simtec.h + * + * Copyright 2008 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben@simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simtec Audio support. +*/ + +/** + * struct s3c24xx_audio_simtec_pdata - platform data for simtec audio + * @use_mpllin: Select codec clock from MPLLin + * @output_cdclk: Need to output CDCLK to the codec + * @have_mic: Set if we have a MIC socket + * @have_lout: Set if we have a LineOut socket + * @amp_gpio: GPIO pin to enable the AMP + * @amp_gain: Option GPIO to control AMP gain + */ +struct s3c24xx_audio_simtec_pdata { + unsigned int use_mpllin:1; + unsigned int output_cdclk:1; + + unsigned int have_mic:1; + unsigned int have_lout:1; + + int amp_gpio; + int amp_gain[2]; + + void (*startup)(void); +}; + +extern int simtec_audio_add(const char *codec_name, + struct s3c24xx_audio_simtec_pdata *pdata); diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index 0fad7571030e..07659dad1748 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -33,6 +33,11 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + #define S3C64XX_IISMOD_IMS_PCLK (0 << 10) #define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) diff --git a/fs/char_dev.c b/fs/char_dev.c index 7c27a8ebef6a..3cbc57f932d2 100644 --- a/fs/char_dev.c +++ b/fs/char_dev.c @@ -238,8 +238,10 @@ int alloc_chrdev_region(dev_t *dev, unsigned baseminor, unsigned count, } /** - * register_chrdev() - Register a major number for character devices. + * __register_chrdev() - create and register a cdev occupying a range of minors * @major: major device number or 0 for dynamic allocation + * @baseminor: first of the requested range of minor numbers + * @count: the number of minor numbers required * @name: name of this range of devices * @fops: file operations associated with this devices * @@ -255,19 +257,17 @@ int alloc_chrdev_region(dev_t *dev, unsigned baseminor, unsigned count, * /dev. It only helps to keep track of the different owners of devices. If * your module name has only one type of devices it's ok to use e.g. the name * of the module here. - * - * This function registers a range of 256 minor numbers. The first minor number - * is 0. */ -int register_chrdev(unsigned int major, const char *name, - const struct file_operations *fops) +int __register_chrdev(unsigned int major, unsigned int baseminor, + unsigned int count, const char *name, + const struct file_operations *fops) { struct char_device_struct *cd; struct cdev *cdev; char *s; int err = -ENOMEM; - cd = __register_chrdev_region(major, 0, 256, name); + cd = __register_chrdev_region(major, baseminor, count, name); if (IS_ERR(cd)) return PTR_ERR(cd); @@ -281,7 +281,7 @@ int register_chrdev(unsigned int major, const char *name, for (s = strchr(kobject_name(&cdev->kobj),'/'); s; s = strchr(s, '/')) *s = '!'; - err = cdev_add(cdev, MKDEV(cd->major, 0), 256); + err = cdev_add(cdev, MKDEV(cd->major, baseminor), count); if (err) goto out; @@ -291,7 +291,7 @@ int register_chrdev(unsigned int major, const char *name, out: kobject_put(&cdev->kobj); out2: - kfree(__unregister_chrdev_region(cd->major, 0, 256)); + kfree(__unregister_chrdev_region(cd->major, baseminor, count)); return err; } @@ -317,10 +317,23 @@ void unregister_chrdev_region(dev_t from, unsigned count) } } -void unregister_chrdev(unsigned int major, const char *name) +/** + * __unregister_chrdev - unregister and destroy a cdev + * @major: major device number + * @baseminor: first of the range of minor numbers + * @count: the number of minor numbers this cdev is occupying + * @name: name of this range of devices + * + * Unregister and destroy the cdev occupying the region described by + * @major, @baseminor and @count. This function undoes what + * __register_chrdev() did. + */ +void __unregister_chrdev(unsigned int major, unsigned int baseminor, + unsigned int count, const char *name) { struct char_device_struct *cd; - cd = __unregister_chrdev_region(major, 0, 256); + + cd = __unregister_chrdev_region(major, baseminor, count); if (cd && cd->cdev) cdev_del(cd->cdev); kfree(cd); @@ -569,6 +582,6 @@ EXPORT_SYMBOL(cdev_alloc); EXPORT_SYMBOL(cdev_del); EXPORT_SYMBOL(cdev_add); EXPORT_SYMBOL(cdev_index); -EXPORT_SYMBOL(register_chrdev); -EXPORT_SYMBOL(unregister_chrdev); +EXPORT_SYMBOL(__register_chrdev); +EXPORT_SYMBOL(__unregister_chrdev); EXPORT_SYMBOL(directly_mappable_cdev_bdi); diff --git a/include/linux/fs.h b/include/linux/fs.h index 26da98f61116..a79f48373e7e 100644 --- a/include/linux/fs.h +++ b/include/linux/fs.h @@ -1997,12 +1997,25 @@ extern void bd_release_from_disk(struct block_device *, struct gendisk *); #define CHRDEV_MAJOR_HASH_SIZE 255 extern int alloc_chrdev_region(dev_t *, unsigned, unsigned, const char *); extern int register_chrdev_region(dev_t, unsigned, const char *); -extern int register_chrdev(unsigned int, const char *, - const struct file_operations *); -extern void unregister_chrdev(unsigned int, const char *); +extern int __register_chrdev(unsigned int major, unsigned int baseminor, + unsigned int count, const char *name, + const struct file_operations *fops); +extern void __unregister_chrdev(unsigned int major, unsigned int baseminor, + unsigned int count, const char *name); extern void unregister_chrdev_region(dev_t, unsigned); extern void chrdev_show(struct seq_file *,off_t); +static inline int register_chrdev(unsigned int major, const char *name, + const struct file_operations *fops) +{ + return __register_chrdev(major, 0, 256, name, fops); +} + +static inline void unregister_chrdev(unsigned int major, const char *name) +{ + __unregister_chrdev(major, 0, 256, name); +} + /* fs/block_dev.c */ #define BDEVNAME_SIZE 32 /* Largest string for a blockdev identifier */ #define BDEVT_SIZE 10 /* Largest string for MAJ:MIN for blkdev */ diff --git a/include/linux/tty.h b/include/linux/tty.h index e8c6c9136c97..0d3974f59c53 100644 --- a/include/linux/tty.h +++ b/include/linux/tty.h @@ -23,7 +23,7 @@ */ #define NR_UNIX98_PTY_DEFAULT 4096 /* Default maximum for Unix98 ptys */ #define NR_UNIX98_PTY_MAX (1 << MINORBITS) /* Absolute limit */ -#define NR_LDISCS 19 +#define NR_LDISCS 20 /* line disciplines */ #define N_TTY 0 @@ -47,6 +47,8 @@ #define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */ #define N_PPS 18 /* Pulse per Second */ +#define N_V253 19 /* Codec control over voice modem */ + /* * This character is the same as _POSIX_VDISABLE: it cannot be used as * a c_cc[] character, but indicates that a particular special character diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 251fc1cd5002..3dae3f799b9b 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -32,6 +32,9 @@ #include "control.h" #include "info.h" +/* maximum number of devices on the AC97 bus */ +#define AC97_BUS_MAX_DEVICES 4 + /* * AC'97 codec registers */ @@ -642,4 +645,10 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime); /* ad hoc AC97 device driver access */ extern struct bus_type ac97_bus_type; +/* AC97 platform_data adding function */ +static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data) +{ + ac97->dev.platform_data = data; +} + #endif /* __SOUND_AC97_CODEC_H */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 82aed3f47534..1f57bb92eb5a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -138,7 +138,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 9) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; diff --git a/include/sound/core.h b/include/sound/core.h index 309cb9659a05..a61499c22b0b 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -93,15 +93,6 @@ struct snd_device { #define snd_device(n) list_entry(n, struct snd_device, list) -/* monitor files for graceful shutdown (hotplug) */ - -struct snd_monitor_file { - struct file *file; - const struct file_operations *disconnected_f_op; - struct list_head shutdown_list; /* still need to shutdown */ - struct list_head list; /* link of monitor files */ -}; - /* main structure for soundcard */ struct snd_card { @@ -311,9 +302,7 @@ int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); -#ifndef snd_card_set_dev #define snd_card_set_dev(card, devptr) ((card)->dev = (devptr)) -#endif /* device.c */ @@ -340,18 +329,17 @@ unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); struct resource; void release_and_free_resource(struct resource *res); -#ifdef CONFIG_SND_VERBOSE_PRINTK -void snd_verbose_printk(const char *file, int line, const char *format, ...) - __attribute__ ((format (printf, 3, 4))); -#endif -#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK) -void snd_verbose_printd(const char *file, int line, const char *format, ...) - __attribute__ ((format (printf, 3, 4))); -#endif - /* --- */ -#ifdef CONFIG_SND_VERBOSE_PRINTK +#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +void __snd_printk(unsigned int level, const char *file, int line, + const char *format, ...) + __attribute__ ((format (printf, 4, 5))); +#else +#define __snd_printk(level, file, line, format, args...) \ + printk(format, ##args) +#endif + /** * snd_printk - printk wrapper * @fmt: format string @@ -360,15 +348,9 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * when configured with CONFIG_SND_VERBOSE_PRINTK. */ #define snd_printk(fmt, args...) \ - snd_verbose_printk(__FILE__, __LINE__, fmt ,##args) -#else -#define snd_printk(fmt, args...) \ - printk(fmt ,##args) -#endif + __snd_printk(0, __FILE__, __LINE__, fmt, ##args) #ifdef CONFIG_SND_DEBUG - -#ifdef CONFIG_SND_VERBOSE_PRINTK /** * snd_printd - debug printk * @fmt: format string @@ -377,11 +359,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * Ignored when CONFIG_SND_DEBUG is not set. */ #define snd_printd(fmt, args...) \ - snd_verbose_printd(__FILE__, __LINE__, fmt ,##args) -#else -#define snd_printd(fmt, args...) \ - printk(fmt ,##args) -#endif + __snd_printk(1, __FILE__, __LINE__, fmt, ##args) /** * snd_BUG - give a BUG warning message and stack trace @@ -428,9 +406,10 @@ static inline int __snd_bug_on(int cond) * Works like snd_printk() for debugging purposes. * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. */ -#define snd_printdd(format, args...) snd_printk(format, ##args) +#define snd_printdd(format, args...) \ + __snd_printk(2, __FILE__, __LINE__, format, ##args) #else -#define snd_printdd(format, args...) /* nothing */ +#define snd_printdd(format, args...) do { } while (0) #endif @@ -438,12 +417,10 @@ static inline int __snd_bug_on(int cond) /* for easier backward-porting */ #if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) -#ifndef gameport_set_dev_parent #define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev)) #define gameport_set_port_data(gp,r) ((gp)->port_data = (r)) #define gameport_get_port_data(gp) (gp)->port_data #endif -#endif /* PCI quirk list helper */ struct snd_pci_quirk { diff --git a/include/sound/info.h b/include/sound/info.h index 7c2ee1a21b00..112e8949e1a7 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -110,13 +110,13 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer); static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {} #endif -int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \ +int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) \ __attribute__ ((format (printf, 2, 3))); int snd_info_init(void); int snd_info_done(void); int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len); -char *snd_info_get_str(char *dest, char *src, int len); +const char *snd_info_get_str(char *dest, const char *src, int len); struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent); diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 7ccce94a5255..c42506212649 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -47,7 +47,11 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */ #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ +#ifdef CONFIG_SND_DMA_SGBUF #define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#endif /* * info for buffer allocation @@ -60,6 +64,7 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +#ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages */ @@ -107,6 +112,7 @@ static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset) { return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE; } +#endif /* CONFIG_SND_DMA_SGBUF */ /* allocate/release a buffer */ int snd_dma_alloc_pages(int type, struct device *dev, size_t size, diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 23893523dc8c..de6d981de5d6 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -902,6 +902,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +#ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling */ @@ -927,6 +928,28 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream, unsigned int ofs, unsigned int size); +#else /* !SND_DMA_SGBUF */ +/* + * fake using a continuous buffer + */ +static inline dma_addr_t +snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs) +{ + return substream->runtime->dma_addr + ofs; +} + +static inline void * +snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs) +{ + return substream->runtime->dma_area + ofs; +} + +#define snd_pcm_sgbuf_ops_page NULL + +#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size) (size) + +#endif /* SND_DMA_SGBUF */ + /* handle mmap counter - PCM mmap callback should handle this counter properly */ static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area) { @@ -965,4 +988,6 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) #define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime) +const char *snd_pcm_format_name(snd_pcm_format_t format); + #endif /* __SOUND_PCM_H */ diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h new file mode 100644 index 000000000000..c0227361a876 --- /dev/null +++ b/include/sound/sh_fsi.h @@ -0,0 +1,83 @@ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori@renesas.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +/* flags format + + * 0xABCDEEFF + * + * A: channel size for TDM (input) + * B: channel size for TDM (ooutput) + * C: inversion + * D: mode + * E: input format + * F: output format + */ + +#include <linux/clk.h> +#include <sound/soc.h> + +/* TDM channel */ +#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28) +#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24) + +#define SH_FSI_CH_IMASK 0xF0000000 +#define SH_FSI_CH_OMASK 0x0F000000 +#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28) +#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24) + +/* clock inversion */ +#define SH_FSI_INVERSION_MASK 0x00F00000 +#define SH_FSI_LRM_INV (1 << 20) +#define SH_FSI_BRM_INV (1 << 21) +#define SH_FSI_LRS_INV (1 << 22) +#define SH_FSI_BRS_INV (1 << 23) + +/* mode */ +#define SH_FSI_MODE_MASK 0x000F0000 +#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */ +#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */ + +/* DI format */ +#define SH_FSI_FMT_MASK 0x000000FF +#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8) +#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0) +#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK) +#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK) + +#define SH_FSI_FMT_MONO (1 << 0) +#define SH_FSI_FMT_MONO_DELAY (1 << 1) +#define SH_FSI_FMT_PCM (1 << 2) +#define SH_FSI_FMT_I2S (1 << 3) +#define SH_FSI_FMT_TDM (1 << 4) +#define SH_FSI_FMT_TDM_DELAY (1 << 5) + +#define SH_FSI_IFMT_TDM_CH(x) \ + (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x)) +#define SH_FSI_IFMT_TDM_DELAY_CH(x) \ + (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x)) + +#define SH_FSI_OFMT_TDM_CH(x) \ + (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x)) +#define SH_FSI_OFMT_TDM_DELAY_CH(x) \ + (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x)) + +struct sh_fsi_platform_info { + unsigned long porta_flags; + unsigned long portb_flags; +}; + +extern struct snd_soc_dai fsi_soc_dai[2]; +extern struct snd_soc_platform fsi_soc_platform; + +#endif /* __SOUND_FSI_H */ diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7eee9b6d..97ca9af414dc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -27,8 +27,8 @@ struct snd_pcm_substream; #define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ #define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ #define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ +#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ #define SND_SOC_DAIFMT_AC97 5 /* AC97 */ /* left and right justified also known as MSB and LSB respectively */ @@ -38,7 +38,7 @@ struct snd_pcm_substream; /* * DAI Clock gating. * - * DAI bit clocks can be be gated (disabled) when not the DAI is not + * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ @@ -51,21 +51,21 @@ struct snd_pcm_substream; * format. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ +#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ +#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ +#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ /* * DAI hardware clock masters. * * This is wrt the codec, the inverse is true for the interface - * i.e. if the codec is clk and frm master then the interface is + * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ +#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ +#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ #define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ +#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 @@ -78,7 +78,13 @@ struct snd_pcm_substream; #define SND_SOC_CLOCK_IN 0 #define SND_SOC_CLOCK_OUT 1 -#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ +#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE |\ + SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) @@ -106,7 +112,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -116,12 +122,12 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); /* * Digital Audio Interface. * - * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 - * operations an capabilities. Codec and platfom drivers will register a this + * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 + * operations and capabilities. Codec and platform drivers will register this * structure for every DAI they have. * * This structure covers the clocking, formating and ALSA operations for each - * interface a + * interface. */ struct snd_soc_dai_ops { /* @@ -140,7 +146,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* @@ -179,6 +186,7 @@ struct snd_soc_dai { int ac97_control; struct device *dev; + void *ac97_pdata; /* platform_data for the ac97 codec */ /* DAI callbacks */ int (*probe)(struct platform_device *pdev, diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ec8a45f9a069..c1410e3191e3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -137,6 +137,12 @@ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} /* stream domain */ +#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} @@ -279,9 +285,11 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, /* dapm events */ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); +void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec); /* dapm audio pin control and status */ int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin); @@ -311,6 +319,8 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_aif_in, /* audio interface input */ + snd_soc_dapm_aif_out, /* audio interface output */ }; /* diff --git a/include/sound/soc.h b/include/sound/soc.h index cf6111d72b17..475cb7ed6bec 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -135,6 +135,28 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) } +#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right, \ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_bool_ext, \ @@ -183,14 +205,28 @@ struct snd_soc_jack_gpio; #endif typedef int (*hw_write_t)(void *,const char* ,int); -typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +enum snd_soc_control_type { + SND_SOC_CUSTOM, + SND_SOC_I2C, + SND_SOC_SPI, +}; + int snd_soc_register_platform(struct snd_soc_platform *platform); void snd_soc_unregister_platform(struct snd_soc_platform *platform); int snd_soc_register_codec(struct snd_soc_codec *codec); void snd_soc_unregister_codec(struct snd_soc_codec *codec); +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control); + +#ifdef CONFIG_PM +int snd_soc_suspend_device(struct device *dev); +int snd_soc_resume_device(struct device *dev); +#endif /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); @@ -216,9 +252,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, /* codec register bit access */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value); + unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); @@ -356,8 +392,10 @@ struct snd_soc_codec { int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); int (*display_register)(struct snd_soc_codec *, char *, size_t, unsigned int); + int (*volatile_register)(unsigned int); + int (*readable_register)(unsigned int); hw_write_t hw_write; - hw_read_t hw_read; + unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); void *reg_cache; short reg_cache_size; short reg_cache_step; @@ -369,8 +407,6 @@ struct snd_soc_codec { enum snd_soc_bias_level bias_level; enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - struct list_head up_list; - struct list_head down_list; /* codec DAI's */ struct snd_soc_dai *dai; @@ -379,6 +415,7 @@ struct snd_soc_codec { #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_reg; struct dentry *debugfs_pop_time; + struct dentry *debugfs_dapm; #endif }; diff --git a/include/sound/tlv.h b/include/sound/tlv.h index d136ea2181ed..9fd5b19ccf5c 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -35,6 +35,8 @@ #define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */ #define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */ #define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */ +#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ +#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ @@ -42,6 +44,18 @@ #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } +/* dB scale specified with min/max values instead of step */ +#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \ + SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \ + (min_dB), (max_dB) +#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) } +#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \ + unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) } + /* linear volume between min_dB and max_dB (.01dB unit) */ #define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \ SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \ diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h new file mode 100644 index 000000000000..381319c7000c --- /dev/null +++ b/include/sound/uda1380.h @@ -0,0 +1,22 @@ +/* + * UDA1380 ALSA SoC Codec driver + * + * Copyright 2009 Philipp Zabel + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __UDA1380_H +#define __UDA1380_H + +struct uda1380_platform_data { + int gpio_power; + int gpio_reset; + int dac_clk; +#define UDA1380_DAC_CLK_SYSCLK 0 +#define UDA1380_DAC_CLK_WSPLL 1 +}; + +#endif /* __UDA1380_H */ diff --git a/include/sound/version.h b/include/sound/version.h index 456f1359e1c0..22939142dd23 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.20" +#define CONFIG_SND_VERSION "1.0.21" #define CONFIG_SND_DATE "" diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h new file mode 100644 index 000000000000..9c661f2f8cda --- /dev/null +++ b/include/sound/wm8993.h @@ -0,0 +1,44 @@ +/* + * linux/sound/wm8993.h -- Platform data for WM8993 + * + * Copyright 2009 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_WM8993_H +#define __LINUX_SND_WM8993_H + +/* Note that EQ1 only contains the enable/disable bit so will be + ignored but is included for simplicity. + */ +struct wm8993_retune_mobile_setting { + const char *name; + unsigned int rate; + u16 config[24]; +}; + +struct wm8993_platform_data { + struct wm8993_retune_mobile_setting *retune_configs; + int num_retune_configs; + + /* LINEOUT can be differential or single ended */ + unsigned int lineout1_diff:1; + unsigned int lineout2_diff:1; + + /* Common mode feedback */ + unsigned int lineout1fb:1; + unsigned int lineout2fb:1; + + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ + unsigned int micbias1_lvl:1; + unsigned int micbias2_lvl:1; + + /* Jack detect threashold levels, see datasheet for values */ + unsigned int jd_scthr:2; + unsigned int jd_thr:2; +}; + +#endif diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h index 05ead6698434..444cd6ba0ba7 100644 --- a/include/sound/ymfpci.h +++ b/include/sound/ymfpci.h @@ -331,6 +331,7 @@ struct snd_ymfpci { struct snd_ac97 *ac97; struct snd_rawmidi *rawmidi; struct snd_timer *timer; + unsigned int timer_ticks; struct pci_dev *pci; struct snd_card *card; diff --git a/sound/Kconfig b/sound/Kconfig index 1eceb85287c5..439e15c8faa3 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -32,6 +32,34 @@ config SOUND_OSS_CORE bool default n +config SOUND_OSS_CORE_PRECLAIM + bool "Preclaim OSS device numbers" + depends on SOUND_OSS_CORE + default y + help + With this option enabled, the kernel will claim all OSS device + numbers if any OSS support (native or emulation) is enabled + whether the respective module is loaded or not and try to load the + appropriate module using sound-slot/service-* and char-major-* + module aliases when one of the device numbers is opened. With + this option disabled, kernel will only claim actually in-use + device numbers and opening a missing device will generate only the + standard char-major-* aliases. + + The only visible difference is use of additional module aliases + and whether OSS sound devices appear multiple times in + /proc/devices. sound-slot/service-* module aliases are scheduled + to be removed (ie. PRECLAIM won't be available) and this option is + to make the transition easier. This option can be overridden + during boot using the kernel parameter soundcore.preclaim_oss. + + Disabling this allows alternative OSS implementations. + + Please read Documentation/feature-removal-schedule.txt for + details. + + If unusre, say Y. + source "sound/oss/dmasound/Kconfig" if !M68K diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index c570ebd9d177..4e34d19ddbc0 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -170,6 +170,13 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; int ret; + pxa2xx_audio_ops_t *pdata = dev->dev.platform_data; + + if (dev->id >= 0) { + dev_err(&dev->dev, "PXA2xx has only one AC97 port.\n"); + ret = -ENXIO; + goto err_dev; + } ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, 0, &card); @@ -200,6 +207,8 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) snprintf(card->longname, sizeof(card->longname), "%s (%s)", dev->dev.driver->name, card->mixername); + if (pdata && pdata->codec_pdata[0]) + snd_ac97_dev_add_pdata(ac97_bus->codec[0], pdata->codec_pdata[0]); snd_card_set_dev(card, &dev->dev); ret = snd_card_register(card); if (ret == 0) { @@ -212,6 +221,7 @@ err_remove: err: if (card) snd_card_free(card); +err_dev: return ret; } diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 6205f37d547c..743ac6a29065 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -136,6 +136,9 @@ int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; + if (!prtd || !prtd->params) + return 0; + DCSR(prtd->dma_ch) &= ~DCSR_RUN; DCSR(prtd->dma_ch) = 0; DCMD(prtd->dma_ch) = 0; diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6061fb5f4e1c..c15682a2f9db 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -206,4 +206,8 @@ config SND_PCM_XRUN_DEBUG config SND_VMASTER bool +config SND_DMA_SGBUF + def_bool y + depends on X86 + source "sound/core/seq/Kconfig" diff --git a/sound/core/Makefile b/sound/core/Makefile index 4229052e7b91..350a08d277f4 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,7 +13,7 @@ snd-pcm-objs := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ pcm_memory.o snd-page-alloc-y := memalloc.o -snd-page-alloc-$(CONFIG_HAS_DMA) += sgbuf.o +snd-page-alloc-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-rawmidi-objs := rawmidi.o snd-timer-objs := timer.o diff --git a/sound/core/control.c b/sound/core/control.c index 17b8d47a5cd0..a8b7fabe645e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -414,7 +414,7 @@ int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id) EXPORT_SYMBOL(snd_ctl_remove_id); /** - * snd_ctl_remove_unlocked_id - remove the unlocked control of the given id and release it + * snd_ctl_remove_user_ctl - remove and release the unlocked user control * @file: active control handle * @id: the control id to remove * @@ -423,8 +423,8 @@ EXPORT_SYMBOL(snd_ctl_remove_id); * * Returns 0 if successful, or a negative error code on failure. */ -static int snd_ctl_remove_unlocked_id(struct snd_ctl_file * file, - struct snd_ctl_elem_id *id) +static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, + struct snd_ctl_elem_id *id) { struct snd_card *card = file->card; struct snd_kcontrol *kctl; @@ -433,15 +433,23 @@ static int snd_ctl_remove_unlocked_id(struct snd_ctl_file * file, down_write(&card->controls_rwsem); kctl = snd_ctl_find_id(card, id); if (kctl == NULL) { - up_write(&card->controls_rwsem); - return -ENOENT; + ret = -ENOENT; + goto error; + } + if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_USER)) { + ret = -EINVAL; + goto error; } for (idx = 0; idx < kctl->count; idx++) if (kctl->vd[idx].owner != NULL && kctl->vd[idx].owner != file) { - up_write(&card->controls_rwsem); - return -EBUSY; + ret = -EBUSY; + goto error; } ret = snd_ctl_remove(card, kctl); + if (ret < 0) + goto error; + card->user_ctl_count--; +error: up_write(&card->controls_rwsem); return ret; } @@ -951,7 +959,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (card->user_ctl_count >= MAX_USER_CONTROLS) return -ENOMEM; - if (info->count > 1024) + if (info->count < 1) return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| @@ -1052,18 +1060,10 @@ static int snd_ctl_elem_remove(struct snd_ctl_file *file, struct snd_ctl_elem_id __user *_id) { struct snd_ctl_elem_id id; - int err; if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; - err = snd_ctl_remove_unlocked_id(file, &id); - if (! err) { - struct snd_card *card = file->card; - down_write(&card->controls_rwsem); - card->user_ctl_count--; - up_write(&card->controls_rwsem); - } - return err; + return snd_ctl_remove_user_ctl(file, &id); } static int snd_ctl_subscribe_events(struct snd_ctl_file *file, int __user *ptr) diff --git a/sound/core/info.c b/sound/core/info.c index 35df614f6c55..d749a0d394a7 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -88,12 +88,10 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, char *nbuf; nsize = PAGE_ALIGN(nsize); - nbuf = kmalloc(nsize, GFP_KERNEL); + nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL); if (! nbuf) return -ENOMEM; - memcpy(nbuf, buffer->buffer, buffer->len); - kfree(buffer->buffer); buffer->buffer = nbuf; buffer->len = nsize; return 0; @@ -108,7 +106,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, * * Returns the size of output string. */ -int snd_iprintf(struct snd_info_buffer *buffer, char *fmt,...) +int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) { va_list args; int len, res; @@ -727,7 +725,7 @@ EXPORT_SYMBOL(snd_info_get_line); * Returns the updated pointer of the original string so that * it can be used for the next call. */ -char *snd_info_get_str(char *dest, char *src, int len) +const char *snd_info_get_str(char *dest, const char *src, int len) { int c; diff --git a/sound/core/init.c b/sound/core/init.c index d5d40d78c409..ec4a50ce5656 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -31,6 +31,14 @@ #include <sound/control.h> #include <sound/info.h> +/* monitor files for graceful shutdown (hotplug) */ +struct snd_monitor_file { + struct file *file; + const struct file_operations *disconnected_f_op; + struct list_head shutdown_list; /* still need to shutdown */ + struct list_head list; /* link of monitor files */ +}; + static DEFINE_SPINLOCK(shutdown_lock); static LIST_HEAD(shutdown_files); diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 1b3534d67686..9e92441f9b78 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -199,6 +199,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, case SNDRV_DMA_TYPE_DEV: dmab->area = snd_malloc_dev_pages(device, size, &dmab->addr); break; +#endif +#ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: snd_malloc_sgbuf_pages(device, size, dmab, NULL); break; @@ -269,6 +271,8 @@ void snd_dma_free_pages(struct snd_dma_buffer *dmab) case SNDRV_DMA_TYPE_DEV: snd_free_dev_pages(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); break; +#endif +#ifdef CONFIG_SND_DMA_SGBUF case SNDRV_DMA_TYPE_DEV_SG: snd_free_sgbuf_pages(dmab); break; diff --git a/sound/core/misc.c b/sound/core/misc.c index a9710e0c97af..23a032c6d487 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -24,6 +24,20 @@ #include <linux/ioport.h> #include <sound/core.h> +#ifdef CONFIG_SND_DEBUG + +#ifdef CONFIG_SND_DEBUG_VERBOSE +#define DEFAULT_DEBUG_LEVEL 2 +#else +#define DEFAULT_DEBUG_LEVEL 1 +#endif + +static int debug = DEFAULT_DEBUG_LEVEL; +module_param(debug, int, 0644); +MODULE_PARM_DESC(debug, "Debug level (0 = disable)"); + +#endif /* CONFIG_SND_DEBUG */ + void release_and_free_resource(struct resource *res) { if (res) { @@ -35,46 +49,53 @@ void release_and_free_resource(struct resource *res) EXPORT_SYMBOL(release_and_free_resource); #ifdef CONFIG_SND_VERBOSE_PRINTK -void snd_verbose_printk(const char *file, int line, const char *format, ...) +/* strip the leading path if the given path is absolute */ +static const char *sanity_file_name(const char *path) { - va_list args; - - if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') { - char tmp[] = "<0>"; + if (*path == '/') + return strrchr(path, '/') + 1; + else + return path; +} + +/* print file and line with a certain printk prefix */ +static int print_snd_pfx(unsigned int level, const char *path, int line, + const char *format) +{ + const char *file = sanity_file_name(path); + char tmp[] = "<0>"; + const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT; + int ret = 0; + + if (format[0] == '<' && format[2] == '>') { tmp[1] = format[1]; - printk("%sALSA %s:%d: ", tmp, file, line); - format += 3; - } else { - printk("ALSA %s:%d: ", file, line); + pfx = tmp; + ret = 1; } - va_start(args, format); - vprintk(format, args); - va_end(args); + printk("%sALSA %s:%d: ", pfx, file, line); + return ret; } - -EXPORT_SYMBOL(snd_verbose_printk); +#else +#define print_snd_pfx(level, path, line, format) 0 #endif -#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK) -void snd_verbose_printd(const char *file, int line, const char *format, ...) +#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) +void __snd_printk(unsigned int level, const char *path, int line, + const char *format, ...) { va_list args; - if (format[0] == '<' && format[1] >= '0' && format[1] <= '7' && format[2] == '>') { - char tmp[] = "<0>"; - tmp[1] = format[1]; - printk("%sALSA %s:%d: ", tmp, file, line); - format += 3; - } else { - printk(KERN_DEBUG "ALSA %s:%d: ", file, line); - } +#ifdef CONFIG_SND_DEBUG + if (debug < level) + return; +#endif va_start(args, format); + if (print_snd_pfx(level, path, line, format)) + format += 3; /* skip the printk level-prefix */ vprintk(format, args); va_end(args); - } - -EXPORT_SYMBOL(snd_verbose_printd); +EXPORT_SYMBOL_GPL(__snd_printk); #endif #ifdef CONFIG_PCI diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 5dcd8a526970..772423889eb3 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1154,7 +1154,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_mixer_oss *mixer = entry->private_data; - char line[128], str[32], idxstr[16], *cptr; + char line[128], str[32], idxstr[16]; + const char *cptr; int ch, idx; struct snd_mixer_oss_assign_table *tbl; struct slot *slot; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index dbe406b82591..d9c96353121a 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1043,10 +1043,15 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) runtime->oss.channels = params_channels(params); runtime->oss.rate = params_rate(params); - runtime->oss.params = 0; - runtime->oss.prepare = 1; vfree(runtime->oss.buffer); runtime->oss.buffer = vmalloc(runtime->oss.period_bytes); + if (!runtime->oss.buffer) { + err = -ENOMEM; + goto failure; + } + + runtime->oss.params = 0; + runtime->oss.prepare = 1; runtime->oss.buffer_used = 0; if (runtime->dma_area) snd_pcm_format_set_silence(runtime->format, runtime->dma_area, bytes_to_samples(runtime, runtime->dma_bytes)); @@ -2836,7 +2841,8 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_pcm_str *pstr = entry->private_data; - char line[128], str[32], task_name[32], *ptr; + char line[128], str[32], task_name[32]; + const char *ptr; int idx1; struct snd_pcm_oss_setup *setup, *setup1, template; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 145931a9ff30..0c1440121c22 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -162,18 +162,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } -#ifdef CONFIG_SND_VERBOSE_PROCFS - -#define STATE(v) [SNDRV_PCM_STATE_##v] = #v -#define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v -#define READY(v) [SNDRV_PCM_READY_##v] = #v -#define XRUN(v) [SNDRV_PCM_XRUN_##v] = #v -#define SILENCE(v) [SNDRV_PCM_SILENCE_##v] = #v -#define TSTAMP(v) [SNDRV_PCM_TSTAMP_##v] = #v -#define ACCESS(v) [SNDRV_PCM_ACCESS_##v] = #v -#define START(v) [SNDRV_PCM_START_##v] = #v #define FORMAT(v) [SNDRV_PCM_FORMAT_##v] = #v -#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v static char *snd_pcm_format_names[] = { FORMAT(S8), @@ -216,10 +205,23 @@ static char *snd_pcm_format_names[] = { FORMAT(U18_3BE), }; -static const char *snd_pcm_format_name(snd_pcm_format_t format) +const char *snd_pcm_format_name(snd_pcm_format_t format) { return snd_pcm_format_names[format]; } +EXPORT_SYMBOL_GPL(snd_pcm_format_name); + +#ifdef CONFIG_SND_VERBOSE_PROCFS + +#define STATE(v) [SNDRV_PCM_STATE_##v] = #v +#define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v +#define READY(v) [SNDRV_PCM_READY_##v] = #v +#define XRUN(v) [SNDRV_PCM_XRUN_##v] = #v +#define SILENCE(v) [SNDRV_PCM_SILENCE_##v] = #v +#define TSTAMP(v) [SNDRV_PCM_TSTAMP_##v] = #v +#define ACCESS(v) [SNDRV_PCM_ACCESS_##v] = #v +#define START(v) [SNDRV_PCM_START_##v] = #v +#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v static char *snd_pcm_stream_names[] = { STREAM(PLAYBACK), diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9db60d831bb2..30f410832a25 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -197,12 +197,16 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, avail = snd_pcm_capture_avail(runtime); if (avail > runtime->avail_max) runtime->avail_max = avail; - if (avail >= runtime->stop_threshold) { - if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { + if (avail >= runtime->buffer_size) { snd_pcm_drain_done(substream); - else + return -EPIPE; + } + } else { + if (avail >= runtime->stop_threshold) { xrun(substream); - return -EPIPE; + return -EPIPE; + } } if (avail >= runtime->control->avail_min) wake_up(&runtime->sleep); diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index a6d42808828c..caa7796bc2f5 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -304,6 +304,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, EXPORT_SYMBOL(snd_pcm_lib_preallocate_pages_for_all); +#ifdef CONFIG_SND_DMA_SGBUF /** * snd_pcm_sgbuf_ops_page - get the page struct at the given offset * @substream: the pcm substream instance @@ -349,6 +350,7 @@ unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream, return size; } EXPORT_SYMBOL(snd_pcm_sgbuf_get_chunk_size); +#endif /* CONFIG_SND_DMA_SGBUF */ /** * snd_pcm_lib_malloc_pages - allocate the DMA buffer diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ac2150e0670d..59e5fbe6af51 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1343,8 +1343,6 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { - if (substream->f_flags & O_NONBLOCK) - return -EAGAIN; substream->runtime->trigger_master = substream; return 0; } @@ -1392,7 +1390,6 @@ static struct action_ops snd_pcm_action_drain_init = { struct drain_rec { struct snd_pcm_substream *substream; wait_queue_t wait; - snd_pcm_uframes_t stop_threshold; }; static int snd_pcm_drop(struct snd_pcm_substream *substream); @@ -1404,13 +1401,15 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream); * After this call, all streams are supposed to be either SETUP or DRAINING * (capture only) state. */ -static int snd_pcm_drain(struct snd_pcm_substream *substream) +static int snd_pcm_drain(struct snd_pcm_substream *substream, + struct file *file) { struct snd_card *card; struct snd_pcm_runtime *runtime; struct snd_pcm_substream *s; int result = 0; int i, num_drecs; + int nonblock = 0; struct drain_rec *drec, drec_tmp, *d; card = substream->pcm->card; @@ -1428,6 +1427,15 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } + if (file) { + if (file->f_flags & O_NONBLOCK) + nonblock = 1; + } else if (substream->f_flags & O_NONBLOCK) + nonblock = 1; + + if (nonblock) + goto lock; /* no need to allocate waitqueues */ + /* allocate temporary record for drain sync */ down_read(&snd_pcm_link_rwsem); if (snd_pcm_stream_linked(substream)) { @@ -1449,16 +1457,11 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) d->substream = s; init_waitqueue_entry(&d->wait, current); add_wait_queue(&runtime->sleep, &d->wait); - /* stop_threshold fixup to avoid endless loop when - * stop_threshold > buffer_size - */ - d->stop_threshold = runtime->stop_threshold; - if (runtime->stop_threshold > runtime->buffer_size) - runtime->stop_threshold = runtime->buffer_size; } } up_read(&snd_pcm_link_rwsem); + lock: snd_pcm_stream_lock_irq(substream); /* resume pause */ if (substream->runtime->status->state == SNDRV_PCM_STATE_PAUSED) @@ -1466,9 +1469,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) /* pre-start/stop - all running streams are changed to DRAINING state */ result = snd_pcm_action(&snd_pcm_action_drain_init, substream, 0); - if (result < 0) { - snd_pcm_stream_unlock_irq(substream); - goto _error; + if (result < 0) + goto unlock; + /* in non-blocking, we don't wait in ioctl but let caller poll */ + if (nonblock) { + result = -EAGAIN; + goto unlock; } for (;;) { @@ -1504,18 +1510,18 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream) } } + unlock: snd_pcm_stream_unlock_irq(substream); - _error: - for (i = 0; i < num_drecs; i++) { - d = &drec[i]; - runtime = d->substream->runtime; - remove_wait_queue(&runtime->sleep, &d->wait); - runtime->stop_threshold = d->stop_threshold; + if (!nonblock) { + for (i = 0; i < num_drecs; i++) { + d = &drec[i]; + runtime = d->substream->runtime; + remove_wait_queue(&runtime->sleep, &d->wait); + } + if (drec != &drec_tmp) + kfree(drec); } - - if (drec != &drec_tmp) - kfree(drec); snd_power_unlock(card); return result; @@ -2208,6 +2214,9 @@ static snd_pcm_sframes_t snd_pcm_playback_rewind(struct snd_pcm_substream *subst case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2253,6 +2262,9 @@ static snd_pcm_sframes_t snd_pcm_capture_rewind(struct snd_pcm_substream *substr case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2299,6 +2311,9 @@ static snd_pcm_sframes_t snd_pcm_playback_forward(struct snd_pcm_substream *subs case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2345,6 +2360,9 @@ static snd_pcm_sframes_t snd_pcm_capture_forward(struct snd_pcm_substream *subst case SNDRV_PCM_STATE_XRUN: ret = -EPIPE; goto __end; + case SNDRV_PCM_STATE_SUSPENDED: + ret = -ESTRPIPE; + goto __end; default: ret = -EBADFD; goto __end; @@ -2544,7 +2562,7 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_hw_params_old_user(substream, arg); #endif case SNDRV_PCM_IOCTL_DRAIN: - return snd_pcm_drain(substream); + return snd_pcm_drain(substream, file); case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_PAUSE: diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 473247c8e6d3..c0adc14c91f0 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -274,7 +274,7 @@ static int open_substream(struct snd_rawmidi *rmidi, return err; substream->opened = 1; if (substream->use_count++ == 0) - substream->active_sensing = 1; + substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; rmidi->streams[substream->stream].substream_opened++; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 0a711d2d04f0..9dfb2f77be60 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -20,6 +20,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include <sound/asoundef.h> #include "seq_oss_midi.h" #include "seq_oss_readq.h" #include "seq_oss_timer.h" @@ -476,19 +477,20 @@ snd_seq_oss_midi_reset(struct seq_oss_devinfo *dp, int dev) ev.source.port = dp->port; if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_SYNTH) { ev.type = SNDRV_SEQ_EVENT_SENSING; - snd_seq_oss_dispatch(dp, &ev, 0, 0); /* active sensing */ + snd_seq_oss_dispatch(dp, &ev, 0, 0); } for (c = 0; c < 16; c++) { ev.type = SNDRV_SEQ_EVENT_CONTROLLER; ev.data.control.channel = c; - ev.data.control.param = 123; - snd_seq_oss_dispatch(dp, &ev, 0, 0); /* all notes off */ + ev.data.control.param = MIDI_CTL_ALL_NOTES_OFF; + snd_seq_oss_dispatch(dp, &ev, 0, 0); if (dp->seq_mode == SNDRV_SEQ_OSS_MODE_MUSIC) { - ev.data.control.param = 121; - snd_seq_oss_dispatch(dp, &ev, 0, 0); /* reset all controllers */ + ev.data.control.param = + MIDI_CTL_RESET_CONTROLLERS; + snd_seq_oss_dispatch(dp, &ev, 0, 0); ev.type = SNDRV_SEQ_EVENT_PITCHBEND; ev.data.control.value = 0; - snd_seq_oss_dispatch(dp, &ev, 0, 0); /* bender off */ + snd_seq_oss_dispatch(dp, &ev, 0, 0); } } } diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 4d26146a62cc..ebaf1b541dcd 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -120,7 +120,8 @@ static int dump_midi(struct snd_rawmidi_substream *substream, const char *buf, i return -EINVAL; runtime = substream->runtime; if ((tmp = runtime->avail) < count) { - snd_printd("warning, output event was lost (count = %i, available = %i)\n", count, tmp); + if (printk_ratelimit()) + snd_printk(KERN_ERR "MIDI output buffer overrun\n"); return -ENOMEM; } if (snd_rawmidi_kernel_write(substream, buf, count) < count) @@ -236,6 +237,7 @@ static int midisynth_use(void *private_data, struct snd_seq_port_subscribe *info memset(¶ms, 0, sizeof(params)); params.avail_min = 1; params.buffer_size = output_buffer_size; + params.no_active_sensing = 1; if ((err = snd_rawmidi_output_params(msynth->output_rfile.output, ¶ms)) < 0) { snd_rawmidi_kernel_release(&msynth->output_rfile); return err; @@ -248,12 +250,9 @@ static int midisynth_use(void *private_data, struct snd_seq_port_subscribe *info static int midisynth_unuse(void *private_data, struct snd_seq_port_subscribe *info) { struct seq_midisynth *msynth = private_data; - unsigned char buf = 0xff; /* MIDI reset */ if (snd_BUG_ON(!msynth->output_rfile.output)) return -EINVAL; - /* sending single MIDI reset message to shut the device up */ - snd_rawmidi_kernel_write(msynth->output_rfile.output, &buf, 1); snd_rawmidi_drain_output(msynth->output_rfile.output); return snd_rawmidi_kernel_release(&msynth->output_rfile); } diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 257624bd1997..3b9b550109cb 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -353,7 +353,8 @@ static void master_free(struct snd_kcontrol *kcontrol) * * The optional argument @tlv can be used to specify the TLV information * for dB scale of the master control. It should be a single element - * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB. + * with #SNDRV_CTL_TLVT_DB_SCALE, #SNDRV_CTL_TLV_DB_MINMAX or + * #SNDRV_CTL_TLVT_DB_MINMAX_MUTE type, and should be the max 0dB. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -384,7 +385,10 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, kctl->private_free = master_free; /* additional (constant) TLV read */ - if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) { + if (tlv && + (tlv[0] == SNDRV_CTL_TLVT_DB_SCALE || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX || + tlv[0] == SNDRV_CTL_TLVT_DB_MINMAX_MUTE)) { kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; memcpy(master->tlv, tlv, sizeof(master->tlv)); kctl->tlv.p = master->tlv; diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 54239d2e0997..6ba066c41d2e 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -25,12 +25,15 @@ #include <linux/slab.h> #include <linux/time.h> #include <linux/wait.h> +#include <linux/hrtimer.h> +#include <linux/math64.h> #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/control.h> #include <sound/tlv.h> #include <sound/pcm.h> #include <sound/rawmidi.h> +#include <sound/info.h> #include <sound/initval.h> MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); @@ -39,7 +42,7 @@ MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); #define MAX_PCM_DEVICES 4 -#define MAX_PCM_SUBSTREAMS 16 +#define MAX_PCM_SUBSTREAMS 128 #define MAX_MIDI_DEVICES 2 #if 0 /* emu10k1 emulation */ @@ -148,6 +151,10 @@ static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; +#ifdef CONFIG_HIGH_RES_TIMERS +static int hrtimer = 1; +#endif +static int fake_buffer = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for dummy soundcard."); @@ -161,6 +168,12 @@ module_param_array(pcm_substreams, int, NULL, 0444); MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-16) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); +module_param(fake_buffer, bool, 0444); +MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations."); +#ifdef CONFIG_HIGH_RES_TIMERS +module_param(hrtimer, bool, 0644); +MODULE_PARM_DESC(hrtimer, "Use hrtimer as the timer source."); +#endif static struct platform_device *devices[SNDRV_CARDS]; @@ -171,137 +184,324 @@ static struct platform_device *devices[SNDRV_CARDS]; #define MIXER_ADDR_CD 4 #define MIXER_ADDR_LAST 4 +struct dummy_timer_ops { + int (*create)(struct snd_pcm_substream *); + void (*free)(struct snd_pcm_substream *); + int (*prepare)(struct snd_pcm_substream *); + int (*start)(struct snd_pcm_substream *); + int (*stop)(struct snd_pcm_substream *); + snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *); +}; + struct snd_dummy { struct snd_card *card; struct snd_pcm *pcm; spinlock_t mixer_lock; int mixer_volume[MIXER_ADDR_LAST+1][2]; int capture_source[MIXER_ADDR_LAST+1][2]; + const struct dummy_timer_ops *timer_ops; }; -struct snd_dummy_pcm { - struct snd_dummy *dummy; +/* + * system timer interface + */ + +struct dummy_systimer_pcm { spinlock_t lock; struct timer_list timer; - unsigned int pcm_buffer_size; - unsigned int pcm_period_size; - unsigned int pcm_bps; /* bytes per second */ - unsigned int pcm_hz; /* HZ */ - unsigned int pcm_irq_pos; /* IRQ position */ - unsigned int pcm_buf_pos; /* position in buffer */ + unsigned long base_time; + unsigned int frac_pos; /* fractional sample position (based HZ) */ + unsigned int frac_period_rest; + unsigned int frac_buffer_size; /* buffer_size * HZ */ + unsigned int frac_period_size; /* period_size * HZ */ + unsigned int rate; + int elapsed; struct snd_pcm_substream *substream; }; - -static inline void snd_card_dummy_pcm_timer_start(struct snd_dummy_pcm *dpcm) +static void dummy_systimer_rearm(struct dummy_systimer_pcm *dpcm) { - dpcm->timer.expires = 1 + jiffies; + dpcm->timer.expires = jiffies + + (dpcm->frac_period_rest + dpcm->rate - 1) / dpcm->rate; add_timer(&dpcm->timer); } -static inline void snd_card_dummy_pcm_timer_stop(struct snd_dummy_pcm *dpcm) +static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) { - del_timer(&dpcm->timer); + unsigned long delta; + + delta = jiffies - dpcm->base_time; + if (!delta) + return; + dpcm->base_time += delta; + delta *= dpcm->rate; + dpcm->frac_pos += delta; + while (dpcm->frac_pos >= dpcm->frac_buffer_size) + dpcm->frac_pos -= dpcm->frac_buffer_size; + while (dpcm->frac_period_rest <= delta) { + dpcm->elapsed++; + dpcm->frac_period_rest += dpcm->frac_period_size; + } + dpcm->frac_period_rest -= delta; } -static int snd_card_dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int dummy_systimer_start(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dummy_pcm *dpcm = runtime->private_data; - int err = 0; + struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; + spin_lock(&dpcm->lock); + dpcm->base_time = jiffies; + dummy_systimer_rearm(dpcm); + spin_unlock(&dpcm->lock); + return 0; +} +static int dummy_systimer_stop(struct snd_pcm_substream *substream) +{ + struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; spin_lock(&dpcm->lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - snd_card_dummy_pcm_timer_start(dpcm); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - snd_card_dummy_pcm_timer_stop(dpcm); - break; - default: - err = -EINVAL; - break; - } + del_timer(&dpcm->timer); spin_unlock(&dpcm->lock); return 0; } -static int snd_card_dummy_pcm_prepare(struct snd_pcm_substream *substream) +static int dummy_systimer_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dummy_pcm *dpcm = runtime->private_data; - int bps; - - bps = snd_pcm_format_width(runtime->format) * runtime->rate * - runtime->channels / 8; - - if (bps <= 0) - return -EINVAL; - - dpcm->pcm_bps = bps; - dpcm->pcm_hz = HZ; - dpcm->pcm_buffer_size = snd_pcm_lib_buffer_bytes(substream); - dpcm->pcm_period_size = snd_pcm_lib_period_bytes(substream); - dpcm->pcm_irq_pos = 0; - dpcm->pcm_buf_pos = 0; + struct dummy_systimer_pcm *dpcm = runtime->private_data; - snd_pcm_format_set_silence(runtime->format, runtime->dma_area, - bytes_to_samples(runtime, runtime->dma_bytes)); + dpcm->frac_pos = 0; + dpcm->rate = runtime->rate; + dpcm->frac_buffer_size = runtime->buffer_size * HZ; + dpcm->frac_period_size = runtime->period_size * HZ; + dpcm->frac_period_rest = dpcm->frac_period_size; + dpcm->elapsed = 0; return 0; } -static void snd_card_dummy_pcm_timer_function(unsigned long data) +static void dummy_systimer_callback(unsigned long data) { - struct snd_dummy_pcm *dpcm = (struct snd_dummy_pcm *)data; + struct dummy_systimer_pcm *dpcm = (struct dummy_systimer_pcm *)data; unsigned long flags; + int elapsed = 0; spin_lock_irqsave(&dpcm->lock, flags); - dpcm->timer.expires = 1 + jiffies; - add_timer(&dpcm->timer); - dpcm->pcm_irq_pos += dpcm->pcm_bps; - dpcm->pcm_buf_pos += dpcm->pcm_bps; - dpcm->pcm_buf_pos %= dpcm->pcm_buffer_size * dpcm->pcm_hz; - if (dpcm->pcm_irq_pos >= dpcm->pcm_period_size * dpcm->pcm_hz) { - dpcm->pcm_irq_pos %= dpcm->pcm_period_size * dpcm->pcm_hz; - spin_unlock_irqrestore(&dpcm->lock, flags); + dummy_systimer_update(dpcm); + dummy_systimer_rearm(dpcm); + elapsed = dpcm->elapsed; + dpcm->elapsed = 0; + spin_unlock_irqrestore(&dpcm->lock, flags); + if (elapsed) + snd_pcm_period_elapsed(dpcm->substream); +} + +static snd_pcm_uframes_t +dummy_systimer_pointer(struct snd_pcm_substream *substream) +{ + struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; + snd_pcm_uframes_t pos; + + spin_lock(&dpcm->lock); + dummy_systimer_update(dpcm); + pos = dpcm->frac_pos / HZ; + spin_unlock(&dpcm->lock); + return pos; +} + +static int dummy_systimer_create(struct snd_pcm_substream *substream) +{ + struct dummy_systimer_pcm *dpcm; + + dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL); + if (!dpcm) + return -ENOMEM; + substream->runtime->private_data = dpcm; + init_timer(&dpcm->timer); + dpcm->timer.data = (unsigned long) dpcm; + dpcm->timer.function = dummy_systimer_callback; + spin_lock_init(&dpcm->lock); + dpcm->substream = substream; + return 0; +} + +static void dummy_systimer_free(struct snd_pcm_substream *substream) +{ + kfree(substream->runtime->private_data); +} + +static struct dummy_timer_ops dummy_systimer_ops = { + .create = dummy_systimer_create, + .free = dummy_systimer_free, + .prepare = dummy_systimer_prepare, + .start = dummy_systimer_start, + .stop = dummy_systimer_stop, + .pointer = dummy_systimer_pointer, +}; + +#ifdef CONFIG_HIGH_RES_TIMERS +/* + * hrtimer interface + */ + +struct dummy_hrtimer_pcm { + ktime_t base_time; + ktime_t period_time; + atomic_t running; + struct hrtimer timer; + struct tasklet_struct tasklet; + struct snd_pcm_substream *substream; +}; + +static void dummy_hrtimer_pcm_elapsed(unsigned long priv) +{ + struct dummy_hrtimer_pcm *dpcm = (struct dummy_hrtimer_pcm *)priv; + if (atomic_read(&dpcm->running)) snd_pcm_period_elapsed(dpcm->substream); - } else - spin_unlock_irqrestore(&dpcm->lock, flags); } -static snd_pcm_uframes_t snd_card_dummy_pcm_pointer(struct snd_pcm_substream *substream) +static enum hrtimer_restart dummy_hrtimer_callback(struct hrtimer *timer) +{ + struct dummy_hrtimer_pcm *dpcm; + + dpcm = container_of(timer, struct dummy_hrtimer_pcm, timer); + if (!atomic_read(&dpcm->running)) + return HRTIMER_NORESTART; + tasklet_schedule(&dpcm->tasklet); + hrtimer_forward_now(timer, dpcm->period_time); + return HRTIMER_RESTART; +} + +static int dummy_hrtimer_start(struct snd_pcm_substream *substream) +{ + struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; + + dpcm->base_time = hrtimer_cb_get_time(&dpcm->timer); + hrtimer_start(&dpcm->timer, dpcm->period_time, HRTIMER_MODE_REL); + atomic_set(&dpcm->running, 1); + return 0; +} + +static int dummy_hrtimer_stop(struct snd_pcm_substream *substream) +{ + struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; + + atomic_set(&dpcm->running, 0); + hrtimer_cancel(&dpcm->timer); + return 0; +} + +static inline void dummy_hrtimer_sync(struct dummy_hrtimer_pcm *dpcm) +{ + tasklet_kill(&dpcm->tasklet); +} + +static snd_pcm_uframes_t +dummy_hrtimer_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dummy_pcm *dpcm = runtime->private_data; + struct dummy_hrtimer_pcm *dpcm = runtime->private_data; + u64 delta; + u32 pos; + + delta = ktime_us_delta(hrtimer_cb_get_time(&dpcm->timer), + dpcm->base_time); + delta = div_u64(delta * runtime->rate + 999999, 1000000); + div_u64_rem(delta, runtime->buffer_size, &pos); + return pos; +} - return bytes_to_frames(runtime, dpcm->pcm_buf_pos / dpcm->pcm_hz); +static int dummy_hrtimer_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct dummy_hrtimer_pcm *dpcm = runtime->private_data; + unsigned int period, rate; + long sec; + unsigned long nsecs; + + dummy_hrtimer_sync(dpcm); + period = runtime->period_size; + rate = runtime->rate; + sec = period / rate; + period %= rate; + nsecs = div_u64((u64)period * 1000000000UL + rate - 1, rate); + dpcm->period_time = ktime_set(sec, nsecs); + + return 0; } -static struct snd_pcm_hardware snd_card_dummy_playback = +static int dummy_hrtimer_create(struct snd_pcm_substream *substream) { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), - .formats = USE_FORMATS, - .rates = USE_RATE, - .rate_min = USE_RATE_MIN, - .rate_max = USE_RATE_MAX, - .channels_min = USE_CHANNELS_MIN, - .channels_max = USE_CHANNELS_MAX, - .buffer_bytes_max = MAX_BUFFER_SIZE, - .period_bytes_min = 64, - .period_bytes_max = MAX_PERIOD_SIZE, - .periods_min = USE_PERIODS_MIN, - .periods_max = USE_PERIODS_MAX, - .fifo_size = 0, + struct dummy_hrtimer_pcm *dpcm; + + dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL); + if (!dpcm) + return -ENOMEM; + substream->runtime->private_data = dpcm; + hrtimer_init(&dpcm->timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + dpcm->timer.function = dummy_hrtimer_callback; + dpcm->substream = substream; + atomic_set(&dpcm->running, 0); + tasklet_init(&dpcm->tasklet, dummy_hrtimer_pcm_elapsed, + (unsigned long)dpcm); + return 0; +} + +static void dummy_hrtimer_free(struct snd_pcm_substream *substream) +{ + struct dummy_hrtimer_pcm *dpcm = substream->runtime->private_data; + dummy_hrtimer_sync(dpcm); + kfree(dpcm); +} + +static struct dummy_timer_ops dummy_hrtimer_ops = { + .create = dummy_hrtimer_create, + .free = dummy_hrtimer_free, + .prepare = dummy_hrtimer_prepare, + .start = dummy_hrtimer_start, + .stop = dummy_hrtimer_stop, + .pointer = dummy_hrtimer_pointer, }; -static struct snd_pcm_hardware snd_card_dummy_capture = +#endif /* CONFIG_HIGH_RES_TIMERS */ + +/* + * PCM interface + */ + +static int dummy_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), + struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + return dummy->timer_ops->start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + return dummy->timer_ops->stop(substream); + } + return -EINVAL; +} + +static int dummy_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + + return dummy->timer_ops->prepare(substream); +} + +static snd_pcm_uframes_t dummy_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + + return dummy->timer_ops->pointer(substream); +} + +static struct snd_pcm_hardware dummy_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP_VALID), .formats = USE_FORMATS, .rates = USE_RATE, .rate_min = USE_RATE_MIN, @@ -316,123 +516,152 @@ static struct snd_pcm_hardware snd_card_dummy_capture = .fifo_size = 0, }; -static void snd_card_dummy_runtime_free(struct snd_pcm_runtime *runtime) -{ - kfree(runtime->private_data); -} - -static int snd_card_dummy_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) +static int dummy_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); + if (fake_buffer) { + /* runtime->dma_bytes has to be set manually to allow mmap */ + substream->runtime->dma_bytes = params_buffer_bytes(hw_params); + return 0; + } + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); } -static int snd_card_dummy_hw_free(struct snd_pcm_substream *substream) +static int dummy_pcm_hw_free(struct snd_pcm_substream *substream) { + if (fake_buffer) + return 0; return snd_pcm_lib_free_pages(substream); } -static struct snd_dummy_pcm *new_pcm_stream(struct snd_pcm_substream *substream) -{ - struct snd_dummy_pcm *dpcm; - - dpcm = kzalloc(sizeof(*dpcm), GFP_KERNEL); - if (! dpcm) - return dpcm; - init_timer(&dpcm->timer); - dpcm->timer.data = (unsigned long) dpcm; - dpcm->timer.function = snd_card_dummy_pcm_timer_function; - spin_lock_init(&dpcm->lock); - dpcm->substream = substream; - return dpcm; -} - -static int snd_card_dummy_playback_open(struct snd_pcm_substream *substream) +static int dummy_pcm_open(struct snd_pcm_substream *substream) { + struct snd_dummy *dummy = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dummy_pcm *dpcm; int err; - if ((dpcm = new_pcm_stream(substream)) == NULL) - return -ENOMEM; - runtime->private_data = dpcm; - /* makes the infrastructure responsible for freeing dpcm */ - runtime->private_free = snd_card_dummy_runtime_free; - runtime->hw = snd_card_dummy_playback; + dummy->timer_ops = &dummy_systimer_ops; +#ifdef CONFIG_HIGH_RES_TIMERS + if (hrtimer) + dummy->timer_ops = &dummy_hrtimer_ops; +#endif + + err = dummy->timer_ops->create(substream); + if (err < 0) + return err; + + runtime->hw = dummy_pcm_hardware; if (substream->pcm->device & 1) { runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; } if (substream->pcm->device & 2) - runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID); - err = add_playback_constraints(runtime); - if (err < 0) + runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + err = add_playback_constraints(substream->runtime); + else + err = add_capture_constraints(substream->runtime); + if (err < 0) { + dummy->timer_ops->free(substream); return err; - + } return 0; } -static int snd_card_dummy_capture_open(struct snd_pcm_substream *substream) +static int dummy_pcm_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dummy_pcm *dpcm; - int err; + struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + dummy->timer_ops->free(substream); + return 0; +} - if ((dpcm = new_pcm_stream(substream)) == NULL) - return -ENOMEM; - runtime->private_data = dpcm; - /* makes the infrastructure responsible for freeing dpcm */ - runtime->private_free = snd_card_dummy_runtime_free; - runtime->hw = snd_card_dummy_capture; - if (substream->pcm->device == 1) { - runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; - runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; +/* + * dummy buffer handling + */ + +static void *dummy_page[2]; + +static void free_fake_buffer(void) +{ + if (fake_buffer) { + int i; + for (i = 0; i < 2; i++) + if (dummy_page[i]) { + free_page((unsigned long)dummy_page[i]); + dummy_page[i] = NULL; + } } - if (substream->pcm->device & 2) - runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP|SNDRV_PCM_INFO_MMAP_VALID); - err = add_capture_constraints(runtime); - if (err < 0) - return err; +} +static int alloc_fake_buffer(void) +{ + int i; + + if (!fake_buffer) + return 0; + for (i = 0; i < 2; i++) { + dummy_page[i] = (void *)get_zeroed_page(GFP_KERNEL); + if (!dummy_page[i]) { + free_fake_buffer(); + return -ENOMEM; + } + } return 0; } -static int snd_card_dummy_playback_close(struct snd_pcm_substream *substream) +static int dummy_pcm_copy(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + void __user *dst, snd_pcm_uframes_t count) { - return 0; + return 0; /* do nothing */ } -static int snd_card_dummy_capture_close(struct snd_pcm_substream *substream) +static int dummy_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) { - return 0; + return 0; /* do nothing */ +} + +static struct page *dummy_pcm_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return virt_to_page(dummy_page[substream->stream]); /* the same page */ } -static struct snd_pcm_ops snd_card_dummy_playback_ops = { - .open = snd_card_dummy_playback_open, - .close = snd_card_dummy_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_card_dummy_hw_params, - .hw_free = snd_card_dummy_hw_free, - .prepare = snd_card_dummy_pcm_prepare, - .trigger = snd_card_dummy_pcm_trigger, - .pointer = snd_card_dummy_pcm_pointer, +static struct snd_pcm_ops dummy_pcm_ops = { + .open = dummy_pcm_open, + .close = dummy_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dummy_pcm_hw_params, + .hw_free = dummy_pcm_hw_free, + .prepare = dummy_pcm_prepare, + .trigger = dummy_pcm_trigger, + .pointer = dummy_pcm_pointer, }; -static struct snd_pcm_ops snd_card_dummy_capture_ops = { - .open = snd_card_dummy_capture_open, - .close = snd_card_dummy_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_card_dummy_hw_params, - .hw_free = snd_card_dummy_hw_free, - .prepare = snd_card_dummy_pcm_prepare, - .trigger = snd_card_dummy_pcm_trigger, - .pointer = snd_card_dummy_pcm_pointer, +static struct snd_pcm_ops dummy_pcm_ops_no_buf = { + .open = dummy_pcm_open, + .close = dummy_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dummy_pcm_hw_params, + .hw_free = dummy_pcm_hw_free, + .prepare = dummy_pcm_prepare, + .trigger = dummy_pcm_trigger, + .pointer = dummy_pcm_pointer, + .copy = dummy_pcm_copy, + .silence = dummy_pcm_silence, + .page = dummy_pcm_page, }; static int __devinit snd_card_dummy_pcm(struct snd_dummy *dummy, int device, int substreams) { struct snd_pcm *pcm; + struct snd_pcm_ops *ops; int err; err = snd_pcm_new(dummy->card, "Dummy PCM", device, @@ -440,17 +669,28 @@ static int __devinit snd_card_dummy_pcm(struct snd_dummy *dummy, int device, if (err < 0) return err; dummy->pcm = pcm; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_dummy_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_dummy_capture_ops); + if (fake_buffer) + ops = &dummy_pcm_ops_no_buf; + else + ops = &dummy_pcm_ops; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, ops); pcm->private_data = dummy; pcm->info_flags = 0; strcpy(pcm->name, "Dummy PCM"); - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - 0, 64*1024); + if (!fake_buffer) { + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 0, 64*1024); + } return 0; } +/* + * mixer interface + */ + #define DUMMY_VOLUME(xname, xindex, addr) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ @@ -581,6 +821,131 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) return 0; } +#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_PROC_FS) +/* + * proc interface + */ +static void print_formats(struct snd_info_buffer *buffer) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + if (dummy_pcm_hardware.formats & (1ULL << i)) + snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); + } +} + +static void print_rates(struct snd_info_buffer *buffer) +{ + static int rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, + 64000, 88200, 96000, 176400, 192000, + }; + int i; + + if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_CONTINUOUS) + snd_iprintf(buffer, " continuous"); + if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_KNOT) + snd_iprintf(buffer, " knot"); + for (i = 0; i < ARRAY_SIZE(rates); i++) + if (dummy_pcm_hardware.rates & (1 << i)) + snd_iprintf(buffer, " %d", rates[i]); +} + +#define get_dummy_int_ptr(ofs) \ + (unsigned int *)((char *)&dummy_pcm_hardware + (ofs)) +#define get_dummy_ll_ptr(ofs) \ + (unsigned long long *)((char *)&dummy_pcm_hardware + (ofs)) + +struct dummy_hw_field { + const char *name; + const char *format; + unsigned int offset; + unsigned int size; +}; +#define FIELD_ENTRY(item, fmt) { \ + .name = #item, \ + .format = fmt, \ + .offset = offsetof(struct snd_pcm_hardware, item), \ + .size = sizeof(dummy_pcm_hardware.item) } + +static struct dummy_hw_field fields[] = { + FIELD_ENTRY(formats, "%#llx"), + FIELD_ENTRY(rates, "%#x"), + FIELD_ENTRY(rate_min, "%d"), + FIELD_ENTRY(rate_max, "%d"), + FIELD_ENTRY(channels_min, "%d"), + FIELD_ENTRY(channels_max, "%d"), + FIELD_ENTRY(buffer_bytes_max, "%ld"), + FIELD_ENTRY(period_bytes_min, "%ld"), + FIELD_ENTRY(period_bytes_max, "%ld"), + FIELD_ENTRY(periods_min, "%d"), + FIELD_ENTRY(periods_max, "%d"), +}; + +static void dummy_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(fields); i++) { + snd_iprintf(buffer, "%s ", fields[i].name); + if (fields[i].size == sizeof(int)) + snd_iprintf(buffer, fields[i].format, + *get_dummy_int_ptr(fields[i].offset)); + else + snd_iprintf(buffer, fields[i].format, + *get_dummy_ll_ptr(fields[i].offset)); + if (!strcmp(fields[i].name, "formats")) + print_formats(buffer); + else if (!strcmp(fields[i].name, "rates")) + print_rates(buffer); + snd_iprintf(buffer, "\n"); + } +} + +static void dummy_proc_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + char line[64]; + + while (!snd_info_get_line(buffer, line, sizeof(line))) { + char item[20]; + const char *ptr; + unsigned long long val; + int i; + + ptr = snd_info_get_str(item, line, sizeof(item)); + for (i = 0; i < ARRAY_SIZE(fields); i++) { + if (!strcmp(item, fields[i].name)) + break; + } + if (i >= ARRAY_SIZE(fields)) + continue; + snd_info_get_str(item, ptr, sizeof(item)); + if (strict_strtoull(item, 0, &val)) + continue; + if (fields[i].size == sizeof(int)) + *get_dummy_int_ptr(fields[i].offset) = val; + else + *get_dummy_ll_ptr(fields[i].offset) = val; + } +} + +static void __devinit dummy_proc_init(struct snd_dummy *chip) +{ + struct snd_info_entry *entry; + + if (!snd_card_proc_new(chip->card, "dummy_pcm", &entry)) { + snd_info_set_text_ops(entry, chip, dummy_proc_read); + entry->c.text.write = dummy_proc_write; + entry->mode |= S_IWUSR; + } +} +#else +#define dummy_proc_init(x) +#endif /* CONFIG_SND_DEBUG && CONFIG_PROC_FS */ + static int __devinit snd_dummy_probe(struct platform_device *devptr) { struct snd_card *card; @@ -610,6 +975,8 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) strcpy(card->shortname, "Dummy"); sprintf(card->longname, "Dummy %i", dev + 1); + dummy_proc_init(dummy); + snd_card_set_dev(card, &devptr->dev); err = snd_card_register(card); @@ -670,6 +1037,7 @@ static void snd_dummy_unregister_all(void) for (i = 0; i < ARRAY_SIZE(devices); ++i) platform_device_unregister(devices[i]); platform_driver_unregister(&snd_dummy_driver); + free_fake_buffer(); } static int __init alsa_card_dummy_init(void) @@ -680,6 +1048,12 @@ static int __init alsa_card_dummy_init(void) if (err < 0) return err; + err = alloc_fake_buffer(); + if (err < 0) { + platform_driver_unregister(&snd_dummy_driver); + return err; + } + cards = 0; for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 3ee0269e5bd0..02f79d252718 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -1,5 +1,5 @@ /* - * Driver for C-Media's CMI8330 soundcards. + * Driver for C-Media's CMI8330 and CMI8329 soundcards. * Copyright (c) by George Talusan <gstalusan@uwaterloo.ca> * http://www.undergrad.math.uwaterloo.ca/~gstalusa * @@ -35,7 +35,7 @@ * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the CMI8330 codec, + * The driver "magically" routes the capturing to the AD1848 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -64,7 +64,7 @@ /* */ MODULE_AUTHOR("George Talusan <gstalusan@uwaterloo.ca>"); -MODULE_DESCRIPTION("C-Media CMI8330"); +MODULE_DESCRIPTION("C-Media CMI8330/CMI8329"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{C-Media,CMI8330,isapnp:{CMI0001,@@@0001,@X@0001}}}"); @@ -86,38 +86,38 @@ static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; module_param_array(index, int, NULL, 0444); -MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); +MODULE_PARM_DESC(index, "Index value for CMI8330/CMI8329 soundcard."); module_param_array(id, charp, NULL, 0444); -MODULE_PARM_DESC(id, "ID string for CMI8330 soundcard."); +MODULE_PARM_DESC(id, "ID string for CMI8330/CMI8329 soundcard."); module_param_array(enable, bool, NULL, 0444); -MODULE_PARM_DESC(enable, "Enable CMI8330 soundcard."); +MODULE_PARM_DESC(enable, "Enable CMI8330/CMI8329 soundcard."); #ifdef CONFIG_PNP module_param_array(isapnp, bool, NULL, 0444); MODULE_PARM_DESC(isapnp, "PnP detection for specified soundcard."); #endif module_param_array(sbport, long, NULL, 0444); -MODULE_PARM_DESC(sbport, "Port # for CMI8330 SB driver."); +MODULE_PARM_DESC(sbport, "Port # for CMI8330/CMI8329 SB driver."); module_param_array(sbirq, int, NULL, 0444); -MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330 SB driver."); +MODULE_PARM_DESC(sbirq, "IRQ # for CMI8330/CMI8329 SB driver."); module_param_array(sbdma8, int, NULL, 0444); -MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330 SB driver."); +MODULE_PARM_DESC(sbdma8, "DMA8 for CMI8330/CMI8329 SB driver."); module_param_array(sbdma16, int, NULL, 0444); -MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330 SB driver."); +MODULE_PARM_DESC(sbdma16, "DMA16 for CMI8330/CMI8329 SB driver."); module_param_array(wssport, long, NULL, 0444); -MODULE_PARM_DESC(wssport, "Port # for CMI8330 WSS driver."); +MODULE_PARM_DESC(wssport, "Port # for CMI8330/CMI8329 WSS driver."); module_param_array(wssirq, int, NULL, 0444); -MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver."); +MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330/CMI8329 WSS driver."); module_param_array(wssdma, int, NULL, 0444); -MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); +MODULE_PARM_DESC(wssdma, "DMA for CMI8330/CMI8329 WSS driver."); module_param_array(fmport, long, NULL, 0444); -MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); +MODULE_PARM_DESC(fmport, "FM port # for CMI8330/CMI8329 driver."); module_param_array(mpuport, long, NULL, 0444); -MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver."); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330/CMI8329 driver."); module_param_array(mpuirq, int, NULL, 0444); -MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port."); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330/CMI8329 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -156,6 +156,11 @@ static unsigned char snd_cmi8330_image[((CMI8330_CDINGAIN)-16) + 1] = typedef int (*snd_pcm_open_callback_t)(struct snd_pcm_substream *); +enum card_type { + CMI8330, + CMI8329 +}; + struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; @@ -172,11 +177,14 @@ struct snd_cmi8330 { snd_pcm_open_callback_t open; void *private_data; /* sb or wss */ } streams[2]; + + enum card_type type; }; #ifdef CONFIG_PNP static struct pnp_card_device_id snd_cmi8330_pnpids[] = { + { .id = "CMI0001", .devs = { { "@X@0001" }, { "@@@0001" }, { "@H@0001" }, { "A@@0001" } } }, { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, { .id = "" } }; @@ -304,7 +312,7 @@ static int __devinit snd_cmi8330_mixer(struct snd_card *card, struct snd_cmi8330 unsigned int idx; int err; - strcpy(card->mixername, "CMI8330/C3D"); + strcpy(card->mixername, (acard->type == CMI8329) ? "CMI8329" : "CMI8330/C3D"); for (idx = 0; idx < ARRAY_SIZE(snd_cmi8330_controls); idx++) { err = snd_ctl_add(card, @@ -329,6 +337,9 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, struct pnp_dev *pdev; int err; + /* CMI8329 has a device with ID A@@0001, CMI8330 does not */ + acard->type = (id->devs[3].id[0]) ? CMI8329 : CMI8330; + acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL); if (acard->cap == NULL) return -EBUSY; @@ -345,38 +356,45 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); + snd_printk(KERN_ERR "AD1848 PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - fmport[dev] = pnp_port_start(pdev, 1); + if (pnp_port_start(pdev, 1)) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP configure failure\n"); + snd_printk(KERN_ERR "SB16 PnP configure failure\n"); return -EBUSY; } sbport[dev] = pnp_port_start(pdev, 0); sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + /* On CMI8239, the OPL3 port might be present in SB16 PnP resources */ + if (fmport[dev] == SNDRV_AUTO_PORT) { + if (pnp_port_start(pdev, 1)) + fmport[dev] = pnp_port_start(pdev, 1); + else + fmport[dev] = 0x388; /* Or hardwired */ + } /* allocate MPU-401 resources */ pdev = acard->mpu; err = pnp_activate_dev(pdev); - if (err < 0) { - snd_printk(KERN_ERR - "CMI8330/C3D (MPU-401) PnP configure failure\n"); - return -EBUSY; + if (err < 0) + snd_printk(KERN_ERR "MPU-401 PnP configure failure: will be disabled\n"); + else { + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); } - mpuport[dev] = pnp_port_start(pdev, 0); - mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif @@ -430,9 +448,9 @@ static int __devinit snd_cmi8330_pcm(struct snd_card *card, struct snd_cmi8330 * snd_cmi8330_capture_open }; - if ((err = snd_pcm_new(card, "CMI8330", 0, 1, 1, &pcm)) < 0) + if ((err = snd_pcm_new(card, (chip->type == CMI8329) ? "CMI8329" : "CMI8330", 0, 1, 1, &pcm)) < 0) return err; - strcpy(pcm->name, "CMI8330"); + strcpy(pcm->name, (chip->type == CMI8329) ? "CMI8329" : "CMI8330"); pcm->private_data = chip; /* SB16 */ @@ -527,11 +545,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); + snd_printk(KERN_ERR PFX "AD1848 device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); + snd_printk(KERN_ERR PFX "AD1848 not found during probe\n"); return -ENODEV; } @@ -541,11 +559,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) sbdma8[dev], sbdma16[dev], SB_HW_AUTO, &acard->sb)) < 0) { - snd_printk(KERN_ERR PFX "(SB16) device busy??\n"); + snd_printk(KERN_ERR PFX "SB16 device busy??\n"); return err; } if (acard->sb->hardware != SB_HW_16) { - snd_printk(KERN_ERR PFX "(SB16) not found during probe\n"); + snd_printk(KERN_ERR PFX "SB16 not found during probe\n"); return err; } @@ -585,8 +603,8 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) mpuport[dev]); } - strcpy(card->driver, "CMI8330/C3D"); - strcpy(card->shortname, "C-Media CMI8330/C3D"); + strcpy(card->driver, (acard->type == CMI8329) ? "CMI8329" : "CMI8330/C3D"); + strcpy(card->shortname, (acard->type == CMI8329) ? "C-Media CMI8329" : "C-Media CMI8330/C3D"); sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d", card->shortname, acard->wss->port, diff --git a/sound/oss/midibuf.c b/sound/oss/midibuf.c index a40be0cf1d97..782b3b84dac6 100644 --- a/sound/oss/midibuf.c +++ b/sound/oss/midibuf.c @@ -127,15 +127,16 @@ static void midi_poll(unsigned long dummy) for (dev = 0; dev < num_midis; dev++) if (midi_devs[dev] != NULL && midi_out_buf[dev] != NULL) { - int ok = 1; - - while (DATA_AVAIL(midi_out_buf[dev]) && ok) + while (DATA_AVAIL(midi_out_buf[dev])) { + int ok; int c = midi_out_buf[dev]->queue[midi_out_buf[dev]->head]; spin_unlock_irqrestore(&lock,flags);/* Give some time to others */ ok = midi_devs[dev]->outputc(dev, c); spin_lock_irqsave(&lock, flags); + if (!ok) + break; midi_out_buf[dev]->head = (midi_out_buf[dev]->head + 1) % MAX_QUEUE_SIZE; midi_out_buf[dev]->len--; } diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 187f72750e8f..6713110bdc75 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -628,7 +628,7 @@ static void li_setup_dma(dma_chan_t *chan, ASSERT(!(buffer_paddr & 0xFF)); chan->baseval = (buffer_paddr >> 8) | 1 << (37 - 8); - chan->cfgval = (!LI_CCFG_LOCK | + chan->cfgval = ((chan->cfgval & ~LI_CCFG_LOCK) | SHIFT_FIELD(desc->ad1843_slot, LI_CCFG_SLOT) | desc->direction | mode | @@ -638,9 +638,9 @@ static void li_setup_dma(dma_chan_t *chan, tmask = 13 - fragshift; /* See Lithium DMA Notes above. */ ASSERT(size >= 2 && size <= 7); ASSERT(tmask >= 1 && tmask <= 7); - chan->ctlval = (!LI_CCTL_RESET | + chan->ctlval = ((chan->ctlval & ~LI_CCTL_RESET) | SHIFT_FIELD(size, LI_CCTL_SIZE) | - !LI_CCTL_DMA_ENABLE | + (chan->ctlval & ~LI_CCTL_DMA_ENABLE) | SHIFT_FIELD(tmask, LI_CCTL_TMASK) | SHIFT_FIELD(0, LI_CCTL_TPTR)); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 748f6b7d90b7..fb5ee3cc3968 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -135,11 +135,11 @@ config SND_AW2 config SND_AZT3328 - tristate "Aztech AZF3328 / PCI168 (EXPERIMENTAL)" - depends on EXPERIMENTAL + tristate "Aztech AZF3328 / PCI168" select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_RAWMIDI help Say Y here to include support for Aztech AZF3328 (PCI168) soundcards. diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 76d76c08339b..b458d208720b 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -478,45 +478,6 @@ static int snd_ali_reset_5451(struct snd_ali *codec) return 0; } -#ifdef CODEC_RESET - -static int snd_ali_reset_codec(struct snd_ali *codec) -{ - struct pci_dev *pci_dev; - unsigned char bVal; - unsigned int dwVal; - unsigned short wCount, wReg; - - pci_dev = codec->pci_m1533; - - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal | 0x08000000); - udelay(5000); - pci_read_config_dword(pci_dev, 0x7c, &dwVal); - pci_write_config_dword(pci_dev, 0x7c, dwVal & 0xf7ffffff); - udelay(5000); - - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal |= 0x02; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(5000); - bVal = inb(ALI_REG(codec,ALI_SCTRL)); - bVal &= 0xfd; - outb(ALI_REG(codec,ALI_SCTRL),bVal); - udelay(15000); - - wCount = 200; - while (wCount--) { - wReg = snd_ali_codec_read(codec->ac97, AC97_POWERDOWN); - if ((wReg & 0x000f) == 0x000f) - return 0; - udelay(5000); - } - return -1; -} - -#endif - /* * ALI 5451 Controller */ @@ -561,22 +522,6 @@ static void snd_ali_disable_address_interrupt(struct snd_ali *codec) outl(gc, ALI_REG(codec, ALI_GC_CIR)); } -#if 0 /* not used */ -static void snd_ali_enable_voice_irq(struct snd_ali *codec, - unsigned int channel) -{ - unsigned int mask; - struct snd_ali_channel_control *pchregs = &(codec->chregs); - - snd_ali_printk("enable_voice_irq channel=%d\n",channel); - - mask = 1 << (channel & 0x1f); - pchregs->data.ainten = inl(ALI_REG(codec, pchregs->regs.ainten)); - pchregs->data.ainten |= mask; - outl(pchregs->data.ainten, ALI_REG(codec, pchregs->regs.ainten)); -} -#endif - static void snd_ali_disable_voice_irq(struct snd_ali *codec, unsigned int channel) { @@ -677,16 +622,6 @@ static void snd_ali_free_channel_pcm(struct snd_ali *codec, int channel) } } -#if 0 /* not used */ -static void snd_ali_start_voice(struct snd_ali *codec, unsigned int channel) -{ - unsigned int mask = 1 << (channel & 0x1f); - - snd_ali_printk("start_voice: channel=%d\n",channel); - outl(mask, ALI_REG(codec,codec->chregs.regs.start)); -} -#endif - static void snd_ali_stop_voice(struct snd_ali *codec, unsigned int channel) { unsigned int mask = 1 << (channel & 0x1f); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index f290bc56178f..8451a0169f32 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1,6 +1,6 @@ /* * azt3328.c - driver for Aztech AZF3328 based soundcards (e.g. PCI168). - * Copyright (C) 2002, 2005 - 2008 by Andreas Mohr <andi AT lisas.de> + * Copyright (C) 2002, 2005 - 2009 by Andreas Mohr <andi AT lisas.de> * * Framework borrowed from Bart Hartgers's als4000.c. * Driver developed on PCI168 AP(W) version (PCI rev. 10, subsystem ID 1801), @@ -10,6 +10,13 @@ * PCI168 A/AP, sub ID 8000 * Please give me feedback in case you try my driver with one of these!! * + * Keywords: Windows XP Vista 168nt4-125.zip 168win95-125.zip PCI 168 download + * (XP/Vista do not support this card at all but every Linux distribution + * has very good support out of the box; + * just to make sure that the right people hit this and get to know that, + * despite the high level of Internet ignorance - as usual :-P - + * about very good support for this card - on Linux!) + * * GPL LICENSE * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -71,10 +78,11 @@ * - built-in General DirectX timer having a 20 bits counter * with 1us resolution (see below!) * - I2S serial output port for external DAC + * [FIXME: 3.3V or 5V level? maximum rate is 66.2kHz right?] * - supports 33MHz PCI spec 2.1, PCI power management 1.0, compliant with ACPI * - supports hardware volume control * - single chip low cost solution (128 pin QFP) - * - supports programmable Sub-vendor and Sub-system ID + * - supports programmable Sub-vendor and Sub-system ID [24C02 SEEPROM chip] * required for Microsoft's logo compliance (FIXME: where?) * At least the Trident 4D Wave DX has one bit somewhere * to enable writes to PCI subsystem VID registers, that should be it. @@ -82,6 +90,7 @@ * some custom data starting at 0x80. What kind of config settings * are located in our extended PCI space anyway?? * - PCI168 AP(W) card: power amplifier with 4 Watts/channel at 4 Ohms + * [TDA1517P chip] * * Note that this driver now is actually *better* than the Windows driver, * since it additionally supports the card's 1MHz DirectX timer - just try @@ -146,10 +155,15 @@ * to read the Digital Enhanced Game Port. Not sure whether it is fixable. * * TODO + * - use PCI_VDEVICE + * - verify driver status on x86_64 + * - test multi-card driver operation + * - (ab)use 1MHz DirectX timer as kernel clocksource * - test MPU401 MIDI playback etc. * - add more power micro-management (disable various units of the card - * as long as they're unused). However this requires more I/O ports which I - * haven't figured out yet and which thus might not even exist... + * as long as they're unused, to improve audio quality and save power). + * However this requires more I/O ports which I haven't figured out yet + * and which thus might not even exist... * The standard suspend/resume functionality could probably make use of * some improvement, too... * - figure out what all unknown port bits are responsible for @@ -185,25 +199,46 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define SUPPORT_GAMEPORT 1 #endif +/* === Debug settings === + Further diagnostic functionality than the settings below + does not need to be provided, since one can easily write a bash script + to dump the card's I/O ports (those listed in lspci -v -v): + function dump() + { + local descr=$1; local addr=$2; local count=$3 + + echo "${descr}: ${count} @ ${addr}:" + dd if=/dev/port skip=$[${addr}] count=${count} bs=1 2>/dev/null| hexdump -C + } + and then use something like + "dump joy200 0x200 8", "dump mpu388 0x388 4", "dump joy 0xb400 8", + "dump codec00 0xa800 32", "dump mixer 0xb800 64", "dump synth 0xbc00 8", + possibly within a "while true; do ... sleep 1; done" loop. + Tweaking ports could be done using + VALSTRING="`printf "%02x" $value`" + printf "\x""$VALSTRING"|dd of=/dev/port seek=$[${addr}] bs=1 2>/dev/null +*/ + #define DEBUG_MISC 0 #define DEBUG_CALLS 0 #define DEBUG_MIXER 0 -#define DEBUG_PLAY_REC 0 +#define DEBUG_CODEC 0 #define DEBUG_IO 0 #define DEBUG_TIMER 0 #define DEBUG_GAME 0 +#define DEBUG_PM 0 #define MIXER_TESTING 0 #if DEBUG_MISC -#define snd_azf3328_dbgmisc(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgmisc(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgmisc(format, args...) #endif #if DEBUG_CALLS #define snd_azf3328_dbgcalls(format, args...) printk(format, ##args) -#define snd_azf3328_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) -#define snd_azf3328_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) +#define snd_azf3328_dbgcallenter() printk(KERN_DEBUG "--> %s\n", __func__) +#define snd_azf3328_dbgcallleave() printk(KERN_DEBUG "<-- %s\n", __func__) #else #define snd_azf3328_dbgcalls(format, args...) #define snd_azf3328_dbgcallenter() @@ -216,10 +251,10 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define snd_azf3328_dbgmixer(format, args...) #endif -#if DEBUG_PLAY_REC -#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args) +#if DEBUG_CODEC +#define snd_azf3328_dbgcodec(format, args...) printk(KERN_DEBUG format, ##args) #else -#define snd_azf3328_dbgplay(format, args...) +#define snd_azf3328_dbgcodec(format, args...) #endif #if DEBUG_MISC @@ -234,6 +269,12 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #define snd_azf3328_dbggame(format, args...) #endif +#if DEBUG_PM +#define snd_azf3328_dbgpm(format, args...) printk(KERN_DEBUG format, ##args) +#else +#define snd_azf3328_dbgpm(format, args...) +#endif + static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for AZF3328 soundcard."); @@ -250,22 +291,23 @@ static int seqtimer_scaling = 128; module_param(seqtimer_scaling, int, 0444); MODULE_PARM_DESC(seqtimer_scaling, "Set 1024000Hz sequencer timer scale factor (lockup danger!). Default 128."); -struct snd_azf3328_audio_stream { +struct snd_azf3328_codec_data { + unsigned long io_base; struct snd_pcm_substream *substream; - int enabled; - int running; - unsigned long portbase; + bool running; + const char *name; }; -enum snd_azf3328_stream_index { - AZF_PLAYBACK = 0, - AZF_CAPTURE = 1, +enum snd_azf3328_codec_type { + AZF_CODEC_PLAYBACK = 0, + AZF_CODEC_CAPTURE = 1, + AZF_CODEC_I2S_OUT = 2, }; struct snd_azf3328 { /* often-used fields towards beginning, then grouped */ - unsigned long codec_io; /* usually 0xb000, size 128 */ + unsigned long ctrl_io; /* usually 0xb000, size 128 */ unsigned long game_io; /* usually 0xb400, size 8 */ unsigned long mpu_io; /* usually 0xb800, size 4 */ unsigned long opl3_io; /* usually 0xbc00, size 8 */ @@ -275,15 +317,17 @@ struct snd_azf3328 { struct snd_timer *timer; - struct snd_pcm *pcm; - struct snd_azf3328_audio_stream audio_stream[2]; + struct snd_pcm *pcm[3]; + + /* playback, recording and I2S out codecs */ + struct snd_azf3328_codec_data codecs[3]; struct snd_card *card; struct snd_rawmidi *rmidi; #ifdef SUPPORT_GAMEPORT struct gameport *gameport; - int axes[4]; + u16 axes[4]; #endif struct pci_dev *pci; @@ -293,16 +337,16 @@ struct snd_azf3328 { * If we need to add more registers here, then we might try to fold this * into some transparent combined shadow register handling with * CONFIG_PM register storage below, but that's slightly difficult. */ - u16 shadow_reg_codec_6AH; + u16 shadow_reg_ctrl_6AH; #ifdef CONFIG_PM /* register value containers for power management - * Note: not always full I/O range preserved (just like Win driver!) */ - u16 saved_regs_codec[AZF_IO_SIZE_CODEC_PM / 2]; - u16 saved_regs_game [AZF_IO_SIZE_GAME_PM / 2]; - u16 saved_regs_mpu [AZF_IO_SIZE_MPU_PM / 2]; - u16 saved_regs_opl3 [AZF_IO_SIZE_OPL3_PM / 2]; - u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2]; + * Note: not always full I/O range preserved (similar to Win driver!) */ + u32 saved_regs_ctrl[AZF_ALIGN(AZF_IO_SIZE_CTRL_PM) / 4]; + u32 saved_regs_game[AZF_ALIGN(AZF_IO_SIZE_GAME_PM) / 4]; + u32 saved_regs_mpu[AZF_ALIGN(AZF_IO_SIZE_MPU_PM) / 4]; + u32 saved_regs_opl3[AZF_ALIGN(AZF_IO_SIZE_OPL3_PM) / 4]; + u32 saved_regs_mixer[AZF_ALIGN(AZF_IO_SIZE_MIXER_PM) / 4]; #endif }; @@ -316,7 +360,7 @@ MODULE_DEVICE_TABLE(pci, snd_azf3328_ids); static int -snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) +snd_azf3328_io_reg_setb(unsigned reg, u8 mask, bool do_set) { u8 prev = inb(reg), new; @@ -331,39 +375,72 @@ snd_azf3328_io_reg_setb(unsigned reg, u8 mask, int do_set) } static inline void -snd_azf3328_codec_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) +snd_azf3328_codec_outb(const struct snd_azf3328_codec_data *codec, + unsigned reg, + u8 value +) { - outb(value, chip->codec_io + reg); + outb(value, codec->io_base + reg); } static inline u8 -snd_azf3328_codec_inb(const struct snd_azf3328 *chip, unsigned reg) +snd_azf3328_codec_inb(const struct snd_azf3328_codec_data *codec, unsigned reg) { - return inb(chip->codec_io + reg); + return inb(codec->io_base + reg); } static inline void -snd_azf3328_codec_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) +snd_azf3328_codec_outw(const struct snd_azf3328_codec_data *codec, + unsigned reg, + u16 value +) { - outw(value, chip->codec_io + reg); + outw(value, codec->io_base + reg); } static inline u16 -snd_azf3328_codec_inw(const struct snd_azf3328 *chip, unsigned reg) +snd_azf3328_codec_inw(const struct snd_azf3328_codec_data *codec, unsigned reg) { - return inw(chip->codec_io + reg); + return inw(codec->io_base + reg); } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) +snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, + unsigned reg, + u32 value +) { - outl(value, chip->codec_io + reg); + outl(value, codec->io_base + reg); } static inline u32 -snd_azf3328_codec_inl(const struct snd_azf3328 *chip, unsigned reg) +snd_azf3328_codec_inl(const struct snd_azf3328_codec_data *codec, unsigned reg) +{ + return inl(codec->io_base + reg); +} + +static inline void +snd_azf3328_ctrl_outb(const struct snd_azf3328 *chip, unsigned reg, u8 value) +{ + outb(value, chip->ctrl_io + reg); +} + +static inline u8 +snd_azf3328_ctrl_inb(const struct snd_azf3328 *chip, unsigned reg) +{ + return inb(chip->ctrl_io + reg); +} + +static inline void +snd_azf3328_ctrl_outw(const struct snd_azf3328 *chip, unsigned reg, u16 value) +{ + outw(value, chip->ctrl_io + reg); +} + +static inline void +snd_azf3328_ctrl_outl(const struct snd_azf3328 *chip, unsigned reg, u32 value) { - return inl(chip->codec_io + reg); + outl(value, chip->ctrl_io + reg); } static inline void @@ -404,13 +481,13 @@ snd_azf3328_mixer_inw(const struct snd_azf3328 *chip, unsigned reg) #define AZF_MUTE_BIT 0x80 -static int +static bool snd_azf3328_mixer_set_mute(const struct snd_azf3328 *chip, - unsigned reg, int do_mute + unsigned reg, bool do_mute ) { unsigned long portbase = chip->mixer_io + reg + 1; - int updated; + bool updated; /* the mute bit is on the *second* (i.e. right) register of a * left/right channel setting */ @@ -569,7 +646,7 @@ snd_azf3328_get_mixer(struct snd_kcontrol *kcontrol, { struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; - unsigned int oreg, val; + u16 oreg, val; snd_azf3328_dbgcallenter(); snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); @@ -600,7 +677,7 @@ snd_azf3328_put_mixer(struct snd_kcontrol *kcontrol, { struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; - unsigned int oreg, nreg, val; + u16 oreg, nreg, val; snd_azf3328_dbgcallenter(); snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); @@ -709,7 +786,7 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol, { struct snd_azf3328 *chip = snd_kcontrol_chip(kcontrol); struct azf3328_mixer_reg reg; - unsigned int oreg, nreg, val; + u16 oreg, nreg, val; snd_azf3328_mixer_reg_decode(®, kcontrol->private_value); oreg = snd_azf3328_mixer_inw(chip, reg.reg); @@ -867,14 +944,15 @@ snd_azf3328_hw_free(struct snd_pcm_substream *substream) static void snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, - unsigned reg, + enum snd_azf3328_codec_type codec_type, enum azf_freq_t bitrate, unsigned int format_width, unsigned int channels ) { - u16 val = 0xff00; unsigned long flags; + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; + u16 val = 0xff00; snd_azf3328_dbgcallenter(); switch (bitrate) { @@ -917,7 +995,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, spin_lock_irqsave(&chip->reg_lock, flags); /* set bitrate/format */ - snd_azf3328_codec_outw(chip, reg, val); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_SOUNDFORMAT, val); /* changing the bitrate/format settings switches off the * audio output with an annoying click in case of 8/16bit format change @@ -926,11 +1004,11 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, * (FIXME: yes, it works, but what exactly am I doing here?? :) * FIXME: does this have some side effects for full-duplex * or other dramatic side effects? */ - if (reg == IDX_IO_PLAY_SOUNDFORMAT) /* only do it for playback */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | - DMA_PLAY_SOMETHING1 | - DMA_PLAY_SOMETHING2 | + if (codec_type == AZF_CODEC_PLAYBACK) /* only do it for playback */ + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS) | + DMA_RUN_SOMETHING1 | + DMA_RUN_SOMETHING2 | SOMETHING_ALMOST_ALWAYS_SET | DMA_EPILOGUE_SOMETHING | DMA_SOMETHING_ELSE @@ -942,112 +1020,134 @@ snd_azf3328_codec_setfmt(struct snd_azf3328 *chip, static inline void snd_azf3328_codec_setfmt_lowpower(struct snd_azf3328 *chip, - unsigned reg + enum snd_azf3328_codec_type codec_type ) { /* choose lowest frequency for low power consumption. * While this will cause louder noise due to rather coarse frequency, * it should never matter since output should always * get disabled properly when idle anyway. */ - snd_azf3328_codec_setfmt(chip, reg, AZF_FREQ_4000, 8, 1); + snd_azf3328_codec_setfmt(chip, codec_type, AZF_FREQ_4000, 8, 1); } static void -snd_azf3328_codec_reg_6AH_update(struct snd_azf3328 *chip, +snd_azf3328_ctrl_reg_6AH_update(struct snd_azf3328 *chip, unsigned bitmask, - int enable + bool enable ) { - if (enable) - chip->shadow_reg_codec_6AH &= ~bitmask; + bool do_mask = !enable; + if (do_mask) + chip->shadow_reg_ctrl_6AH |= bitmask; else - chip->shadow_reg_codec_6AH |= bitmask; - snd_azf3328_dbgplay("6AH_update mask 0x%04x enable %d: val 0x%04x\n", - bitmask, enable, chip->shadow_reg_codec_6AH); - snd_azf3328_codec_outw(chip, IDX_IO_6AH, chip->shadow_reg_codec_6AH); + chip->shadow_reg_ctrl_6AH &= ~bitmask; + snd_azf3328_dbgcodec("6AH_update mask 0x%04x do_mask %d: val 0x%04x\n", + bitmask, do_mask, chip->shadow_reg_ctrl_6AH); + snd_azf3328_ctrl_outw(chip, IDX_IO_6AH, chip->shadow_reg_ctrl_6AH); } static inline void -snd_azf3328_codec_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_ctrl_enable_codecs(struct snd_azf3328 *chip, bool enable) { - snd_azf3328_dbgplay("codec_enable %d\n", enable); + snd_azf3328_dbgcodec("codec_enable %d\n", enable); /* no idea what exactly is being done here, but I strongly assume it's * PM related */ - snd_azf3328_codec_reg_6AH_update( + snd_azf3328_ctrl_reg_6AH_update( chip, IO_6A_PAUSE_PLAYBACK_BIT8, enable ); } static void -snd_azf3328_codec_activity(struct snd_azf3328 *chip, - enum snd_azf3328_stream_index stream_type, - int enable +snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, + enum snd_azf3328_codec_type codec_type, + bool enable ) { - int need_change = (chip->audio_stream[stream_type].running != enable); + struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; + bool need_change = (codec->running != enable); - snd_azf3328_dbgplay( - "codec_activity: type %d, enable %d, need_change %d\n", - stream_type, enable, need_change + snd_azf3328_dbgcodec( + "codec_activity: %s codec, enable %d, need_change %d\n", + codec->name, enable, need_change ); if (need_change) { - enum snd_azf3328_stream_index other = - (stream_type == AZF_PLAYBACK) ? - AZF_CAPTURE : AZF_PLAYBACK; - /* small check to prevent shutting down the other party - * in case it's active */ - if ((enable) || !(chip->audio_stream[other].running)) - snd_azf3328_codec_enable(chip, enable); + static const struct { + enum snd_azf3328_codec_type other1; + enum snd_azf3328_codec_type other2; + } peer_codecs[3] = + { { AZF_CODEC_CAPTURE, AZF_CODEC_I2S_OUT }, + { AZF_CODEC_PLAYBACK, AZF_CODEC_I2S_OUT }, + { AZF_CODEC_PLAYBACK, AZF_CODEC_CAPTURE } }; + bool call_function; + + if (enable) + /* if enable codec, call enable_codecs func + to enable codec supply... */ + call_function = 1; + else { + /* ...otherwise call enable_codecs func + (which globally shuts down operation of codecs) + only in case the other codecs are currently + not active either! */ + call_function = + ((!chip->codecs[peer_codecs[codec_type].other1] + .running) + && (!chip->codecs[peer_codecs[codec_type].other2] + .running)); + } + if (call_function) + snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too * (reduce noise and power consumption) */ if (!enable) snd_azf3328_codec_setfmt_lowpower( chip, - chip->audio_stream[stream_type].portbase - + IDX_IO_PLAY_SOUNDFORMAT + codec_type ); + codec->running = enable; } - chip->audio_stream[stream_type].running = enable; } static void -snd_azf3328_setdmaa(struct snd_azf3328 *chip, - long unsigned int addr, - unsigned int count, - unsigned int size, - enum snd_azf3328_stream_index stream_type +snd_azf3328_codec_setdmaa(struct snd_azf3328 *chip, + enum snd_azf3328_codec_type codec_type, + unsigned long addr, + unsigned int count, + unsigned int size ) { + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; snd_azf3328_dbgcallenter(); - if (!chip->audio_stream[stream_type].running) { - /* AZF3328 uses a two buffer pointer DMA playback approach */ + if (!codec->running) { + /* AZF3328 uses a two buffer pointer DMA transfer approach */ - unsigned long flags, portbase, addr_area2; + unsigned long flags, addr_area2; /* width 32bit (prevent overflow): */ - unsigned long count_areas, count_tmp; + u32 count_areas, lengths; - portbase = chip->audio_stream[stream_type].portbase; count_areas = size/2; addr_area2 = addr+count_areas; count_areas--; /* max. index */ - snd_azf3328_dbgplay("set DMA: buf1 %08lx[%lu], buf2 %08lx[%lu]\n", addr, count_areas, addr_area2, count_areas); + snd_azf3328_dbgcodec("setdma: buffers %08lx[%u] / %08lx[%u]\n", + addr, count_areas, addr_area2, count_areas); /* build combined I/O buffer length word */ - count_tmp = count_areas; - count_areas |= (count_tmp << 16); + lengths = (count_areas << 16) | (count_areas); spin_lock_irqsave(&chip->reg_lock, flags); - outl(addr, portbase + IDX_IO_PLAY_DMA_START_1); - outl(addr_area2, portbase + IDX_IO_PLAY_DMA_START_2); - outl(count_areas, portbase + IDX_IO_PLAY_DMA_LEN_1); + snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_1, addr); + snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_START_2, + addr_area2); + snd_azf3328_codec_outl(codec, IDX_IO_CODEC_DMA_LENGTHS, + lengths); spin_unlock_irqrestore(&chip->reg_lock, flags); } snd_azf3328_dbgcallleave(); } static int -snd_azf3328_playback_prepare(struct snd_pcm_substream *substream) +snd_azf3328_codec_prepare(struct snd_pcm_substream *substream) { #if 0 struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); @@ -1058,157 +1158,161 @@ snd_azf3328_playback_prepare(struct snd_pcm_substream *substream) snd_azf3328_dbgcallenter(); #if 0 - snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, AZF_CODEC_..., runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_PLAYBACK); + snd_azf3328_codec_setdmaa(chip, AZF_CODEC_..., + runtime->dma_addr, count, size); #endif snd_azf3328_dbgcallleave(); return 0; } static int -snd_azf3328_capture_prepare(struct snd_pcm_substream *substream) -{ -#if 0 - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - unsigned int count = snd_pcm_lib_period_bytes(substream); -#endif - - snd_azf3328_dbgcallenter(); -#if 0 - snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, - runtime->rate, - snd_pcm_format_width(runtime->format), - runtime->channels); - snd_azf3328_setdmaa(chip, runtime->dma_addr, count, size, AZF_CAPTURE); -#endif - snd_azf3328_dbgcallleave(); - return 0; -} - -static int -snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) +snd_azf3328_codec_trigger(enum snd_azf3328_codec_type codec_type, + struct snd_pcm_substream *substream, int cmd) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; struct snd_pcm_runtime *runtime = substream->runtime; int result = 0; - unsigned int status1; - int previously_muted; + u16 flags1; + bool previously_muted = 0; + bool is_playback_codec = (AZF_CODEC_PLAYBACK == codec_type); - snd_azf3328_dbgcalls("snd_azf3328_playback_trigger cmd %d\n", cmd); + snd_azf3328_dbgcalls("snd_azf3328_codec_trigger cmd %d\n", cmd); switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_azf3328_dbgplay("START PLAYBACK\n"); - - /* mute WaveOut (avoid clicking during setup) */ - previously_muted = - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + snd_azf3328_dbgcodec("START %s\n", codec->name); + + if (is_playback_codec) { + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 1 + ); + } - snd_azf3328_codec_setfmt(chip, IDX_IO_PLAY_SOUNDFORMAT, + snd_azf3328_codec_setfmt(chip, codec_type, runtime->rate, snd_pcm_format_width(runtime->format), runtime->channels); spin_lock(&chip->reg_lock); /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); - /* stop playback */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + /* stop transfer */ + flags1 &= ~DMA_RESUME; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); /* FIXME: clear interrupts or what??? */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_IRQTYPE, 0xffff); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_IRQTYPE, 0xffff); spin_unlock(&chip->reg_lock); - snd_azf3328_setdmaa(chip, runtime->dma_addr, + snd_azf3328_codec_setdmaa(chip, codec_type, runtime->dma_addr, snd_pcm_lib_period_bytes(substream), - snd_pcm_lib_buffer_bytes(substream), - AZF_PLAYBACK); + snd_pcm_lib_buffer_bytes(substream) + ); spin_lock(&chip->reg_lock); #ifdef WIN9X /* FIXME: enable playback/recording??? */ - status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 |= DMA_RUN_SOMETHING1 | DMA_RUN_SOMETHING2; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); - /* start playback again */ + /* start transfer again */ /* FIXME: what is this value (0x0010)??? */ - status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); #else /* NT4 */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, 0x0000); - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - DMA_PLAY_SOMETHING1); - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - DMA_PLAY_SOMETHING1 | - DMA_PLAY_SOMETHING2); - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + DMA_RUN_SOMETHING1); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + DMA_RUN_SOMETHING1 | + DMA_RUN_SOMETHING2); + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, DMA_RESUME | SOMETHING_ALMOST_ALWAYS_SET | DMA_EPILOGUE_SOMETHING | DMA_SOMETHING_ELSE); #endif spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 1); - - /* now unmute WaveOut */ - if (!previously_muted) - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_ctrl_codec_activity(chip, codec_type, 1); + + if (is_playback_codec) { + /* now unmute WaveOut */ + if (!previously_muted) + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 0 + ); + } - snd_azf3328_dbgplay("STARTED PLAYBACK\n"); + snd_azf3328_dbgcodec("STARTED %s\n", codec->name); break; case SNDRV_PCM_TRIGGER_RESUME: - snd_azf3328_dbgplay("RESUME PLAYBACK\n"); - /* resume playback if we were active */ + snd_azf3328_dbgcodec("RESUME %s\n", codec->name); + /* resume codec if we were active */ spin_lock(&chip->reg_lock); - if (chip->audio_stream[AZF_PLAYBACK].running) - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME); + if (codec->running) + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + snd_azf3328_codec_inw( + codec, IDX_IO_CODEC_DMA_FLAGS + ) | DMA_RESUME + ); spin_unlock(&chip->reg_lock); break; case SNDRV_PCM_TRIGGER_STOP: - snd_azf3328_dbgplay("STOP PLAYBACK\n"); - - /* mute WaveOut (avoid clicking during setup) */ - previously_muted = - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); + snd_azf3328_dbgcodec("STOP %s\n", codec->name); + + if (is_playback_codec) { + /* mute WaveOut (avoid clicking during setup) */ + previously_muted = + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 1 + ); + } spin_lock(&chip->reg_lock); /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS); + flags1 = snd_azf3328_codec_inw(codec, IDX_IO_CODEC_DMA_FLAGS); - /* stop playback */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + /* stop transfer */ + flags1 &= ~DMA_RESUME; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); /* hmm, is this really required? we're resetting the same bit * immediately thereafter... */ - status1 |= DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 |= DMA_RUN_SOMETHING1; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); - status1 &= ~DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, status1); + flags1 &= ~DMA_RUN_SOMETHING1; + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, flags1); spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); - - /* now unmute WaveOut */ - if (!previously_muted) - snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 0); + snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); + + if (is_playback_codec) { + /* now unmute WaveOut */ + if (!previously_muted) + snd_azf3328_mixer_set_mute( + chip, IDX_MIXER_WAVEOUT, 0 + ); + } - snd_azf3328_dbgplay("STOPPED PLAYBACK\n"); + snd_azf3328_dbgcodec("STOPPED %s\n", codec->name); break; case SNDRV_PCM_TRIGGER_SUSPEND: - snd_azf3328_dbgplay("SUSPEND PLAYBACK\n"); - /* make sure playback is stopped */ - snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME); + snd_azf3328_dbgcodec("SUSPEND %s\n", codec->name); + /* make sure codec is stopped */ + snd_azf3328_codec_outw(codec, IDX_IO_CODEC_DMA_FLAGS, + snd_azf3328_codec_inw( + codec, IDX_IO_CODEC_DMA_FLAGS + ) & ~DMA_RESUME + ); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); @@ -1217,7 +1321,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); break; default: - printk(KERN_ERR "FIXME: unknown trigger mode!\n"); + snd_printk(KERN_ERR "FIXME: unknown trigger mode!\n"); return -EINVAL; } @@ -1225,172 +1329,74 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd) return result; } -/* this is just analogous to playback; I'm not quite sure whether recording - * should actually be triggered like that */ static int -snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd) +snd_azf3328_codec_playback_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int result = 0; - unsigned int status1; - - snd_azf3328_dbgcalls("snd_azf3328_capture_trigger cmd %d\n", cmd); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - - snd_azf3328_dbgplay("START CAPTURE\n"); - - snd_azf3328_codec_setfmt(chip, IDX_IO_REC_SOUNDFORMAT, - runtime->rate, - snd_pcm_format_width(runtime->format), - runtime->channels); - - spin_lock(&chip->reg_lock); - /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); - - /* stop recording */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - /* FIXME: clear interrupts or what??? */ - snd_azf3328_codec_outw(chip, IDX_IO_REC_IRQTYPE, 0xffff); - spin_unlock(&chip->reg_lock); - - snd_azf3328_setdmaa(chip, runtime->dma_addr, - snd_pcm_lib_period_bytes(substream), - snd_pcm_lib_buffer_bytes(substream), - AZF_CAPTURE); - - spin_lock(&chip->reg_lock); -#ifdef WIN9X - /* FIXME: enable playback/recording??? */ - status1 |= DMA_PLAY_SOMETHING1 | DMA_PLAY_SOMETHING2; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - /* start capture again */ - /* FIXME: what is this value (0x0010)??? */ - status1 |= DMA_RESUME | DMA_EPILOGUE_SOMETHING; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); -#else - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - 0x0000); - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - DMA_PLAY_SOMETHING1); - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - DMA_PLAY_SOMETHING1 | - DMA_PLAY_SOMETHING2); - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - DMA_RESUME | - SOMETHING_ALMOST_ALWAYS_SET | - DMA_EPILOGUE_SOMETHING | - DMA_SOMETHING_ELSE); -#endif - spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_CAPTURE, 1); - - snd_azf3328_dbgplay("STARTED CAPTURE\n"); - break; - case SNDRV_PCM_TRIGGER_RESUME: - snd_azf3328_dbgplay("RESUME CAPTURE\n"); - /* resume recording if we were active */ - spin_lock(&chip->reg_lock); - if (chip->audio_stream[AZF_CAPTURE].running) - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME); - spin_unlock(&chip->reg_lock); - break; - case SNDRV_PCM_TRIGGER_STOP: - snd_azf3328_dbgplay("STOP CAPTURE\n"); - - spin_lock(&chip->reg_lock); - /* first, remember current value: */ - status1 = snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS); - - /* stop recording */ - status1 &= ~DMA_RESUME; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - status1 |= DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - - status1 &= ~DMA_PLAY_SOMETHING1; - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, status1); - spin_unlock(&chip->reg_lock); - snd_azf3328_codec_activity(chip, AZF_CAPTURE, 0); + return snd_azf3328_codec_trigger(AZF_CODEC_PLAYBACK, substream, cmd); +} - snd_azf3328_dbgplay("STOPPED CAPTURE\n"); - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - snd_azf3328_dbgplay("SUSPEND CAPTURE\n"); - /* make sure recording is stopped */ - snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS, - snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME); - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n"); - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n"); - break; - default: - printk(KERN_ERR "FIXME: unknown trigger mode!\n"); - return -EINVAL; - } +static int +snd_azf3328_codec_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + return snd_azf3328_codec_trigger(AZF_CODEC_CAPTURE, substream, cmd); +} - snd_azf3328_dbgcallleave(); - return result; +static int +snd_azf3328_codec_i2s_out_trigger(struct snd_pcm_substream *substream, int cmd) +{ + return snd_azf3328_codec_trigger(AZF_CODEC_I2S_OUT, substream, cmd); } static snd_pcm_uframes_t -snd_azf3328_playback_pointer(struct snd_pcm_substream *substream) +snd_azf3328_codec_pointer(struct snd_pcm_substream *substream, + enum snd_azf3328_codec_type codec_type +) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + const struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + const struct snd_azf3328_codec_data *codec = &chip->codecs[codec_type]; unsigned long bufptr, result; snd_pcm_uframes_t frmres; #ifdef QUERY_HARDWARE - bufptr = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_START_1); + bufptr = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_START_1); #else bufptr = substream->runtime->dma_addr; #endif - result = snd_azf3328_codec_inl(chip, IDX_IO_PLAY_DMA_CURRPOS); + result = snd_azf3328_codec_inl(codec, IDX_IO_CODEC_DMA_CURRPOS); /* calculate offset */ result -= bufptr; frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgplay("PLAY @ 0x%8lx, frames %8ld\n", result, frmres); + snd_azf3328_dbgcodec("%s @ 0x%8lx, frames %8ld\n", + codec->name, result, frmres); return frmres; } static snd_pcm_uframes_t -snd_azf3328_capture_pointer(struct snd_pcm_substream *substream) +snd_azf3328_codec_playback_pointer(struct snd_pcm_substream *substream) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - unsigned long bufptr, result; - snd_pcm_uframes_t frmres; + return snd_azf3328_codec_pointer(substream, AZF_CODEC_PLAYBACK); +} -#ifdef QUERY_HARDWARE - bufptr = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_START_1); -#else - bufptr = substream->runtime->dma_addr; -#endif - result = snd_azf3328_codec_inl(chip, IDX_IO_REC_DMA_CURRPOS); +static snd_pcm_uframes_t +snd_azf3328_codec_capture_pointer(struct snd_pcm_substream *substream) +{ + return snd_azf3328_codec_pointer(substream, AZF_CODEC_CAPTURE); +} - /* calculate offset */ - result -= bufptr; - frmres = bytes_to_frames( substream->runtime, result); - snd_azf3328_dbgplay("REC @ 0x%8lx, frames %8ld\n", result, frmres); - return frmres; +static snd_pcm_uframes_t +snd_azf3328_codec_i2s_out_pointer(struct snd_pcm_substream *substream) +{ + return snd_azf3328_codec_pointer(substream, AZF_CODEC_I2S_OUT); } /******************************************************************/ #ifdef SUPPORT_GAMEPORT static inline void -snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, + bool enable +) { snd_azf3328_io_reg_setb( chip->game_io+IDX_GAME_HWCONFIG, @@ -1400,7 +1406,9 @@ snd_azf3328_gameport_irq_enable(struct snd_azf3328 *chip, int enable) } static inline void -snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, + bool enable +) { snd_azf3328_io_reg_setb( chip->game_io+IDX_GAME_HWCONFIG, @@ -1409,10 +1417,27 @@ snd_azf3328_gameport_legacy_address_enable(struct snd_azf3328 *chip, int enable) ); } +static void +snd_azf3328_gameport_set_counter_frequency(struct snd_azf3328 *chip, + unsigned int freq_cfg +) +{ + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + 0x02, + (freq_cfg & 1) != 0 + ); + snd_azf3328_io_reg_setb( + chip->game_io+IDX_GAME_HWCONFIG, + 0x04, + (freq_cfg & 2) != 0 + ); +} + static inline void -snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, int enable) +snd_azf3328_gameport_axis_circuit_enable(struct snd_azf3328 *chip, bool enable) { - snd_azf3328_codec_reg_6AH_update( + snd_azf3328_ctrl_reg_6AH_update( chip, IO_6A_SOMETHING2_GAMEPORT, enable ); } @@ -1447,6 +1472,8 @@ snd_azf3328_gameport_open(struct gameport *gameport, int mode) break; } + snd_azf3328_gameport_set_counter_frequency(chip, + GAME_HWCFG_ADC_COUNTER_FREQ_STD); snd_azf3328_gameport_axis_circuit_enable(chip, (res == 0)); return res; @@ -1458,6 +1485,8 @@ snd_azf3328_gameport_close(struct gameport *gameport) struct snd_azf3328 *chip = gameport_get_port_data(gameport); snd_azf3328_dbggame("gameport_close\n"); + snd_azf3328_gameport_set_counter_frequency(chip, + GAME_HWCFG_ADC_COUNTER_FREQ_1_200); snd_azf3328_gameport_axis_circuit_enable(chip, 0); } @@ -1491,7 +1520,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport, val = snd_azf3328_game_inb(chip, IDX_GAME_AXES_CONFIG); if (val & GAME_AXES_SAMPLING_READY) { - for (i = 0; i < 4; ++i) { + for (i = 0; i < ARRAY_SIZE(chip->axes); ++i) { /* configure the axis to read */ val = (i << 4) | 0x0f; snd_azf3328_game_outb(chip, IDX_GAME_AXES_CONFIG, val); @@ -1514,7 +1543,7 @@ snd_azf3328_gameport_cooked_read(struct gameport *gameport, snd_azf3328_game_outw(chip, IDX_GAME_AXIS_VALUE, 0xffff); spin_unlock_irqrestore(&chip->reg_lock, flags); - for (i = 0; i < 4; i++) { + for (i = 0; i < ARRAY_SIZE(chip->axes); i++) { axes[i] = chip->axes[i]; if (axes[i] == 0xffff) axes[i] = -1; @@ -1552,6 +1581,8 @@ snd_azf3328_gameport(struct snd_azf3328 *chip, int dev) /* DISABLE legacy address: we don't need it! */ snd_azf3328_gameport_legacy_address_enable(chip, 0); + snd_azf3328_gameport_set_counter_frequency(chip, + GAME_HWCFG_ADC_COUNTER_FREQ_1_200); snd_azf3328_gameport_axis_circuit_enable(chip, 0); gameport_register_port(chip->gameport); @@ -1585,40 +1616,77 @@ snd_azf3328_gameport_interrupt(struct snd_azf3328 *chip) static inline void snd_azf3328_irq_log_unknown_type(u8 which) { - snd_azf3328_dbgplay( + snd_azf3328_dbgcodec( "azt3328: unknown IRQ type (%x) occurred, please report!\n", which ); } +static inline void +snd_azf3328_codec_interrupt(struct snd_azf3328 *chip, u8 status) +{ + u8 which; + enum snd_azf3328_codec_type codec_type; + const struct snd_azf3328_codec_data *codec; + + for (codec_type = AZF_CODEC_PLAYBACK; + codec_type <= AZF_CODEC_I2S_OUT; + ++codec_type) { + + /* skip codec if there's no interrupt for it */ + if (!(status & (1 << codec_type))) + continue; + + codec = &chip->codecs[codec_type]; + + spin_lock(&chip->reg_lock); + which = snd_azf3328_codec_inb(codec, IDX_IO_CODEC_IRQTYPE); + /* ack all IRQ types immediately */ + snd_azf3328_codec_outb(codec, IDX_IO_CODEC_IRQTYPE, which); + spin_unlock(&chip->reg_lock); + + if ((chip->pcm[codec_type]) && (codec->substream)) { + snd_pcm_period_elapsed(codec->substream); + snd_azf3328_dbgcodec("%s period done (#%x), @ %x\n", + codec->name, + which, + snd_azf3328_codec_inl( + codec, IDX_IO_CODEC_DMA_CURRPOS + ) + ); + } else + printk(KERN_WARNING "azt3328: irq handler problem!\n"); + if (which & IRQ_SOMETHING) + snd_azf3328_irq_log_unknown_type(which); + } +} + static irqreturn_t snd_azf3328_interrupt(int irq, void *dev_id) { struct snd_azf3328 *chip = dev_id; - u8 status, which; -#if DEBUG_PLAY_REC + u8 status; +#if DEBUG_CODEC static unsigned long irq_count; #endif - status = snd_azf3328_codec_inb(chip, IDX_IO_IRQSTATUS); + status = snd_azf3328_ctrl_inb(chip, IDX_IO_IRQSTATUS); /* fast path out, to ease interrupt sharing */ if (!(status & - (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) + (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT + |IRQ_GAMEPORT|IRQ_MPU401|IRQ_TIMER) )) return IRQ_NONE; /* must be interrupt for another device */ - snd_azf3328_dbgplay( - "irq_count %ld! IDX_IO_PLAY_FLAGS %04x, " - "IDX_IO_PLAY_IRQTYPE %04x, IDX_IO_IRQSTATUS %04x\n", + snd_azf3328_dbgcodec( + "irq_count %ld! IDX_IO_IRQSTATUS %04x\n", irq_count++ /* debug-only */, - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS), - snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE), status ); if (status & IRQ_TIMER) { - /* snd_azf3328_dbgplay("timer %ld\n", + /* snd_azf3328_dbgcodec("timer %ld\n", snd_azf3328_codec_inl(chip, IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK ); */ @@ -1626,71 +1694,36 @@ snd_azf3328_interrupt(int irq, void *dev_id) snd_timer_interrupt(chip->timer, chip->timer->sticks); /* ACK timer */ spin_lock(&chip->reg_lock); - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07); + snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0x07); spin_unlock(&chip->reg_lock); - snd_azf3328_dbgplay("azt3328: timer IRQ\n"); + snd_azf3328_dbgcodec("azt3328: timer IRQ\n"); } - if (status & IRQ_PLAYBACK) { - spin_lock(&chip->reg_lock); - which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE); - /* ack all IRQ types immediately */ - snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which); - spin_unlock(&chip->reg_lock); - if (chip->pcm && chip->audio_stream[AZF_PLAYBACK].substream) { - snd_pcm_period_elapsed( - chip->audio_stream[AZF_PLAYBACK].substream - ); - snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n", - which, - snd_azf3328_codec_inl( - chip, IDX_IO_PLAY_DMA_CURRPOS - ) - ); - } else - printk(KERN_WARNING "azt3328: irq handler problem!\n"); - if (which & IRQ_PLAY_SOMETHING) - snd_azf3328_irq_log_unknown_type(which); - } - if (status & IRQ_RECORDING) { - spin_lock(&chip->reg_lock); - which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE); - /* ack all IRQ types immediately */ - snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which); - spin_unlock(&chip->reg_lock); + if (status & (IRQ_PLAYBACK|IRQ_RECORDING|IRQ_I2S_OUT)) + snd_azf3328_codec_interrupt(chip, status); - if (chip->pcm && chip->audio_stream[AZF_CAPTURE].substream) { - snd_pcm_period_elapsed( - chip->audio_stream[AZF_CAPTURE].substream - ); - snd_azf3328_dbgplay("REC period done (#%x), @ %x\n", - which, - snd_azf3328_codec_inl( - chip, IDX_IO_REC_DMA_CURRPOS - ) - ); - } else - printk(KERN_WARNING "azt3328: irq handler problem!\n"); - if (which & IRQ_REC_SOMETHING) - snd_azf3328_irq_log_unknown_type(which); - } if (status & IRQ_GAMEPORT) snd_azf3328_gameport_interrupt(chip); + /* MPU401 has less critical IRQ requirements * than timer and playback/recording, right? */ if (status & IRQ_MPU401) { snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data); /* hmm, do we have to ack the IRQ here somehow? - * If so, then I don't know how... */ - snd_azf3328_dbgplay("azt3328: MPU401 IRQ\n"); + * If so, then I don't know how yet... */ + snd_azf3328_dbgcodec("azt3328: MPU401 IRQ\n"); } return IRQ_HANDLED; } /*****************************************************************/ -static const struct snd_pcm_hardware snd_azf3328_playback = +/* as long as we think we have identical snd_pcm_hardware parameters + for playback, capture and i2s out, we can use the same physical struct + since the struct is simply being copied into a member. +*/ +static const struct snd_pcm_hardware snd_azf3328_hardware = { /* FIXME!! Correct? */ .info = SNDRV_PCM_INFO_MMAP | @@ -1718,31 +1751,6 @@ static const struct snd_pcm_hardware snd_azf3328_playback = .fifo_size = 0, }; -static const struct snd_pcm_hardware snd_azf3328_capture = -{ - /* FIXME */ - .info = SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID, - .formats = SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE, - .rates = SNDRV_PCM_RATE_5512 | - SNDRV_PCM_RATE_8000_48000 | - SNDRV_PCM_RATE_KNOT, - .rate_min = AZF_FREQ_4000, - .rate_max = AZF_FREQ_66200, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = 65536, - .period_bytes_min = 64, - .period_bytes_max = 65536, - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - static unsigned int snd_azf3328_fixed_rates[] = { AZF_FREQ_4000, @@ -1770,14 +1778,19 @@ static struct snd_pcm_hw_constraint_list snd_azf3328_hw_constraints_rates = { /*****************************************************************/ static int -snd_azf3328_playback_open(struct snd_pcm_substream *substream) +snd_azf3328_pcm_open(struct snd_pcm_substream *substream, + enum snd_azf3328_codec_type codec_type +) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_PLAYBACK].substream = substream; - runtime->hw = snd_azf3328_playback; + chip->codecs[codec_type].substream = substream; + + /* same parameters for all our codecs - at least we think so... */ + runtime->hw = snd_azf3328_hardware; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &snd_azf3328_hw_constraints_rates); snd_azf3328_dbgcallleave(); @@ -1785,40 +1798,52 @@ snd_azf3328_playback_open(struct snd_pcm_substream *substream) } static int +snd_azf3328_playback_open(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_open(substream, AZF_CODEC_PLAYBACK); +} + +static int snd_azf3328_capture_open(struct snd_pcm_substream *substream) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; + return snd_azf3328_pcm_open(substream, AZF_CODEC_CAPTURE); +} - snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_CAPTURE].substream = substream; - runtime->hw = snd_azf3328_capture; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &snd_azf3328_hw_constraints_rates); - snd_azf3328_dbgcallleave(); - return 0; +static int +snd_azf3328_i2s_out_open(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_open(substream, AZF_CODEC_I2S_OUT); } static int -snd_azf3328_playback_close(struct snd_pcm_substream *substream) +snd_azf3328_pcm_close(struct snd_pcm_substream *substream, + enum snd_azf3328_codec_type codec_type +) { struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_PLAYBACK].substream = NULL; + chip->codecs[codec_type].substream = NULL; snd_azf3328_dbgcallleave(); return 0; } static int +snd_azf3328_playback_close(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_close(substream, AZF_CODEC_PLAYBACK); +} + +static int snd_azf3328_capture_close(struct snd_pcm_substream *substream) { - struct snd_azf3328 *chip = snd_pcm_substream_chip(substream); + return snd_azf3328_pcm_close(substream, AZF_CODEC_CAPTURE); +} - snd_azf3328_dbgcallenter(); - chip->audio_stream[AZF_CAPTURE].substream = NULL; - snd_azf3328_dbgcallleave(); - return 0; +static int +snd_azf3328_i2s_out_close(struct snd_pcm_substream *substream) +{ + return snd_azf3328_pcm_close(substream, AZF_CODEC_I2S_OUT); } /******************************************************************/ @@ -1829,9 +1854,9 @@ static struct snd_pcm_ops snd_azf3328_playback_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_playback_prepare, - .trigger = snd_azf3328_playback_trigger, - .pointer = snd_azf3328_playback_pointer + .prepare = snd_azf3328_codec_prepare, + .trigger = snd_azf3328_codec_playback_trigger, + .pointer = snd_azf3328_codec_playback_pointer }; static struct snd_pcm_ops snd_azf3328_capture_ops = { @@ -1840,30 +1865,67 @@ static struct snd_pcm_ops snd_azf3328_capture_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_azf3328_hw_params, .hw_free = snd_azf3328_hw_free, - .prepare = snd_azf3328_capture_prepare, - .trigger = snd_azf3328_capture_trigger, - .pointer = snd_azf3328_capture_pointer + .prepare = snd_azf3328_codec_prepare, + .trigger = snd_azf3328_codec_capture_trigger, + .pointer = snd_azf3328_codec_capture_pointer +}; + +static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { + .open = snd_azf3328_i2s_out_open, + .close = snd_azf3328_i2s_out_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_azf3328_hw_params, + .hw_free = snd_azf3328_hw_free, + .prepare = snd_azf3328_codec_prepare, + .trigger = snd_azf3328_codec_i2s_out_trigger, + .pointer = snd_azf3328_codec_i2s_out_pointer }; static int __devinit -snd_azf3328_pcm(struct snd_azf3328 *chip, int device) +snd_azf3328_pcm(struct snd_azf3328 *chip) { +enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + struct snd_pcm *pcm; int err; snd_azf3328_dbgcallenter(); - if ((err = snd_pcm_new(chip->card, "AZF3328 DSP", device, 1, 1, &pcm)) < 0) + + err = snd_pcm_new(chip->card, "AZF3328 DSP", AZF_PCMDEV_STD, + 1, 1, &pcm); + if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_azf3328_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_azf3328_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_azf3328_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_azf3328_capture_ops); pcm->private_data = chip; pcm->info_flags = 0; strcpy(pcm->name, chip->card->shortname); - chip->pcm = pcm; + /* same pcm object for playback/capture (see snd_pcm_new() above) */ + chip->pcm[AZF_CODEC_PLAYBACK] = pcm; + chip->pcm[AZF_CODEC_CAPTURE] = pcm; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_pci_data(chip->pci), 64*1024, 64*1024); + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); + + err = snd_pcm_new(chip->card, "AZF3328 I2S OUT", AZF_PCMDEV_I2S_OUT, + 1, 0, &pcm); + if (err < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_azf3328_i2s_out_ops); + + pcm->private_data = chip; + pcm->info_flags = 0; + strcpy(pcm->name, chip->card->shortname); + chip->pcm[AZF_CODEC_I2S_OUT] = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(chip->pci), + 64*1024, 64*1024); snd_azf3328_dbgcallleave(); return 0; @@ -1902,7 +1964,7 @@ snd_azf3328_timer_start(struct snd_timer *timer) snd_azf3328_dbgtimer("setting timer countdown value %d, add COUNTDOWN|IRQ\n", delay); delay |= TIMER_COUNTDOWN_ENABLE | TIMER_IRQ_ENABLE; spin_lock_irqsave(&chip->reg_lock, flags); - snd_azf3328_codec_outl(chip, IDX_IO_TIMER_VALUE, delay); + snd_azf3328_ctrl_outl(chip, IDX_IO_TIMER_VALUE, delay); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); return 0; @@ -1919,7 +1981,7 @@ snd_azf3328_timer_stop(struct snd_timer *timer) spin_lock_irqsave(&chip->reg_lock, flags); /* disable timer countdown and interrupt */ /* FIXME: should we write TIMER_IRQ_ACK here? */ - snd_azf3328_codec_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); + snd_azf3328_ctrl_outb(chip, IDX_IO_TIMER_VALUE + 3, 0); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_azf3328_dbgcallleave(); return 0; @@ -2035,7 +2097,7 @@ snd_azf3328_test_bit(unsigned unsigned reg, int bit) outb(val, reg); - printk(KERN_ERR "reg %04x bit %d: %02x %02x %02x\n", + printk(KERN_DEBUG "reg %04x bit %d: %02x %02x %02x\n", reg, bit, val, valoff, valon ); } @@ -2048,9 +2110,9 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) u16 tmp; snd_azf3328_dbgmisc( - "codec_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " + "ctrl_io 0x%lx, game_io 0x%lx, mpu_io 0x%lx, " "opl3_io 0x%lx, mixer_io 0x%lx, irq %d\n", - chip->codec_io, chip->game_io, chip->mpu_io, + chip->ctrl_io, chip->game_io, chip->mpu_io, chip->opl3_io, chip->mixer_io, chip->irq ); @@ -2083,9 +2145,9 @@ snd_azf3328_debug_show_ports(const struct snd_azf3328 *chip) inb(0x38c + tmp) ); - for (tmp = 0; tmp < AZF_IO_SIZE_CODEC; tmp += 2) - snd_azf3328_dbgmisc("codec 0x%02x: 0x%04x\n", - tmp, snd_azf3328_codec_inw(chip, tmp) + for (tmp = 0; tmp < AZF_IO_SIZE_CTRL; tmp += 2) + snd_azf3328_dbgmisc("ctrl 0x%02x: 0x%04x\n", + tmp, snd_azf3328_ctrl_inw(chip, tmp) ); for (tmp = 0; tmp < AZF_IO_SIZE_MIXER; tmp += 2) @@ -2106,7 +2168,8 @@ snd_azf3328_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_azf3328_dev_free, }; - u16 tmp; + u8 dma_init; + enum snd_azf3328_codec_type codec_type; *rchip = NULL; @@ -2138,14 +2201,21 @@ snd_azf3328_create(struct snd_card *card, if (err < 0) goto out_err; - chip->codec_io = pci_resource_start(pci, 0); + chip->ctrl_io = pci_resource_start(pci, 0); chip->game_io = pci_resource_start(pci, 1); chip->mpu_io = pci_resource_start(pci, 2); - chip->opl3_io = pci_resource_start(pci, 3); + chip->opl3_io = pci_resource_start(pci, 3); chip->mixer_io = pci_resource_start(pci, 4); - chip->audio_stream[AZF_PLAYBACK].portbase = chip->codec_io + 0x00; - chip->audio_stream[AZF_CAPTURE].portbase = chip->codec_io + 0x20; + chip->codecs[AZF_CODEC_PLAYBACK].io_base = + chip->ctrl_io + AZF_IO_OFFS_CODEC_PLAYBACK; + chip->codecs[AZF_CODEC_PLAYBACK].name = "PLAYBACK"; + chip->codecs[AZF_CODEC_CAPTURE].io_base = + chip->ctrl_io + AZF_IO_OFFS_CODEC_CAPTURE; + chip->codecs[AZF_CODEC_CAPTURE].name = "CAPTURE"; + chip->codecs[AZF_CODEC_I2S_OUT].io_base = + chip->ctrl_io + AZF_IO_OFFS_CODEC_I2S_OUT; + chip->codecs[AZF_CODEC_I2S_OUT].name = "I2S_OUT"; if (request_irq(pci->irq, snd_azf3328_interrupt, IRQF_SHARED, card->shortname, chip)) { @@ -2168,20 +2238,25 @@ snd_azf3328_create(struct snd_card *card, if (err < 0) goto out_err; - /* shutdown codecs to save power */ - /* have snd_azf3328_codec_activity() act properly */ - chip->audio_stream[AZF_PLAYBACK].running = 1; - snd_azf3328_codec_activity(chip, AZF_PLAYBACK, 0); + /* standard codec init stuff */ + /* default DMA init value */ + dma_init = DMA_RUN_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; - /* standard chip init stuff */ - /* default IRQ init value */ - tmp = DMA_PLAY_SOMETHING2|DMA_EPILOGUE_SOMETHING|DMA_SOMETHING_ELSE; + for (codec_type = AZF_CODEC_PLAYBACK; + codec_type <= AZF_CODEC_I2S_OUT; ++codec_type) { + struct snd_azf3328_codec_data *codec = + &chip->codecs[codec_type]; - spin_lock_irq(&chip->reg_lock); - snd_azf3328_codec_outb(chip, IDX_IO_PLAY_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_REC_FLAGS, tmp); - snd_azf3328_codec_outb(chip, IDX_IO_SOMETHING_FLAGS, tmp); - spin_unlock_irq(&chip->reg_lock); + /* shutdown codecs to save power */ + /* have ...ctrl_codec_activity() act properly */ + codec->running = 1; + snd_azf3328_ctrl_codec_activity(chip, codec_type, 0); + + spin_lock_irq(&chip->reg_lock); + snd_azf3328_codec_outb(codec, IDX_IO_CODEC_DMA_FLAGS, + dma_init); + spin_unlock_irq(&chip->reg_lock); + } snd_card_set_dev(card, &pci->dev); @@ -2229,8 +2304,11 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) card->private_data = chip; + /* chose to use MPU401_HW_AZT2320 ID instead of MPU401_HW_MPU401, + since our hardware ought to be similar, thus use same ID. */ err = snd_mpu401_uart_new( - card, 0, MPU401_HW_MPU401, chip->mpu_io, MPU401_INFO_INTEGRATED, + card, 0, + MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED, pci->irq, 0, &chip->rmidi ); if (err < 0) { @@ -2244,7 +2322,7 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) if (err < 0) goto out_err; - err = snd_azf3328_pcm(chip, 0); + err = snd_azf3328_pcm(chip); if (err < 0) goto out_err; @@ -2266,14 +2344,14 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) opl3->private_data = chip; sprintf(card->longname, "%s at 0x%lx, irq %i", - card->shortname, chip->codec_io, chip->irq); + card->shortname, chip->ctrl_io, chip->irq); err = snd_card_register(card); if (err < 0) goto out_err; #ifdef MODULE - printk( + printk(KERN_INFO "azt3328: Sound driver for Aztech AZF3328-based soundcards such as PCI168.\n" "azt3328: Hardware was completely undocumented, unfortunately.\n" "azt3328: Feel free to contact andi AT lisas.de for bug reports etc.!\n" @@ -2308,36 +2386,52 @@ snd_azf3328_remove(struct pci_dev *pci) } #ifdef CONFIG_PM +static inline void +snd_azf3328_suspend_regs(unsigned long io_addr, unsigned count, u32 *saved_regs) +{ + unsigned reg; + + for (reg = 0; reg < count; ++reg) { + *saved_regs = inl(io_addr); + snd_azf3328_dbgpm("suspend: io 0x%04lx: 0x%08x\n", + io_addr, *saved_regs); + ++saved_regs; + io_addr += sizeof(*saved_regs); + } +} + static int snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_azf3328 *chip = card->private_data; - unsigned reg; + u16 *saved_regs_ctrl_u16; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); + snd_pcm_suspend_all(chip->pcm[AZF_CODEC_PLAYBACK]); + snd_pcm_suspend_all(chip->pcm[AZF_CODEC_I2S_OUT]); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) - chip->saved_regs_mixer[reg] = inw(chip->mixer_io + reg * 2); + snd_azf3328_suspend_regs(chip->mixer_io, + ARRAY_SIZE(chip->saved_regs_mixer), chip->saved_regs_mixer); /* make sure to disable master volume etc. to prevent looping sound */ snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1); snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) - chip->saved_regs_codec[reg] = inw(chip->codec_io + reg * 2); + snd_azf3328_suspend_regs(chip->ctrl_io, + ARRAY_SIZE(chip->saved_regs_ctrl), chip->saved_regs_ctrl); /* manually store the one currently relevant write-only reg, too */ - chip->saved_regs_codec[IDX_IO_6AH / 2] = chip->shadow_reg_codec_6AH; + saved_regs_ctrl_u16 = (u16 *)chip->saved_regs_ctrl; + saved_regs_ctrl_u16[IDX_IO_6AH / 2] = chip->shadow_reg_ctrl_6AH; - for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) - chip->saved_regs_game[reg] = inw(chip->game_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) - chip->saved_regs_mpu[reg] = inw(chip->mpu_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) - chip->saved_regs_opl3[reg] = inw(chip->opl3_io + reg * 2); + snd_azf3328_suspend_regs(chip->game_io, + ARRAY_SIZE(chip->saved_regs_game), chip->saved_regs_game); + snd_azf3328_suspend_regs(chip->mpu_io, + ARRAY_SIZE(chip->saved_regs_mpu), chip->saved_regs_mpu); + snd_azf3328_suspend_regs(chip->opl3_io, + ARRAY_SIZE(chip->saved_regs_opl3), chip->saved_regs_opl3); pci_disable_device(pci); pci_save_state(pci); @@ -2345,12 +2439,28 @@ snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state) return 0; } +static inline void +snd_azf3328_resume_regs(const u32 *saved_regs, + unsigned long io_addr, + unsigned count +) +{ + unsigned reg; + + for (reg = 0; reg < count; ++reg) { + outl(*saved_regs, io_addr); + snd_azf3328_dbgpm("resume: io 0x%04lx: 0x%08x --> 0x%08x\n", + io_addr, *saved_regs, inl(io_addr)); + ++saved_regs; + io_addr += sizeof(*saved_regs); + } +} + static int snd_azf3328_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); - struct snd_azf3328 *chip = card->private_data; - unsigned reg; + const struct snd_azf3328 *chip = card->private_data; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -2362,16 +2472,24 @@ snd_azf3328_resume(struct pci_dev *pci) } pci_set_master(pci); - for (reg = 0; reg < AZF_IO_SIZE_GAME_PM / 2; ++reg) - outw(chip->saved_regs_game[reg], chip->game_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; ++reg) - outw(chip->saved_regs_mpu[reg], chip->mpu_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_OPL3_PM / 2; ++reg) - outw(chip->saved_regs_opl3[reg], chip->opl3_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; ++reg) - outw(chip->saved_regs_mixer[reg], chip->mixer_io + reg * 2); - for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; ++reg) - outw(chip->saved_regs_codec[reg], chip->codec_io + reg * 2); + snd_azf3328_resume_regs(chip->saved_regs_game, chip->game_io, + ARRAY_SIZE(chip->saved_regs_game)); + snd_azf3328_resume_regs(chip->saved_regs_mpu, chip->mpu_io, + ARRAY_SIZE(chip->saved_regs_mpu)); + snd_azf3328_resume_regs(chip->saved_regs_opl3, chip->opl3_io, + ARRAY_SIZE(chip->saved_regs_opl3)); + + snd_azf3328_resume_regs(chip->saved_regs_mixer, chip->mixer_io, + ARRAY_SIZE(chip->saved_regs_mixer)); + + /* unfortunately with 32bit transfers, IDX_MIXER_PLAY_MASTER (0x02) + and IDX_MIXER_RESET (offset 0x00) get touched at the same time, + resulting in a mixer reset condition persisting until _after_ + master vol was restored. Thus master vol needs an extra restore. */ + outw(((u16 *)chip->saved_regs_mixer)[1], chip->mixer_io + 2); + + snd_azf3328_resume_regs(chip->saved_regs_ctrl, chip->ctrl_io, + ARRAY_SIZE(chip->saved_regs_ctrl)); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h index 974e05122f00..6f46b97650cc 100644 --- a/sound/pci/azt3328.h +++ b/sound/pci/azt3328.h @@ -6,50 +6,59 @@ /*** main I/O area port indices ***/ /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */ -#define AZF_IO_SIZE_CODEC 0x80 -#define AZF_IO_SIZE_CODEC_PM 0x70 +#define AZF_IO_SIZE_CTRL 0x80 +#define AZF_IO_SIZE_CTRL_PM 0x70 -/* the driver initialisation suggests a layout of 4 main areas: - * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??). +/* the driver initialisation suggests a layout of 4 areas + * within the main card control I/O: + * from 0x00 (playback codec), from 0x20 (recording codec) + * and from 0x40 (most certainly I2S out codec). * And another area from 0x60 to 0x6f (DirectX timer, IRQ management, * power management etc.???). */ -/** playback area **/ -#define IDX_IO_PLAY_FLAGS 0x00 /* PU:0x0000 */ +#define AZF_IO_OFFS_CODEC_PLAYBACK 0x00 +#define AZF_IO_OFFS_CODEC_CAPTURE 0x20 +#define AZF_IO_OFFS_CODEC_I2S_OUT 0x40 + +#define IDX_IO_CODEC_DMA_FLAGS 0x00 /* PU:0x0000 */ /* able to reactivate output after output muting due to 8/16bit * output change, just like 0x0002. * 0x0001 is the only bit that's able to start the DMA counter */ - #define DMA_RESUME 0x0001 /* paused if cleared ? */ + #define DMA_RESUME 0x0001 /* paused if cleared? */ /* 0x0002 *temporarily* set during DMA stopping. hmm * both 0x0002 and 0x0004 set in playback setup. */ /* able to reactivate output after output muting due to 8/16bit * output change, just like 0x0001. */ - #define DMA_PLAY_SOMETHING1 0x0002 /* \ alternated (toggled) */ + #define DMA_RUN_SOMETHING1 0x0002 /* \ alternated (toggled) */ /* 0x0004: NOT able to reactivate output */ - #define DMA_PLAY_SOMETHING2 0x0004 /* / bits */ + #define DMA_RUN_SOMETHING2 0x0004 /* / bits */ #define SOMETHING_ALMOST_ALWAYS_SET 0x0008 /* ???; can be modified */ #define DMA_EPILOGUE_SOMETHING 0x0010 #define DMA_SOMETHING_ELSE 0x0020 /* ??? */ - #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused ? not modifiable */ -#define IDX_IO_PLAY_IRQTYPE 0x02 /* PU:0x0001 */ + #define SOMETHING_UNMODIFIABLE 0xffc0 /* unused? not modifiable */ +#define IDX_IO_CODEC_IRQTYPE 0x02 /* PU:0x0001 */ /* write back to flags in case flags are set, in order to ACK IRQ in handler * (bit 1 of port 0x64 indicates interrupt for one of these three types) * sometimes in this case it just writes 0xffff to globally ACK all IRQs * settings written are not reflected when reading back, though. - * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows ? */ - #define IRQ_PLAY_SOMETHING 0x0001 /* something & ACK */ - #define IRQ_FINISHED_PLAYBUF_1 0x0002 /* 1st dmabuf finished & ACK */ - #define IRQ_FINISHED_PLAYBUF_2 0x0004 /* 2nd dmabuf finished & ACK */ + * seems to be IRQ, too (frequently used: port |= 0x07 !), but who knows? */ + #define IRQ_SOMETHING 0x0001 /* something & ACK */ + #define IRQ_FINISHED_DMABUF_1 0x0002 /* 1st dmabuf finished & ACK */ + #define IRQ_FINISHED_DMABUF_2 0x0004 /* 2nd dmabuf finished & ACK */ #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */ #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */ - #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused ? not modifiable */ -#define IDX_IO_PLAY_DMA_START_1 0x04 /* start address of 1st DMA play area, PU:0x00000000 */ -#define IDX_IO_PLAY_DMA_START_2 0x08 /* start address of 2nd DMA play area, PU:0x00000000 */ -#define IDX_IO_PLAY_DMA_LEN_1 0x0c /* length of 1st DMA play area, PU:0x0000 */ -#define IDX_IO_PLAY_DMA_LEN_2 0x0e /* length of 2nd DMA play area, PU:0x0000 */ -#define IDX_IO_PLAY_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */ -#define IDX_IO_PLAY_DMA_CURROFS 0x14 /* offset within current DMA play area, PU:0x0000 */ -#define IDX_IO_PLAY_SOUNDFORMAT 0x16 /* PU:0x0010 */ + #define IRQMASK_UNMODIFIABLE 0xffe0 /* unused? not modifiable */ + /* start address of 1st DMA transfer area, PU:0x00000000 */ +#define IDX_IO_CODEC_DMA_START_1 0x04 + /* start address of 2nd DMA transfer area, PU:0x00000000 */ +#define IDX_IO_CODEC_DMA_START_2 0x08 + /* both lengths of DMA transfer areas, PU:0x00000000 + length1: offset 0x0c, length2: offset 0x0e */ +#define IDX_IO_CODEC_DMA_LENGTHS 0x0c +#define IDX_IO_CODEC_DMA_CURRPOS 0x10 /* current DMA position, PU:0x00000000 */ + /* offset within current DMA transfer area, PU:0x0000 */ +#define IDX_IO_CODEC_DMA_CURROFS 0x14 +#define IDX_IO_CODEC_SOUNDFORMAT 0x16 /* PU:0x0010 */ /* all unspecified bits can't be modified */ #define SOUNDFORMAT_FREQUENCY_MASK 0x000f #define SOUNDFORMAT_XTAL1 0x00 @@ -76,6 +85,7 @@ #define SOUNDFORMAT_FLAG_16BIT 0x0010 #define SOUNDFORMAT_FLAG_2CHANNELS 0x0020 + /* define frequency helpers, for maximum value safety */ enum azf_freq_t { #define AZF_FREQ(rate) AZF_FREQ_##rate = rate @@ -96,29 +106,6 @@ enum azf_freq_t { #undef AZF_FREQ }; -/** recording area (see also: playback bit flag definitions) **/ -#define IDX_IO_REC_FLAGS 0x20 /* ??, PU:0x0000 */ -#define IDX_IO_REC_IRQTYPE 0x22 /* ??, PU:0x0000 */ - #define IRQ_REC_SOMETHING 0x0001 /* something & ACK */ - #define IRQ_FINISHED_RECBUF_1 0x0002 /* 1st dmabuf finished & ACK */ - #define IRQ_FINISHED_RECBUF_2 0x0004 /* 2nd dmabuf finished & ACK */ - /* hmm, maybe these are just the corresponding *recording* flags ? - * but OTOH they are most likely at port 0x22 instead */ - #define IRQMASK_SOME_STATUS_1 0x0008 /* \ related bits */ - #define IRQMASK_SOME_STATUS_2 0x0010 /* / (checked together in loop) */ -#define IDX_IO_REC_DMA_START_1 0x24 /* PU:0x00000000 */ -#define IDX_IO_REC_DMA_START_2 0x28 /* PU:0x00000000 */ -#define IDX_IO_REC_DMA_LEN_1 0x2c /* PU:0x0000 */ -#define IDX_IO_REC_DMA_LEN_2 0x2e /* PU:0x0000 */ -#define IDX_IO_REC_DMA_CURRPOS 0x30 /* PU:0x00000000 */ -#define IDX_IO_REC_DMA_CURROFS 0x34 /* PU:0x00000000 */ -#define IDX_IO_REC_SOUNDFORMAT 0x36 /* PU:0x0000 */ - -/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/ -#define IDX_IO_SOMETHING_FLAGS 0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */ -/* general */ -#define IDX_IO_42H 0x42 /* PU:0x0001 */ - /** DirectX timer, main interrupt area (FIXME: and something else?) **/ #define IDX_IO_TIMER_VALUE 0x60 /* found this timer area by pure luck :-) */ /* timer countdown value; triggers IRQ when timer is finished */ @@ -133,17 +120,19 @@ enum azf_freq_t { #define IDX_IO_IRQSTATUS 0x64 /* some IRQ bit in here might also be used to signal a power-management timer * timeout, to request shutdown of the chip (e.g. AD1815JS has such a thing). - * Some OPL3 hardware (e.g. in LM4560) has some special timer hardware which - * can trigger an OPL3 timer IRQ, so maybe there's such a thing as well... */ + * OPL3 hardware contains several timers which confusingly in most cases + * are NOT routed to an IRQ, but some designs (e.g. LM4560) DO support that, + * so I wouldn't be surprised at all to discover that AZF3328 + * supports that thing as well... */ #define IRQ_PLAYBACK 0x0001 #define IRQ_RECORDING 0x0002 - #define IRQ_UNKNOWN1 0x0004 /* most probably I2S port */ + #define IRQ_I2S_OUT 0x0004 /* this IS I2S, right!? (untested) */ #define IRQ_GAMEPORT 0x0008 /* Interrupt of Digital(ly) Enhanced Game Port */ #define IRQ_MPU401 0x0010 #define IRQ_TIMER 0x0020 /* DirectX timer */ - #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly I2S port? */ - #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly I2S port? */ + #define IRQ_UNKNOWN2 0x0040 /* probably unused, or possibly OPL3 timer? */ + #define IRQ_UNKNOWN3 0x0080 /* probably unused, or possibly OPL3 timer? */ #define IDX_IO_66H 0x66 /* writing 0xffff returns 0x0000 */ /* this is set to e.g. 0x3ff or 0x300, and writable; * maybe some buffer limit, but I couldn't find out more, PU:0x00ff: */ @@ -206,7 +195,7 @@ enum azf_freq_t { /*** Gameport area port indices ***/ /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ #define AZF_IO_SIZE_GAME 0x08 -#define AZF_IO_SIZE_GAME_PM 0x06 +#define AZF_IO_SIZE_GAME_PM 0x06 enum { AZF_GAME_LEGACY_IO_PORT = 0x200 @@ -272,6 +261,12 @@ enum { * 11 --> 1/200: */ #define GAME_HWCFG_ADC_COUNTER_FREQ_MASK 0x06 + /* FIXME: these values might be reversed... */ + #define GAME_HWCFG_ADC_COUNTER_FREQ_STD 0 + #define GAME_HWCFG_ADC_COUNTER_FREQ_1_2 1 + #define GAME_HWCFG_ADC_COUNTER_FREQ_1_20 2 + #define GAME_HWCFG_ADC_COUNTER_FREQ_1_200 3 + /* enable gameport legacy I/O address (0x200) * I was unable to locate any configurability for a different address: */ #define GAME_HWCFG_LEGACY_ADDRESS_ENABLE 0x08 @@ -281,6 +276,7 @@ enum { #define AZF_IO_SIZE_MPU_PM 0x04 /*** OPL3 synth ***/ +/* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ #define AZF_IO_SIZE_OPL3 0x08 #define AZF_IO_SIZE_OPL3_PM 0x06 /* hmm, given that a standard OPL3 has 4 registers only, @@ -340,4 +336,7 @@ enum { #define SET_CHAN_LEFT 1 #define SET_CHAN_RIGHT 2 +/* helper macro to align I/O port ranges to 32bit I/O width */ +#define AZF_ALIGN(x) (((x) + 3) & (~3)) + #endif /* __SOUND_AZT3328_H */ diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index 4eb55aa33612..b5189495d58a 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -35,7 +35,7 @@ #ifdef CONFIG_SND_CS46XX_NEW_DSP -#define CS46XX_MIN_PERIOD_SIZE 1 +#define CS46XX_MIN_PERIOD_SIZE 64 #define CS46XX_MAX_PERIOD_SIZE 1024*1024 #else #define CS46XX_MIN_PERIOD_SIZE 2048 diff --git a/sound/pci/ctxfi/ct20k2reg.h b/sound/pci/ctxfi/ct20k2reg.h index 2d07986f57cc..e0394e3996e8 100644 --- a/sound/pci/ctxfi/ct20k2reg.h +++ b/sound/pci/ctxfi/ct20k2reg.h @@ -11,9 +11,12 @@ /* Timer Registers */ -#define TIMER_TIMR 0x1B7004 -#define INTERRUPT_GIP 0x1B7010 -#define INTERRUPT_GIE 0x1B7014 +#define WC 0x1b7000 +#define TIMR 0x1b7004 +# define TIMR_IE (1<<15) +# define TIMR_IP (1<<14) +#define GIP 0x1b7010 +#define GIE 0x1b7014 /* I2C Registers */ #define I2C_IF_ADDRESS 0x1B9000 diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index a7f4a671f7b7..fee35cfc0c7f 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -63,7 +63,7 @@ static int amixer_set_input(struct amixer *amixer, struct rsc *rsc) hw = amixer->rsc.hw; hw->amixer_set_mode(amixer->rsc.ctrl_blk, AMIXER_Y_IMMEDIATE); amixer->input = rsc; - if (NULL == rsc) + if (!rsc) hw->amixer_set_x(amixer->rsc.ctrl_blk, BLANK_SLOT); else hw->amixer_set_x(amixer->rsc.ctrl_blk, @@ -99,7 +99,7 @@ static int amixer_set_sum(struct amixer *amixer, struct sum *sum) hw = amixer->rsc.hw; amixer->sum = sum; - if (NULL == sum) { + if (!sum) { hw->amixer_set_se(amixer->rsc.ctrl_blk, 0); } else { hw->amixer_set_se(amixer->rsc.ctrl_blk, 1); @@ -124,20 +124,20 @@ static int amixer_commit_write(struct amixer *amixer) /* Program master and conjugate resources */ amixer->rsc.ops->master(&amixer->rsc); - if (NULL != input) + if (input) input->ops->master(input); - if (NULL != sum) + if (sum) sum->rsc.ops->master(&sum->rsc); for (i = 0; i < amixer->rsc.msr; i++) { hw->amixer_set_dirty_all(amixer->rsc.ctrl_blk); - if (NULL != input) { + if (input) { hw->amixer_set_x(amixer->rsc.ctrl_blk, input->ops->output_slot(input)); input->ops->next_conj(input); } - if (NULL != sum) { + if (sum) { hw->amixer_set_sadr(amixer->rsc.ctrl_blk, sum->rsc.ops->index(&sum->rsc)); sum->rsc.ops->next_conj(&sum->rsc); @@ -147,10 +147,10 @@ static int amixer_commit_write(struct amixer *amixer) amixer->rsc.ops->next_conj(&amixer->rsc); } amixer->rsc.ops->master(&amixer->rsc); - if (NULL != input) + if (input) input->ops->master(input); - if (NULL != sum) + if (sum) sum->rsc.ops->master(&sum->rsc); return 0; @@ -303,7 +303,7 @@ int amixer_mgr_create(void *hw, struct amixer_mgr **ramixer_mgr) *ramixer_mgr = NULL; amixer_mgr = kzalloc(sizeof(*amixer_mgr), GFP_KERNEL); - if (NULL == amixer_mgr) + if (!amixer_mgr) return -ENOMEM; err = rsc_mgr_init(&amixer_mgr->mgr, AMIXER, AMIXER_RESOURCE_NUM, hw); @@ -456,7 +456,7 @@ int sum_mgr_create(void *hw, struct sum_mgr **rsum_mgr) *rsum_mgr = NULL; sum_mgr = kzalloc(sizeof(*sum_mgr), GFP_KERNEL); - if (NULL == sum_mgr) + if (!sum_mgr) return -ENOMEM; err = rsc_mgr_init(&sum_mgr->mgr, SUM, SUM_RESOURCE_NUM, hw); diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index a49c76647307..b1b3a644f738 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -136,7 +136,7 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct snd_pcm_runtime *runtime; struct ct_vm *vm; - if (NULL == apcm->substream) + if (!apcm->substream) return 0; runtime = apcm->substream->runtime; @@ -144,7 +144,7 @@ static int ct_map_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) apcm->vm_block = vm->map(vm, apcm->substream, runtime->dma_bytes); - if (NULL == apcm->vm_block) + if (!apcm->vm_block) return -ENOENT; return 0; @@ -154,7 +154,7 @@ static void ct_unmap_audio_buffer(struct ct_atc *atc, struct ct_atc_pcm *apcm) { struct ct_vm *vm; - if (NULL == apcm->vm_block) + if (!apcm->vm_block) return; vm = atc->vm; @@ -231,16 +231,16 @@ atc_get_pitch(unsigned int input_rate, unsigned int output_rate) static int select_rom(unsigned int pitch) { - if ((pitch > 0x00428f5c) && (pitch < 0x01b851ec)) { + if (pitch > 0x00428f5c && pitch < 0x01b851ec) { /* 0.26 <= pitch <= 1.72 */ return 1; - } else if ((0x01d66666 == pitch) || (0x01d66667 == pitch)) { + } else if (pitch == 0x01d66666 || pitch == 0x01d66667) { /* pitch == 1.8375 */ return 2; - } else if (0x02000000 == pitch) { + } else if (pitch == 0x02000000) { /* pitch == 2 */ return 3; - } else if ((pitch >= 0x0) && (pitch <= 0x08000000)) { + } else if (pitch >= 0x0 && pitch <= 0x08000000) { /* 0 <= pitch <= 8 */ return 0; } else { @@ -283,7 +283,7 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) /* Get AMIXER resource */ n_amixer = (n_amixer < 2) ? 2 : n_amixer; apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); - if (NULL == apcm->amixers) { + if (!apcm->amixers) { err = -ENOMEM; goto error1; } @@ -311,7 +311,7 @@ static int atc_pcm_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) INIT_VOL, atc->pcm[i+device*2]); mutex_unlock(&atc->atc_mutex); src = src->ops->next_interleave(src); - if (NULL == src) + if (!src) src = apcm->src; } @@ -334,7 +334,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) struct srcimp *srcimp; int i; - if (NULL != apcm->srcimps) { + if (apcm->srcimps) { for (i = 0; i < apcm->n_srcimp; i++) { srcimp = apcm->srcimps[i]; srcimp->ops->unmap(srcimp); @@ -345,7 +345,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) apcm->srcimps = NULL; } - if (NULL != apcm->srccs) { + if (apcm->srccs) { for (i = 0; i < apcm->n_srcc; i++) { src_mgr->put_src(src_mgr, apcm->srccs[i]); apcm->srccs[i] = NULL; @@ -354,7 +354,7 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) apcm->srccs = NULL; } - if (NULL != apcm->amixers) { + if (apcm->amixers) { for (i = 0; i < apcm->n_amixer; i++) { amixer_mgr->put_amixer(amixer_mgr, apcm->amixers[i]); apcm->amixers[i] = NULL; @@ -363,17 +363,17 @@ atc_pcm_release_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) apcm->amixers = NULL; } - if (NULL != apcm->mono) { + if (apcm->mono) { sum_mgr->put_sum(sum_mgr, apcm->mono); apcm->mono = NULL; } - if (NULL != apcm->src) { + if (apcm->src) { src_mgr->put_src(src_mgr, apcm->src); apcm->src = NULL; } - if (NULL != apcm->vm_block) { + if (apcm->vm_block) { /* Undo device virtual mem map */ ct_unmap_audio_buffer(atc, apcm); apcm->vm_block = NULL; @@ -419,7 +419,7 @@ static int atc_pcm_stop(struct ct_atc *atc, struct ct_atc_pcm *apcm) src->ops->set_state(src, SRC_STATE_OFF); src->ops->commit_write(src); - if (NULL != apcm->srccs) { + if (apcm->srccs) { for (i = 0; i < apcm->n_srcc; i++) { src = apcm->srccs[i]; src->ops->set_bm(src, 0); @@ -544,18 +544,18 @@ atc_pcm_capture_get_resources(struct ct_atc *atc, struct ct_atc_pcm *apcm) if (n_srcc) { apcm->srccs = kzalloc(sizeof(void *)*n_srcc, GFP_KERNEL); - if (NULL == apcm->srccs) + if (!apcm->srccs) return -ENOMEM; } if (n_amixer) { apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); - if (NULL == apcm->amixers) { + if (!apcm->amixers) { err = -ENOMEM; goto error1; } } apcm->srcimps = kzalloc(sizeof(void *)*n_srcimp, GFP_KERNEL); - if (NULL == apcm->srcimps) { + if (!apcm->srcimps) { err = -ENOMEM; goto error1; } @@ -818,7 +818,7 @@ static int spdif_passthru_playback_get_resources(struct ct_atc *atc, /* Get AMIXER resource */ n_amixer = (n_amixer < 2) ? 2 : n_amixer; apcm->amixers = kzalloc(sizeof(void *)*n_amixer, GFP_KERNEL); - if (NULL == apcm->amixers) { + if (!apcm->amixers) { err = -ENOMEM; goto error1; } @@ -919,7 +919,7 @@ spdif_passthru_playback_prepare(struct ct_atc *atc, struct ct_atc_pcm *apcm) amixer = apcm->amixers[i]; amixer->ops->setup(amixer, &src->rsc, INIT_VOL, NULL); src = src->ops->next_interleave(src); - if (NULL == src) + if (!src) src = apcm->src; } /* Connect to SPDIFOO */ @@ -1121,7 +1121,7 @@ static int atc_release_resources(struct ct_atc *atc) struct ct_mixer *mixer = NULL; /* disconnect internal mixer objects */ - if (NULL != atc->mixer) { + if (atc->mixer) { mixer = atc->mixer; mixer->set_input_left(mixer, MIX_LINE_IN, NULL); mixer->set_input_right(mixer, MIX_LINE_IN, NULL); @@ -1131,7 +1131,7 @@ static int atc_release_resources(struct ct_atc *atc) mixer->set_input_right(mixer, MIX_SPDIF_IN, NULL); } - if (NULL != atc->daios) { + if (atc->daios) { daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; for (i = 0; i < atc->n_daio; i++) { daio = atc->daios[i]; @@ -1149,7 +1149,7 @@ static int atc_release_resources(struct ct_atc *atc) atc->daios = NULL; } - if (NULL != atc->pcm) { + if (atc->pcm) { sum_mgr = atc->rsc_mgrs[SUM]; for (i = 0; i < atc->n_pcm; i++) sum_mgr->put_sum(sum_mgr, atc->pcm[i]); @@ -1158,7 +1158,7 @@ static int atc_release_resources(struct ct_atc *atc) atc->pcm = NULL; } - if (NULL != atc->srcs) { + if (atc->srcs) { src_mgr = atc->rsc_mgrs[SRC]; for (i = 0; i < atc->n_src; i++) src_mgr->put_src(src_mgr, atc->srcs[i]); @@ -1167,7 +1167,7 @@ static int atc_release_resources(struct ct_atc *atc) atc->srcs = NULL; } - if (NULL != atc->srcimps) { + if (atc->srcimps) { srcimp_mgr = atc->rsc_mgrs[SRCIMP]; for (i = 0; i < atc->n_srcimp; i++) { srcimp = atc->srcimps[i]; @@ -1185,7 +1185,7 @@ static int ct_atc_destroy(struct ct_atc *atc) { int i = 0; - if (NULL == atc) + if (!atc) return 0; if (atc->timer) { @@ -1196,21 +1196,20 @@ static int ct_atc_destroy(struct ct_atc *atc) atc_release_resources(atc); /* Destroy internal mixer objects */ - if (NULL != atc->mixer) + if (atc->mixer) ct_mixer_destroy(atc->mixer); for (i = 0; i < NUM_RSCTYP; i++) { - if ((NULL != rsc_mgr_funcs[i].destroy) && - (NULL != atc->rsc_mgrs[i])) + if (rsc_mgr_funcs[i].destroy && atc->rsc_mgrs[i]) rsc_mgr_funcs[i].destroy(atc->rsc_mgrs[i]); } - if (NULL != atc->hw) + if (atc->hw) destroy_hw_obj((struct hw *)atc->hw); /* Destroy device virtual memory manager object */ - if (NULL != atc->vm) { + if (atc->vm) { ct_vm_destroy(atc->vm); atc->vm = NULL; } @@ -1275,7 +1274,7 @@ int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc) alsa_dev_funcs[MIXER].public_name = atc->chip_name; for (i = 0; i < NUM_CTALSADEVS; i++) { - if (NULL == alsa_dev_funcs[i].create) + if (!alsa_dev_funcs[i].create) continue; err = alsa_dev_funcs[i].create(atc, i, @@ -1312,7 +1311,7 @@ static int __devinit atc_create_hw_devs(struct ct_atc *atc) return err; for (i = 0; i < NUM_RSCTYP; i++) { - if (NULL == rsc_mgr_funcs[i].create) + if (!rsc_mgr_funcs[i].create) continue; err = rsc_mgr_funcs[i].create(atc->hw, &atc->rsc_mgrs[i]); @@ -1339,19 +1338,19 @@ static int atc_get_resources(struct ct_atc *atc) int err, i; atc->daios = kzalloc(sizeof(void *)*(DAIONUM), GFP_KERNEL); - if (NULL == atc->daios) + if (!atc->daios) return -ENOMEM; atc->srcs = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); - if (NULL == atc->srcs) + if (!atc->srcs) return -ENOMEM; atc->srcimps = kzalloc(sizeof(void *)*(2*2), GFP_KERNEL); - if (NULL == atc->srcimps) + if (!atc->srcimps) return -ENOMEM; atc->pcm = kzalloc(sizeof(void *)*(2*4), GFP_KERNEL); - if (NULL == atc->pcm) + if (!atc->pcm) return -ENOMEM; daio_mgr = (struct daio_mgr *)atc->rsc_mgrs[DAIO]; @@ -1648,7 +1647,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, *ratc = NULL; atc = kzalloc(sizeof(*atc), GFP_KERNEL); - if (NULL == atc) + if (!atc) return -ENOMEM; /* Set operations */ diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index deb6cfa73600..af56eb949bde 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -173,7 +173,7 @@ static int dao_set_left_input(struct dao *dao, struct rsc *input) int i; entry = kzalloc((sizeof(*entry) * daio->rscl.msr), GFP_KERNEL); - if (NULL == entry) + if (!entry) return -ENOMEM; /* Program master and conjugate resources */ @@ -201,7 +201,7 @@ static int dao_set_right_input(struct dao *dao, struct rsc *input) int i; entry = kzalloc((sizeof(*entry) * daio->rscr.msr), GFP_KERNEL); - if (NULL == entry) + if (!entry) return -ENOMEM; /* Program master and conjugate resources */ @@ -228,7 +228,7 @@ static int dao_clear_left_input(struct dao *dao) struct daio *daio = &dao->daio; int i; - if (NULL == dao->imappers[0]) + if (!dao->imappers[0]) return 0; entry = dao->imappers[0]; @@ -252,7 +252,7 @@ static int dao_clear_right_input(struct dao *dao) struct daio *daio = &dao->daio; int i; - if (NULL == dao->imappers[daio->rscl.msr]) + if (!dao->imappers[daio->rscl.msr]) return 0; entry = dao->imappers[daio->rscl.msr]; @@ -408,7 +408,7 @@ static int dao_rsc_init(struct dao *dao, return err; dao->imappers = kzalloc(sizeof(void *)*desc->msr*2, GFP_KERNEL); - if (NULL == dao->imappers) { + if (!dao->imappers) { err = -ENOMEM; goto error1; } @@ -442,11 +442,11 @@ error1: static int dao_rsc_uninit(struct dao *dao) { - if (NULL != dao->imappers) { - if (NULL != dao->imappers[0]) + if (dao->imappers) { + if (dao->imappers[0]) dao_clear_left_input(dao); - if (NULL != dao->imappers[dao->daio.rscl.msr]) + if (dao->imappers[dao->daio.rscl.msr]) dao_clear_right_input(dao); kfree(dao->imappers); @@ -555,7 +555,7 @@ static int get_daio_rsc(struct daio_mgr *mgr, /* Allocate mem for daio resource */ if (desc->type <= DAIO_OUT_MAX) { dao = kzalloc(sizeof(*dao), GFP_KERNEL); - if (NULL == dao) { + if (!dao) { err = -ENOMEM; goto error; } @@ -566,7 +566,7 @@ static int get_daio_rsc(struct daio_mgr *mgr, *rdaio = &dao->daio; } else { dai = kzalloc(sizeof(*dai), GFP_KERNEL); - if (NULL == dai) { + if (!dai) { err = -ENOMEM; goto error; } @@ -583,9 +583,9 @@ static int get_daio_rsc(struct daio_mgr *mgr, return 0; error: - if (NULL != dao) + if (dao) kfree(dao); - else if (NULL != dai) + else if (dai) kfree(dai); spin_lock_irqsave(&mgr->mgr_lock, flags); @@ -663,7 +663,7 @@ static int daio_imap_add(struct daio_mgr *mgr, struct imapper *entry) int err; spin_lock_irqsave(&mgr->imap_lock, flags); - if ((0 == entry->addr) && (mgr->init_imap_added)) { + if (!entry->addr && mgr->init_imap_added) { input_mapper_delete(&mgr->imappers, mgr->init_imap, daio_map_op, mgr); mgr->init_imap_added = 0; @@ -707,7 +707,7 @@ int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr) *rdaio_mgr = NULL; daio_mgr = kzalloc(sizeof(*daio_mgr), GFP_KERNEL); - if (NULL == daio_mgr) + if (!daio_mgr) return -ENOMEM; err = rsc_mgr_init(&daio_mgr->mgr, DAIO, DAIO_RESOURCE_NUM, hw); @@ -718,7 +718,7 @@ int daio_mgr_create(void *hw, struct daio_mgr **rdaio_mgr) spin_lock_init(&daio_mgr->imap_lock); INIT_LIST_HEAD(&daio_mgr->imappers); entry = kzalloc(sizeof(*entry), GFP_KERNEL); - if (NULL == entry) { + if (!entry) { err = -ENOMEM; goto error2; } diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index ad3e1d144464..0cf400f879f9 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -168,7 +168,7 @@ static int src_get_rsc_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -494,7 +494,7 @@ static int src_mgr_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -515,7 +515,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -702,7 +702,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -723,7 +723,7 @@ static int amixer_mgr_get_ctrl_blk(void **rblk) *rblk = NULL; /*blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk;*/ @@ -909,7 +909,7 @@ static int dai_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -958,7 +958,7 @@ static int dao_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -1152,7 +1152,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; blk->i2sctl = hw_read_20kx(hw, I2SCTL); @@ -1808,7 +1808,7 @@ static int uaa_to_xfi(struct pci_dev *pci) /* By default, Hendrix card UAA Bar0 should be using memory... */ io_base = pci_resource_start(pci, 0); mem_base = ioremap(io_base, pci_resource_len(pci, 0)); - if (NULL == mem_base) + if (!mem_base) return -ENOENT; /* Read current mode from Mode Change Register */ @@ -1977,7 +1977,7 @@ static int hw_card_shutdown(struct hw *hw) hw->irq = -1; - if (NULL != ((void *)hw->mem_base)) + if (hw->mem_base) iounmap((void *)hw->mem_base); hw->mem_base = (unsigned long)NULL; @@ -2274,7 +2274,7 @@ int __devinit create_20k1_hw_obj(struct hw **rhw) *rhw = NULL; hw20k1 = kzalloc(sizeof(*hw20k1), GFP_KERNEL); - if (NULL == hw20k1) + if (!hw20k1) return -ENOMEM; spin_lock_init(&hw20k1->reg_20k1_lock); diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index dec46d04b041..b6b11bfe7574 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -166,7 +166,7 @@ static int src_get_rsc_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -492,7 +492,7 @@ static int src_mgr_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -513,7 +513,7 @@ static int srcimp_mgr_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -702,7 +702,7 @@ static int amixer_rsc_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -891,7 +891,7 @@ static int dai_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -941,7 +941,7 @@ static int dao_get_ctrl_blk(void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; *rblk = blk; @@ -1092,7 +1092,7 @@ static int daio_mgr_get_ctrl_blk(struct hw *hw, void **rblk) *rblk = NULL; blk = kzalloc(sizeof(*blk), GFP_KERNEL); - if (NULL == blk) + if (!blk) return -ENOMEM; for (i = 0; i < 8; i++) { @@ -1112,6 +1112,26 @@ static int daio_mgr_put_ctrl_blk(void *blk) return 0; } +/* Timer interrupt */ +static int set_timer_irq(struct hw *hw, int enable) +{ + hw_write_20kx(hw, GIE, enable ? IT_INT : 0); + return 0; +} + +static int set_timer_tick(struct hw *hw, unsigned int ticks) +{ + if (ticks) + ticks |= TIMR_IE | TIMR_IP; + hw_write_20kx(hw, TIMR, ticks); + return 0; +} + +static unsigned int get_wc(struct hw *hw) +{ + return hw_read_20kx(hw, WC); +} + /* Card hardware initialization block */ struct dac_conf { unsigned int msr; /* master sample rate in rsrs */ @@ -1841,6 +1861,22 @@ static int hw_have_digit_io_switch(struct hw *hw) return 0; } +static irqreturn_t ct_20k2_interrupt(int irq, void *dev_id) +{ + struct hw *hw = dev_id; + unsigned int status; + + status = hw_read_20kx(hw, GIP); + if (!status) + return IRQ_NONE; + + if (hw->irq_callback) + hw->irq_callback(hw->irq_callback_data, status); + + hw_write_20kx(hw, GIP, status); + return IRQ_HANDLED; +} + static int hw_card_start(struct hw *hw) { int err = 0; @@ -1868,7 +1904,7 @@ static int hw_card_start(struct hw *hw) hw->io_base = pci_resource_start(hw->pci, 2); hw->mem_base = (unsigned long)ioremap(hw->io_base, pci_resource_len(hw->pci, 2)); - if (NULL == (void *)hw->mem_base) { + if (!hw->mem_base) { err = -ENOENT; goto error2; } @@ -1879,12 +1915,15 @@ static int hw_card_start(struct hw *hw) set_field(&gctl, GCTL_UAA, 0); hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); - /*if ((err = request_irq(pci->irq, ct_atc_interrupt, IRQF_SHARED, - atc->chip_details->nm_card, hw))) { - goto error3; + if (hw->irq < 0) { + err = request_irq(pci->irq, ct_20k2_interrupt, IRQF_SHARED, + "ctxfi", hw); + if (err < 0) { + printk(KERN_ERR "XFi: Cannot get irq %d\n", pci->irq); + goto error2; + } + hw->irq = pci->irq; } - hw->irq = pci->irq; - */ pci_set_master(pci); @@ -1923,7 +1962,7 @@ static int hw_card_shutdown(struct hw *hw) hw->irq = -1; - if (NULL != ((void *)hw->mem_base)) + if (hw->mem_base) iounmap((void *)hw->mem_base); hw->mem_base = (unsigned long)NULL; @@ -1972,7 +2011,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) hw_write_20kx(hw, GLOBAL_CNTL_GCTL, gctl); /* Reset all global pending interrupts */ - hw_write_20kx(hw, INTERRUPT_GIE, 0); + hw_write_20kx(hw, GIE, 0); /* Reset all SRC pending interrupts */ hw_write_20kx(hw, SRC_IP, 0); @@ -2149,6 +2188,10 @@ static struct hw ct20k2_preset __devinitdata = { .daio_mgr_set_imapnxt = daio_mgr_set_imapnxt, .daio_mgr_set_imapaddr = daio_mgr_set_imapaddr, .daio_mgr_commit_write = daio_mgr_commit_write, + + .set_timer_irq = set_timer_irq, + .set_timer_tick = set_timer_tick, + .get_wc = get_wc, }; int __devinit create_20k2_hw_obj(struct hw **rhw) diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index f26d7cd9db9f..15c1e7271ea8 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -654,7 +654,7 @@ ct_mixer_kcontrol_new(struct ct_mixer *mixer, struct snd_kcontrol_new *new) int err; kctl = snd_ctl_new1(new, mixer->atc); - if (NULL == kctl) + if (!kctl) return -ENOMEM; if (SNDRV_CTL_ELEM_IFACE_PCM == kctl->id.iface) @@ -837,17 +837,17 @@ static int ct_mixer_get_mem(struct ct_mixer **rmixer) *rmixer = NULL; /* Allocate mem for mixer obj */ mixer = kzalloc(sizeof(*mixer), GFP_KERNEL); - if (NULL == mixer) + if (!mixer) return -ENOMEM; mixer->amixers = kzalloc(sizeof(void *)*(NUM_CT_AMIXERS*CHN_NUM), GFP_KERNEL); - if (NULL == mixer->amixers) { + if (!mixer->amixers) { err = -ENOMEM; goto error1; } mixer->sums = kzalloc(sizeof(void *)*(NUM_CT_SUMS*CHN_NUM), GFP_KERNEL); - if (NULL == mixer->sums) { + if (!mixer->sums) { err = -ENOMEM; goto error2; } diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index 60ea23180acb..d0dc227fbdd3 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -97,7 +97,7 @@ static void ct_atc_pcm_interrupt(struct ct_atc_pcm *atc_pcm) { struct ct_atc_pcm *apcm = atc_pcm; - if (NULL == apcm->substream) + if (!apcm->substream) return; snd_pcm_period_elapsed(apcm->substream); @@ -123,7 +123,7 @@ static int ct_pcm_playback_open(struct snd_pcm_substream *substream) int err; apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (NULL == apcm) + if (!apcm) return -ENOMEM; apcm->substream = substream; @@ -271,7 +271,7 @@ static int ct_pcm_capture_open(struct snd_pcm_substream *substream) int err; apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (NULL == apcm) + if (!apcm) return -ENOMEM; apcm->started = 0; diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 889c495bb7d1..7dfaf67344d4 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -144,7 +144,7 @@ int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw) rsc->msr = msr; rsc->hw = hw; rsc->ops = &rsc_generic_ops; - if (NULL == hw) { + if (!hw) { rsc->ctrl_blk = NULL; return 0; } @@ -216,7 +216,7 @@ int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, mgr->type = NUM_RSCTYP; mgr->rscs = kzalloc(((amount + 8 - 1) / 8), GFP_KERNEL); - if (NULL == mgr->rscs) + if (!mgr->rscs) return -ENOMEM; switch (type) { diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index df43a5cd3938..c749fa720889 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -441,7 +441,7 @@ get_src_rsc(struct src_mgr *mgr, const struct src_desc *desc, struct src **rsrc) else src = kzalloc(sizeof(*src), GFP_KERNEL); - if (NULL == src) { + if (!src) { err = -ENOMEM; goto error1; } @@ -550,7 +550,7 @@ int src_mgr_create(void *hw, struct src_mgr **rsrc_mgr) *rsrc_mgr = NULL; src_mgr = kzalloc(sizeof(*src_mgr), GFP_KERNEL); - if (NULL == src_mgr) + if (!src_mgr) return -ENOMEM; err = rsc_mgr_init(&src_mgr->mgr, SRC, SRC_RESOURCE_NUM, hw); @@ -679,7 +679,7 @@ static int srcimp_rsc_init(struct srcimp *srcimp, /* Reserve memory for imapper nodes */ srcimp->imappers = kzalloc(sizeof(struct imapper)*desc->msr, GFP_KERNEL); - if (NULL == srcimp->imappers) { + if (!srcimp->imappers) { err = -ENOMEM; goto error1; } @@ -833,7 +833,7 @@ int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr) *rsrcimp_mgr = NULL; srcimp_mgr = kzalloc(sizeof(*srcimp_mgr), GFP_KERNEL); - if (NULL == srcimp_mgr) + if (!srcimp_mgr) return -ENOMEM; err = rsc_mgr_init(&srcimp_mgr->mgr, SRCIMP, SRCIMP_RESOURCE_NUM, hw); @@ -844,7 +844,7 @@ int srcimp_mgr_create(void *hw, struct srcimp_mgr **rsrcimp_mgr) spin_lock_init(&srcimp_mgr->imap_lock); INIT_LIST_HEAD(&srcimp_mgr->imappers); entry = kzalloc(sizeof(*entry), GFP_KERNEL); - if (NULL == entry) { + if (!entry) { err = -ENOMEM; goto error2; } diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 67665a7e43c6..6b78752e9503 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -60,7 +60,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) } block = kzalloc(sizeof(*block), GFP_KERNEL); - if (NULL == block) + if (!block) goto out; block->addr = entry->addr; @@ -181,7 +181,7 @@ int ct_vm_create(struct ct_vm **rvm) *rvm = NULL; vm = kzalloc(sizeof(*vm), GFP_KERNEL); - if (NULL == vm) + if (!vm) return -ENOMEM; mutex_init(&vm->lock); @@ -189,7 +189,7 @@ int ct_vm_create(struct ct_vm **rvm) /* Allocate page table pages */ for (i = 0; i < CT_PTP_NUM; i++) { vm->ptp[i] = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (NULL == vm->ptp[i]) + if (!vm->ptp[i]) break; } if (!i) { diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 04438f1d682d..55545e0818b5 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -46,6 +46,20 @@ config SND_HDA_INPUT_JACK Say Y here to enable the jack plugging notification via input layer. +config SND_HDA_PATCH_LOADER + bool "Support initialization patch loading for HD-audio" + depends on EXPERIMENTAL + select FW_LOADER + select SND_HDA_HWDEP + select SND_HDA_RECONFIG + help + Say Y here to allow the HD-audio driver to load a pseudo + firmware file ("patch") for overriding the BIOS setup at + start up. The "patch" file can be specified via patch module + option, such as patch=hda-init. + + This option turns on hwdep and reconfig features automatically. + config SND_HDA_CODEC_REALTEK bool "Build Realtek HD-audio codec support" default y @@ -134,6 +148,19 @@ config SND_HDA_ELD def_bool y depends on SND_HDA_CODEC_INTELHDMI +config SND_HDA_CODEC_CIRRUS + bool "Build Cirrus Logic codec support" + depends on SND_HDA_INTEL + default y + help + Say Y here to include Cirrus Logic codec support in + snd-hda-intel driver, such as CS4206. + + When the HD-audio driver is built as a module, the codec + support code is also built as another module, + snd-hda-codec-cirrus. + This module is automatically loaded at probing. + config SND_HDA_CODEC_CONEXANT bool "Build Conexant HD-audio codec support" default y diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index e3081d4586cc..315a1c4f8998 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -13,6 +13,7 @@ snd-hda-codec-analog-objs := patch_analog.o snd-hda-codec-idt-objs := patch_sigmatel.o snd-hda-codec-si3054-objs := patch_si3054.o snd-hda-codec-atihdmi-objs := patch_atihdmi.o +snd-hda-codec-cirrus-objs := patch_cirrus.o snd-hda-codec-ca0110-objs := patch_ca0110.o snd-hda-codec-conexant-objs := patch_conexant.o snd-hda-codec-via-objs := patch_via.o @@ -41,6 +42,9 @@ endif ifdef CONFIG_SND_HDA_CODEC_ATIHDMI obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o endif +ifdef CONFIG_SND_HDA_CODEC_CIRRUS +obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o +endif ifdef CONFIG_SND_HDA_CODEC_CA0110 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o endif diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index b0275a050870..3f51a981e604 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -24,6 +24,7 @@ #include <linux/workqueue.h> #include <sound/core.h> #include "hda_beep.h" +#include "hda_local.h" enum { DIGBEEP_HZ_STEP = 46875, /* 46.875 Hz */ @@ -118,6 +119,9 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) struct hda_beep *beep; int err; + if (!snd_hda_get_bool_hint(codec, "beep")) + return 0; /* disabled explicitly */ + beep = kzalloc(sizeof(*beep), GFP_KERNEL); if (beep == NULL) return -ENOMEM; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c7df01b72cac..af989f660cca 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -44,6 +44,7 @@ struct hda_vendor_id { /* codec vendor labels */ static struct hda_vendor_id hda_vendor_ids[] = { { 0x1002, "ATI" }, + { 0x1013, "Cirrus Logic" }, { 0x1057, "Motorola" }, { 0x1095, "Silicon Image" }, { 0x10de, "Nvidia" }, @@ -150,7 +151,14 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, { u32 val; - val = (u32)(codec->addr & 0x0f) << 28; + if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) || + (verb & ~0xfff) || (parm & ~0xffff)) { + printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n", + codec->addr, direct, nid, verb, parm); + return ~0; + } + + val = (u32)codec->addr << 28; val |= (u32)direct << 27; val |= (u32)nid << 20; val |= verb << 8; @@ -167,6 +175,9 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd, struct hda_bus *bus = codec->bus; int err; + if (cmd == ~0) + return -1; + if (res) *res = -1; again: @@ -291,11 +302,20 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; int i, conn_len, conns; unsigned int shift, num_elems, mask; + unsigned int wcaps; hda_nid_t prev_nid; if (snd_BUG_ON(!conn_list || max_conns <= 0)) return -EINVAL; + wcaps = get_wcaps(codec, nid); + if (!(wcaps & AC_WCAP_CONN_LIST) && + get_wcaps_type(wcaps) != AC_WID_VOL_KNB) { + snd_printk(KERN_WARNING "hda_codec: " + "connection list not available for 0x%x\n", nid); + return -EINVAL; + } + parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { /* long form */ @@ -316,6 +336,8 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, /* single connection */ parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, 0); + if (parm == -1 && codec->bus->rirb_error) + return -EIO; conn_list[0] = parm & mask; return 1; } @@ -327,9 +349,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, int range_val; hda_nid_t val, n; - if (i % num_elems == 0) + if (i % num_elems == 0) { parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, i); + if (parm == -1 && codec->bus->rirb_error) + return -EIO; + } range_val = !!(parm & (1 << (shift-1))); /* ranges */ val = parm & mask; if (val == 0) { @@ -727,8 +752,7 @@ static int read_pin_defaults(struct hda_codec *codec) for (i = 0; i < codec->num_nodes; i++, nid++) { struct hda_pincfg *pin; unsigned int wcaps = get_wcaps(codec, nid); - unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> - AC_WCAP_TYPE_SHIFT; + unsigned int wid_type = get_wcaps_type(wcaps); if (wid_type != AC_WID_PIN) continue; pin = snd_array_new(&codec->init_pins); @@ -891,7 +915,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * Returns 0 if successful, or a negative error code. */ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - int do_init, struct hda_codec **codecp) + struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -984,11 +1008,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); - if (do_init) { - err = snd_hda_codec_configure(codec); - if (err < 0) - goto error; - } snd_hda_codec_proc_new(codec); snd_hda_create_hwdep(codec); @@ -1042,6 +1061,7 @@ int snd_hda_codec_configure(struct hda_codec *codec) err = init_unsol_queue(codec->bus); return err; } +EXPORT_SYMBOL_HDA(snd_hda_codec_configure); /** * snd_hda_codec_setup_stream - set up the codec for streaming @@ -2356,16 +2376,20 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, hda_nid_t nid; int i; - snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE, + /* this delay seems necessary to avoid click noise at power-down */ + if (power_state == AC_PWRST_D3) + msleep(100); + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); - msleep(10); /* partial workaround for "azx_get_response timeout" */ + /* partial workaround for "azx_get_response timeout" */ + if (power_state == AC_PWRST_D0) + msleep(10); nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int wcaps = get_wcaps(codec, nid); if (wcaps & AC_WCAP_POWER) { - unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> - AC_WCAP_TYPE_SHIFT; + unsigned int wid_type = get_wcaps_type(wcaps); if (power_state == AC_PWRST_D3 && wid_type == AC_WID_PIN) { unsigned int pincap; @@ -2573,7 +2597,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, case 20: case 24: case 32: - if (maxbps >= 32) + if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) val |= 0x40; else if (maxbps >= 24) val |= 0x30; @@ -2700,11 +2724,12 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, bps = 20; } } - else if (streams == AC_SUPFMT_FLOAT32) { - /* should be exclusive */ + if (streams & AC_SUPFMT_FLOAT32) { formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; - bps = 32; - } else if (streams == AC_SUPFMT_AC3) { + if (!bps) + bps = 32; + } + if (streams == AC_SUPFMT_AC3) { /* should be exclusive */ /* temporary hack: we have still no proper support * for the direct AC3 stream... @@ -3102,7 +3127,7 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, tbl = q; if (tbl->value >= 0 && tbl->value < num_configs) { -#ifdef CONFIG_SND_DEBUG_DETECT +#ifdef CONFIG_SND_DEBUG_VERBOSE char tmp[10]; const char *model = NULL; if (models) @@ -3655,8 +3680,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, end_nid = codec->start_nid + codec->num_nodes; for (nid = codec->start_nid; nid < end_nid; nid++) { unsigned int wid_caps = get_wcaps(codec, nid); - unsigned int wid_type = - (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + unsigned int wid_type = get_wcaps_type(wid_caps); unsigned int def_conf; short assoc, loc; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1b75f28ed092..99552fb5f756 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -830,7 +830,8 @@ enum { int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - int do_init, struct hda_codec **codecp); + struct hda_codec **codecp); +int snd_hda_codec_configure(struct hda_codec *codec); /* * low level functions @@ -938,6 +939,13 @@ static inline void snd_hda_power_down(struct hda_codec *codec) {} #define snd_hda_codec_needs_resume(codec) 1 #endif +#ifdef CONFIG_SND_HDA_PATCH_LOADER +/* + * patch firmware + */ +int snd_hda_load_patch(struct hda_bus *bus, const char *patch); +#endif + /* * Codec modularization */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1d5797a96682..b36f6c5a92df 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -121,11 +121,17 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid if (node == NULL) return -ENOMEM; node->nid = nid; - nconns = snd_hda_get_connections(codec, nid, conn_list, - HDA_MAX_CONNECTIONS); - if (nconns < 0) { - kfree(node); - return nconns; + node->wid_caps = get_wcaps(codec, nid); + node->type = get_wcaps_type(node->wid_caps); + if (node->wid_caps & AC_WCAP_CONN_LIST) { + nconns = snd_hda_get_connections(codec, nid, conn_list, + HDA_MAX_CONNECTIONS); + if (nconns < 0) { + kfree(node); + return nconns; + } + } else { + nconns = 0; } if (nconns <= ARRAY_SIZE(node->slist)) node->conn_list = node->slist; @@ -140,8 +146,6 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid } memcpy(node->conn_list, conn_list, nconns * sizeof(hda_nid_t)); node->nconns = nconns; - node->wid_caps = get_wcaps(codec, nid); - node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; if (node->type == AC_WID_PIN) { node->pin_caps = snd_hda_query_pin_caps(codec, node->nid); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 6812fbe80fa4..cc24e6721d74 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -24,6 +24,7 @@ #include <linux/compat.h> #include <linux/mutex.h> #include <linux/ctype.h> +#include <linux/firmware.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -312,12 +313,8 @@ static ssize_t init_verbs_show(struct device *dev, return len; } -static ssize_t init_verbs_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static int parse_init_verbs(struct hda_codec *codec, const char *buf) { - struct snd_hwdep *hwdep = dev_get_drvdata(dev); - struct hda_codec *codec = hwdep->private_data; struct hda_verb *v; int nid, verb, param; @@ -331,6 +328,18 @@ static ssize_t init_verbs_store(struct device *dev, v->nid = nid; v->verb = verb; v->param = param; + return 0; +} + +static ssize_t init_verbs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int err = parse_init_verbs(codec, buf); + if (err < 0) + return err; return count; } @@ -376,19 +385,15 @@ static void remove_trail_spaces(char *str) #define MAX_HINTS 1024 -static ssize_t hints_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static int parse_hints(struct hda_codec *codec, const char *buf) { - struct snd_hwdep *hwdep = dev_get_drvdata(dev); - struct hda_codec *codec = hwdep->private_data; char *key, *val; struct hda_hint *hint; while (isspace(*buf)) buf++; if (!*buf || *buf == '#' || *buf == '\n') - return count; + return 0; if (*buf == '=') return -EINVAL; key = kstrndup_noeol(buf, 1024); @@ -411,7 +416,7 @@ static ssize_t hints_store(struct device *dev, kfree(hint->key); hint->key = key; hint->val = val; - return count; + return 0; } /* allocate a new hint entry */ if (codec->hints.used >= MAX_HINTS) @@ -424,6 +429,18 @@ static ssize_t hints_store(struct device *dev, } hint->key = key; hint->val = val; + return 0; +} + +static ssize_t hints_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int err = parse_hints(codec, buf); + if (err < 0) + return err; return count; } @@ -469,20 +486,24 @@ static ssize_t driver_pin_configs_show(struct device *dev, #define MAX_PIN_CONFIGS 32 -static ssize_t user_pin_configs_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static int parse_user_pin_configs(struct hda_codec *codec, const char *buf) { - struct snd_hwdep *hwdep = dev_get_drvdata(dev); - struct hda_codec *codec = hwdep->private_data; int nid, cfg; - int err; if (sscanf(buf, "%i %i", &nid, &cfg) != 2) return -EINVAL; if (!nid) return -EINVAL; - err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); + return snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); +} + +static ssize_t user_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int err = parse_user_pin_configs(codec, buf); if (err < 0) return err; return count; @@ -553,3 +574,180 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint); #endif /* CONFIG_SND_HDA_RECONFIG */ + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + +/* parser mode */ +enum { + LINE_MODE_NONE, + LINE_MODE_CODEC, + LINE_MODE_MODEL, + LINE_MODE_PINCFG, + LINE_MODE_VERB, + LINE_MODE_HINT, + NUM_LINE_MODES, +}; + +static inline int strmatch(const char *a, const char *b) +{ + return strnicmp(a, b, strlen(b)) == 0; +} + +/* parse the contents after the line "[codec]" + * accept only the line with three numbers, and assign the current codec + */ +static void parse_codec_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + unsigned int vendorid, subid, caddr; + struct hda_codec *codec; + + *codecp = NULL; + if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { + list_for_each_entry(codec, &bus->codec_list, list) { + if (codec->addr == caddr) { + *codecp = codec; + break; + } + } + } +} + +/* parse the contents after the other command tags, [pincfg], [verb], + * [hint] and [model] + * just pass to the sysfs helper (only when any codec was specified) + */ +static void parse_pincfg_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + parse_user_pin_configs(*codecp, buf); +} + +static void parse_verb_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + parse_init_verbs(*codecp, buf); +} + +static void parse_hint_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + parse_hints(*codecp, buf); +} + +static void parse_model_mode(char *buf, struct hda_bus *bus, + struct hda_codec **codecp) +{ + if (!*codecp) + return; + kfree((*codecp)->modelname); + (*codecp)->modelname = kstrdup(buf, GFP_KERNEL); +} + +struct hda_patch_item { + const char *tag; + void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc); +}; + +static struct hda_patch_item patch_items[NUM_LINE_MODES] = { + [LINE_MODE_CODEC] = { "[codec]", parse_codec_mode }, + [LINE_MODE_MODEL] = { "[model]", parse_model_mode }, + [LINE_MODE_VERB] = { "[verb]", parse_verb_mode }, + [LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode }, + [LINE_MODE_HINT] = { "[hint]", parse_hint_mode }, +}; + +/* check the line starting with '[' -- change the parser mode accodingly */ +static int parse_line_mode(char *buf, struct hda_bus *bus) +{ + int i; + for (i = 0; i < ARRAY_SIZE(patch_items); i++) { + if (!patch_items[i].tag) + continue; + if (strmatch(buf, patch_items[i].tag)) + return i; + } + return LINE_MODE_NONE; +} + +/* copy one line from the buffer in fw, and update the fields in fw + * return zero if it reaches to the end of the buffer, or non-zero + * if successfully copied a line + * + * the spaces at the beginning and the end of the line are stripped + */ +static int get_line_from_fw(char *buf, int size, struct firmware *fw) +{ + int len; + const char *p = fw->data; + while (isspace(*p) && fw->size) { + p++; + fw->size--; + } + if (!fw->size) + return 0; + if (size < fw->size) + size = fw->size; + + for (len = 0; len < fw->size; len++) { + if (!*p) + break; + if (*p == '\n') { + p++; + len++; + break; + } + if (len < size) + *buf++ = *p++; + } + *buf = 0; + fw->size -= len; + fw->data = p; + remove_trail_spaces(buf); + return 1; +} + +/* + * load a "patch" firmware file and parse it + */ +int snd_hda_load_patch(struct hda_bus *bus, const char *patch) +{ + int err; + const struct firmware *fw; + struct firmware tmp; + char buf[128]; + struct hda_codec *codec; + int line_mode; + struct device *dev = bus->card->dev; + + if (snd_BUG_ON(!dev)) + return -ENODEV; + err = request_firmware(&fw, patch, dev); + if (err < 0) { + printk(KERN_ERR "hda-codec: Cannot load the patch '%s'\n", + patch); + return err; + } + + tmp = *fw; + line_mode = LINE_MODE_NONE; + codec = NULL; + while (get_line_from_fw(buf, sizeof(buf) - 1, &tmp)) { + if (!*buf || *buf == '#' || *buf == '\n') + continue; + if (*buf == '[') + line_mode = parse_line_mode(buf, bus); + else if (patch_items[line_mode].parser) + patch_items[line_mode].parser(buf, bus, &codec); + } + release_firmware(fw); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_load_patch); +#endif /* CONFIG_SND_HDA_PATCH_LOADER */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 175f07a381ba..20a66f85f0a4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -61,6 +61,9 @@ static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; static int single_cmd; static int enable_msi; +#ifdef CONFIG_SND_HDA_PATCH_LOADER +static char *patch[SNDRV_CARDS]; +#endif module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -84,6 +87,10 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); module_param(enable_msi, int, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); +#ifdef CONFIG_SND_HDA_PATCH_LOADER +module_param_array(patch, charp, NULL, 0444); +MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface."); +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; @@ -1331,8 +1338,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { [AZX_DRIVER_TERA] = 1, }; -static int __devinit azx_codec_create(struct azx *chip, const char *model, - int no_init) +static int __devinit azx_codec_create(struct azx *chip, const char *model) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -1391,7 +1397,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, !no_init, &codec); + err = snd_hda_codec_new(chip->bus, c, &codec); if (err < 0) continue; codecs++; @@ -1401,7 +1407,16 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, snd_printk(KERN_ERR SFX "no codecs initialized\n"); return -ENXIO; } + return 0; +} +/* configure each codec instance */ +static int __devinit azx_codec_configure(struct azx *chip) +{ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) { + snd_hda_codec_configure(codec); + } return 0; } @@ -2284,6 +2299,30 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) } } +/* + * white-list for enable_msi + */ +static struct snd_pci_quirk msi_white_list[] __devinitdata = { + SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1), + {} +}; + +static void __devinit check_msi(struct azx *chip) +{ + const struct snd_pci_quirk *q; + + chip->msi = enable_msi; + if (chip->msi) + return; + q = snd_pci_quirk_lookup(chip->pci, msi_white_list); + if (q) { + printk(KERN_INFO + "hda_intel: msi for device %04x:%04x set to %d\n", + q->subvendor, q->subdevice, q->value); + chip->msi = q->value; + } +} + /* * constructor @@ -2318,7 +2357,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->pci = pci; chip->irq = -1; chip->driver_type = driver_type; - chip->msi = enable_msi; + check_msi(chip); chip->dev_index = dev; INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work); @@ -2526,15 +2565,32 @@ static int __devinit azx_probe(struct pci_dev *pci, return err; } + /* set this here since it's referred in snd_hda_load_patch() */ + snd_card_set_dev(card, &pci->dev); + err = azx_create(card, pci, dev, pci_id->driver_data, &chip); if (err < 0) goto out_free; card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_only[dev]); + err = azx_codec_create(chip, model[dev]); if (err < 0) goto out_free; +#ifdef CONFIG_SND_HDA_PATCH_LOADER + if (patch[dev]) { + snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n", + patch[dev]); + err = snd_hda_load_patch(chip->bus, patch[dev]); + if (err < 0) + goto out_free; + } +#endif + if (!probe_only[dev]) { + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + } /* create PCM streams */ err = snd_hda_build_pcms(chip->bus); @@ -2546,8 +2602,6 @@ static int __devinit azx_probe(struct pci_dev *pci, if (err < 0) goto out_free; - snd_card_set_dev(card, &pci->dev); - err = snd_card_register(card); if (err < 0) goto out_free; @@ -2649,11 +2703,15 @@ static struct pci_device_id azx_ids[] = { /* this entry seems still valid -- i.e. without emu20kx chip */ { PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC }, #endif - /* AMD Generic, PCI class code and Vendor ID for HD Audio */ + /* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */ { PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, .class_mask = 0xffffff, .driver_data = AZX_DRIVER_GENERIC }, + { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_GENERIC }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 83349013b4df..5f1dcc59002b 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -99,7 +99,6 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves); int snd_hda_codec_reset(struct hda_codec *codec); -int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ #define HDA_AMP_MUTE 0x80 @@ -408,6 +407,19 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) return codec->wcaps[nid - codec->start_nid]; } +/* get the widget type from widget capability bits */ +#define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT) + +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = ((chans << 1) | 1) + 1; + + return chans; +} + u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 418c5d1badaa..95f24e4729f8 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -508,17 +508,14 @@ static void print_codec_info(struct snd_info_entry *entry, unsigned int wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); - unsigned int wid_type = - (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + unsigned int wid_type = get_wcaps_type(wid_caps); hda_nid_t conn[HDA_MAX_CONNECTIONS]; int conn_len = 0; snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); if (wid_caps & AC_WCAP_STEREO) { - unsigned int chans; - chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + unsigned int chans = get_wcaps_channels(wid_caps); if (chans == 2) snd_iprintf(buffer, " Stereo"); else diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 403588c6e3f6..215e72a87113 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2982,7 +2982,8 @@ static int patch_ad1988(struct hda_codec *codec) board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST, ad1988_models, ad1988_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for AD1988, trying auto-probe from BIOS...\n"); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = AD1988_AUTO; } @@ -3702,19 +3703,29 @@ static struct hda_amp_list ad1884a_loopbacks[] = { * Port F: Internal speakers */ -static struct hda_input_mux ad1884a_laptop_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, /* port-B */ - { "Internal Mic", 0x1 }, /* port-C */ - { "Dock Mic", 0x4 }, /* port-E */ - { "Mix", 0x3 }, - }, -}; +static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); + int mute = (!ucontrol->value.integer.value[0] && + !ucontrol->value.integer.value[1]); + /* toggle GPIO1 according to the mute state */ + snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + mute ? 0x02 : 0x0); + return ret; +} static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1884a_mobile_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + }, HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), @@ -3729,36 +3740,9 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, { } /* end */ }; -static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - int mute = (!ucontrol->value.integer.value[0] && - !ucontrol->value.integer.value[1]); - /* toggle GPIO1 according to the mute state */ - snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - mute ? 0x02 : 0x0); - return ret; -} - static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ @@ -3828,6 +3812,63 @@ static int ad1884a_hp_init(struct hda_codec *codec) return 0; } +/* mute internal speaker if HP or docking HP is plugged */ +static void ad1884a_laptop_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0); + present &= AC_PINSENSE_PRESENCE; + if (!present) { + present = snd_hda_codec_read(codec, 0x12, 0, + AC_VERB_GET_PIN_SENSE, 0); + present &= AC_PINSENSE_PRESENCE; + } + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, + present ? 0x00 : 0x02); +} + +/* switch to external mic if plugged */ +static void ad1884a_laptop_automic(struct hda_codec *codec) +{ + unsigned int idx; + + if (snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE) + idx = 0; + else if (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE) + idx = 4; + else + idx = 1; + snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx); +} + +/* unsolicited event for HP jack sensing */ +static void ad1884a_laptop_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case AD1884A_HP_EVENT: + ad1884a_laptop_automute(codec); + break; + case AD1884A_MIC_EVENT: + ad1884a_laptop_automic(codec); + break; + } +} + +/* initialize jack-sensing, too */ +static int ad1884a_laptop_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1884a_laptop_automute(codec); + ad1884a_laptop_automic(codec); + return 0; +} + /* additional verbs for laptop model */ static struct hda_verb ad1884a_laptop_verbs[] = { /* Port-A (HP) pin - always unmuted */ @@ -3844,11 +3885,19 @@ static struct hda_verb ad1884a_laptop_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ + /* Port-D (docking line-out) pin - default unmuted */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* analog mix */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* unsolicited event for pin-sense */ {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, + {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, + /* allow to touch GPIO1 (for mute control) */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ { } /* end */ }; @@ -4008,6 +4057,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; @@ -4057,9 +4107,8 @@ static int patch_ad1884a(struct hda_codec *codec) spec->mixers[0] = ad1884a_laptop_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1884a_laptop_capture_source; - codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; - codec->patch_ops.init = ad1884a_hp_init; + codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event; + codec->patch_ops.init = ad1884a_laptop_init; /* set the upper-limit for mixer amp to 0dB for avoiding the * possible damage by overloading */ diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 233e4778bba9..fb684f00156b 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -141,8 +141,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec) /* FIXME: we must check ELD and change the PCM parameters dynamically */ chans = get_wcaps(codec, CVT_NID); - chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13; - chans = ((chans << 1) | 1) + 1; + chans = get_wcaps_channels(chans); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; return 0; diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 019ca7cb56d7..d08353d3bb7f 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -459,8 +459,7 @@ static void parse_input(struct hda_codec *codec) nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int wcaps = get_wcaps(codec, nid); - unsigned int type = (wcaps & AC_WCAP_TYPE) >> - AC_WCAP_TYPE_SHIFT; + unsigned int type = get_wcaps_type(wcaps); if (type != AC_WID_AUD_IN) continue; if (snd_hda_get_connections(codec, nid, &pin, 1) != 1) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c new file mode 100644 index 000000000000..8ba306856d38 --- /dev/null +++ b/sound/pci/hda/patch_cirrus.c @@ -0,0 +1,1194 @@ +/* + * HD audio interface patch for Cirrus Logic CS420x chip + * + * Copyright (c) 2009 Takashi Iwai <tiwai@suse.de> + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +/* + */ + +struct cs_spec { + int board_config; + struct auto_pin_cfg autocfg; + struct hda_multi_out multiout; + struct snd_kcontrol *vmaster_sw; + struct snd_kcontrol *vmaster_vol; + + hda_nid_t dac_nid[AUTO_CFG_MAX_OUTS]; + hda_nid_t slave_dig_outs[2]; + + unsigned int input_idx[AUTO_PIN_LAST]; + unsigned int capsrc_idx[AUTO_PIN_LAST]; + hda_nid_t adc_nid[AUTO_PIN_LAST]; + unsigned int adc_idx[AUTO_PIN_LAST]; + unsigned int num_inputs; + unsigned int cur_input; + unsigned int automic_idx; + hda_nid_t cur_adc; + unsigned int cur_adc_stream_tag; + unsigned int cur_adc_format; + hda_nid_t dig_in; + + struct hda_bind_ctls *capture_bind[2]; + + unsigned int gpio_mask; + unsigned int gpio_dir; + unsigned int gpio_data; + + struct hda_pcm pcm_rec[2]; /* PCM information */ + + unsigned int hp_detect:1; + unsigned int mic_detect:1; +}; + +/* available models */ +enum { + CS420X_MBP55, + CS420X_AUTO, + CS420X_MODELS +}; + +/* Vendor-specific processing widget */ +#define CS420X_VENDOR_NID 0x11 +#define CS_DIG_OUT1_PIN_NID 0x10 +#define CS_DIG_OUT2_PIN_NID 0x15 +#define CS_DMIC1_PIN_NID 0x12 +#define CS_DMIC2_PIN_NID 0x0e + +/* coef indices */ +#define IDX_SPDIF_STAT 0x0000 +#define IDX_SPDIF_CTL 0x0001 +#define IDX_ADC_CFG 0x0002 +/* SZC bitmask, 4 modes below: + * 0 = immediate, + * 1 = digital immediate, analog zero-cross + * 2 = digtail & analog soft-ramp + * 3 = digital soft-ramp, analog zero-cross + */ +#define CS_COEF_ADC_SZC_MASK (3 << 0) +#define CS_COEF_ADC_MIC_SZC_MODE (3 << 0) /* SZC setup for mic */ +#define CS_COEF_ADC_LI_SZC_MODE (3 << 0) /* SZC setup for line-in */ +/* PGA mode: 0 = differential, 1 = signle-ended */ +#define CS_COEF_ADC_MIC_PGA_MODE (1 << 5) /* PGA setup for mic */ +#define CS_COEF_ADC_LI_PGA_MODE (1 << 6) /* PGA setup for line-in */ +#define IDX_DAC_CFG 0x0003 +/* SZC bitmask, 4 modes below: + * 0 = Immediate + * 1 = zero-cross + * 2 = soft-ramp + * 3 = soft-ramp on zero-cross + */ +#define CS_COEF_DAC_HP_SZC_MODE (3 << 0) /* nid 0x02 */ +#define CS_COEF_DAC_LO_SZC_MODE (3 << 2) /* nid 0x03 */ +#define CS_COEF_DAC_SPK_SZC_MODE (3 << 4) /* nid 0x04 */ + +#define IDX_BEEP_CFG 0x0004 +/* 0x0008 - test reg key */ +/* 0x0009 - 0x0014 -> 12 test regs */ +/* 0x0015 - visibility reg */ + + +static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) +{ + snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + AC_VERB_SET_COEF_INDEX, idx); + return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + AC_VERB_GET_PROC_COEF, 0); +} + +static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, + unsigned int coef) +{ + snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + AC_VERB_SET_COEF_INDEX, idx); + snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + AC_VERB_SET_PROC_COEF, coef); +} + + +#define HP_EVENT 1 +#define MIC_EVENT 2 + +/* + * PCM callbacks + */ +static int cs_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, + hinfo); +} + +static int cs_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + +static int cs_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); +} + +/* + * Digital out + */ +static int cs_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int cs_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int cs_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +static int cs_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + +/* + * Analog capture + */ +static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + spec->cur_adc = spec->adc_nid[spec->cur_input]; + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + return 0; +} + +static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct cs_spec *spec = codec->spec; + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + spec->cur_adc = 0; + return 0; +} + +/* + */ +static struct hda_pcm_stream cs_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .open = cs_playback_pcm_open, + .prepare = cs_playback_pcm_prepare, + .cleanup = cs_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream cs_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .prepare = cs_capture_pcm_prepare, + .cleanup = cs_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream cs_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .ops = { + .open = cs_dig_playback_pcm_open, + .close = cs_dig_playback_pcm_close, + .prepare = cs_dig_playback_pcm_prepare, + .cleanup = cs_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream cs_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +static int cs_build_pcms(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct hda_pcm *info = spec->pcm_rec; + + codec->pcm_info = info; + codec->num_pcms = 0; + + info->name = "Cirrus Analog"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cs_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dac_nid[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = + spec->multiout.max_channels; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = cs_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = + spec->adc_nid[spec->cur_input]; + codec->num_pcms++; + + if (!spec->multiout.dig_out_nid && !spec->dig_in) + return 0; + + info++; + info->name = "Cirrus Digital"; + info->pcm_type = spec->autocfg.dig_out_type[0]; + if (!info->pcm_type) + info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->multiout.dig_out_nid) { + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + cs_pcm_digital_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dig_out_nid; + } + if (spec->dig_in) { + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + cs_pcm_digital_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; + } + codec->num_pcms++; + + return 0; +} + +/* + * parse codec topology + */ + +static hda_nid_t get_dac(struct hda_codec *codec, hda_nid_t pin) +{ + hda_nid_t dac; + if (!pin) + return 0; + if (snd_hda_get_connections(codec, pin, &dac, 1) != 1) + return 0; + return dac; +} + +static int is_ext_mic(struct hda_codec *codec, unsigned int idx) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t pin = cfg->input_pins[idx]; + unsigned int val = snd_hda_query_pin_caps(codec, pin); + if (!(val & AC_PINCAP_PRES_DETECT)) + return 0; + val = snd_hda_codec_get_pincfg(codec, pin); + return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX); +} + +static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, + unsigned int *idxp) +{ + int i; + hda_nid_t nid; + + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + hda_nid_t pins[2]; + unsigned int type; + int j, nums; + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } + } + return 0; +} + +static int is_active_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int val; + val = snd_hda_codec_get_pincfg(codec, nid); + return (get_defcfg_connect(val) != AC_JACK_PORT_NONE); +} + +static int parse_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, extra_nids; + hda_nid_t dac; + + for (i = 0; i < cfg->line_outs; i++) { + dac = get_dac(codec, cfg->line_out_pins[i]); + if (!dac) + break; + spec->dac_nid[i] = dac; + } + spec->multiout.num_dacs = i; + spec->multiout.dac_nids = spec->dac_nid; + spec->multiout.max_channels = i * 2; + + /* add HP and speakers */ + extra_nids = 0; + for (i = 0; i < cfg->hp_outs; i++) { + dac = get_dac(codec, cfg->hp_pins[i]); + if (!dac) + break; + if (!i) + spec->multiout.hp_nid = dac; + else + spec->multiout.extra_out_nid[extra_nids++] = dac; + } + for (i = 0; i < cfg->speaker_outs; i++) { + dac = get_dac(codec, cfg->speaker_pins[i]); + if (!dac) + break; + spec->multiout.extra_out_nid[extra_nids++] = dac; + } + + if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { + cfg->speaker_outs = cfg->line_outs; + memcpy(cfg->speaker_pins, cfg->line_out_pins, + sizeof(cfg->speaker_pins)); + cfg->line_outs = 0; + } + + return 0; +} + +static int parse_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t pin = cfg->input_pins[i]; + if (!pin) + continue; + spec->input_idx[spec->num_inputs] = i; + spec->capsrc_idx[i] = spec->num_inputs++; + spec->cur_input = i; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + } + if (!spec->num_inputs) + return 0; + + /* check whether the automatic mic switch is available */ + if (spec->num_inputs == 2 && + spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) { + if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) { + if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) { + spec->mic_detect = 1; + spec->automic_idx = AUTO_PIN_FRONT_MIC; + } + } else { + if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) { + spec->mic_detect = 1; + spec->automic_idx = AUTO_PIN_MIC; + } + } + } + return 0; +} + + +static int parse_digital_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + + if (!cfg->dig_outs) + return 0; + if (snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) < 1) + return 0; + spec->multiout.dig_out_nid = nid; + spec->multiout.share_spdif = 1; + if (cfg->dig_outs > 1 && + snd_hda_get_connections(codec, cfg->dig_out_pins[1], &nid, 1) > 0) { + spec->slave_dig_outs[0] = nid; + codec->slave_dig_outs = spec->slave_dig_outs; + } + return 0; +} + +static int parse_digital_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int idx; + + if (cfg->dig_in_pin) + spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx); + return 0; +} + +/* + * create mixer controls + */ + +static const char *dir_sfx[2] = { "Playback", "Capture" }; + +static int add_mute(struct hda_codec *codec, const char *name, int index, + unsigned int pval, int dir, struct snd_kcontrol **kctlp) +{ + char tmp[44]; + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_IDX(tmp, index, 0, 0, HDA_OUTPUT); + knew.private_value = pval; + snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); + *kctlp = snd_ctl_new1(&knew, codec); + return snd_hda_ctl_add(codec, *kctlp); +} + +static int add_volume(struct hda_codec *codec, const char *name, + int index, unsigned int pval, int dir, + struct snd_kcontrol **kctlp) +{ + char tmp[32]; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); + knew.private_value = pval; + snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); + *kctlp = snd_ctl_new1(&knew, codec); + return snd_hda_ctl_add(codec, *kctlp); +} + +static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) +{ + unsigned int caps; + + /* set the upper-limit for mixer amp to 0dB */ + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + caps &= ~(0x7f << AC_AMPCAP_NUM_STEPS_SHIFT); + caps |= ((caps >> AC_AMPCAP_OFFSET_SHIFT) & 0x7f) + << AC_AMPCAP_NUM_STEPS_SHIFT; + snd_hda_override_amp_caps(codec, dac, HDA_OUTPUT, caps); +} + +static int add_vmaster(struct hda_codec *codec, hda_nid_t dac) +{ + struct cs_spec *spec = codec->spec; + unsigned int tlv[4]; + int err; + + spec->vmaster_sw = + snd_ctl_make_virtual_master("Master Playback Switch", NULL); + err = snd_hda_ctl_add(codec, spec->vmaster_sw); + if (err < 0) + return err; + + snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv); + spec->vmaster_vol = + snd_ctl_make_virtual_master("Master Playback Volume", tlv); + err = snd_hda_ctl_add(codec, spec->vmaster_vol); + if (err < 0) + return err; + return 0; +} + +static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, + int num_ctls, int type) +{ + struct cs_spec *spec = codec->spec; + const char *name; + int err, index; + struct snd_kcontrol *kctl; + static char *speakers[] = { + "Front Speaker", "Surround Speaker", "Bass Speaker" + }; + static char *line_outs[] = { + "Front Line-Out", "Surround Line-Out", "Bass Line-Out" + }; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + index = 0; + switch (type) { + case AUTO_PIN_HP_OUT: + name = "Headphone"; + index = idx; + break; + case AUTO_PIN_SPEAKER_OUT: + if (num_ctls > 1) + name = speakers[idx]; + else + name = "Speaker"; + break; + default: + if (num_ctls > 1) + name = line_outs[idx]; + else + name = "Line-Out"; + break; + } + + err = add_mute(codec, name, index, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, index, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + return 0; +} + +static int build_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, err; + + for (i = 0; i < cfg->line_outs; i++) { + err = add_output(codec, get_dac(codec, cfg->line_out_pins[i]), + i, cfg->line_outs, cfg->line_out_type); + if (err < 0) + return err; + } + for (i = 0; i < cfg->hp_outs; i++) { + err = add_output(codec, get_dac(codec, cfg->hp_pins[i]), + i, cfg->hp_outs, AUTO_PIN_HP_OUT); + if (err < 0) + return err; + } + for (i = 0; i < cfg->speaker_outs; i++) { + err = add_output(codec, get_dac(codec, cfg->speaker_pins[i]), + i, cfg->speaker_outs, AUTO_PIN_SPEAKER_OUT); + if (err < 0) + return err; + } + return 0; +} + +/* + */ + +static struct snd_kcontrol_new cs_capture_ctls[] = { + HDA_BIND_SW("Capture Switch", 0), + HDA_BIND_VOL("Capture Volume", 0), +}; + +static int change_cur_input(struct hda_codec *codec, unsigned int idx, + int force) +{ + struct cs_spec *spec = codec->spec; + + if (spec->cur_input == idx && !force) + return 0; + if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) { + /* stream is running, let's swap the current ADC */ + snd_hda_codec_cleanup_stream(codec, spec->cur_adc); + spec->cur_adc = spec->adc_nid[idx]; + snd_hda_codec_setup_stream(codec, spec->cur_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + } + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[idx]); + spec->cur_input = idx; + return 1; +} + +static int cs_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + unsigned int idx; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = spec->num_inputs; + if (uinfo->value.enumerated.item >= spec->num_inputs) + uinfo->value.enumerated.item = spec->num_inputs - 1; + idx = spec->input_idx[uinfo->value.enumerated.item]; + strcpy(uinfo->value.enumerated.name, auto_pin_cfg_labels[idx]); + return 0; +} + +static int cs_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->capsrc_idx[spec->cur_input]; + return 0; +} + +static int cs_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + unsigned int idx = ucontrol->value.enumerated.item[0]; + + if (idx >= spec->num_inputs) + return -EINVAL; + idx = spec->input_idx[idx]; + return change_cur_input(codec, idx, 0); +} + +static struct snd_kcontrol_new cs_capture_source = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs_capture_source_info, + .get = cs_capture_source_get, + .put = cs_capture_source_put, +}; + +static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, + struct hda_ctl_ops *ops) +{ + struct cs_spec *spec = codec->spec; + struct hda_bind_ctls *bind; + int i, n; + + bind = kzalloc(sizeof(*bind) + sizeof(long) * (spec->num_inputs + 1), + GFP_KERNEL); + if (!bind) + return NULL; + bind->ops = ops; + n = 0; + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!spec->adc_nid[i]) + continue; + bind->values[n++] = + HDA_COMPOSE_AMP_VAL(spec->adc_nid[i], 3, + spec->adc_idx[i], HDA_INPUT); + } + return bind; +} + +static int build_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int i, err; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + + if (spec->num_inputs > 1 && !spec->mic_detect) { + err = snd_hda_ctl_add(codec, + snd_ctl_new1(&cs_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* + */ + +static int build_digital_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + if (!spec->multiout.dig_out_nid) + return 0; + + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); + if (err < 0) + return err; + return 0; +} + +static int build_digital_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + if (spec->dig_in) + return snd_hda_create_spdif_in_ctls(codec, spec->dig_in); + return 0; +} + +/* + * auto-mute and auto-mic switching + */ + +static void cs_automute(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps, present, hp_present; + hda_nid_t nid; + int i; + + hp_present = 0; + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) + continue; + if (caps & AC_PINCAP_TRIG_REQ) + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + hp_present |= (present & AC_PINSENSE_PRESENCE) != 0; + if (hp_present) + break; + } + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + hp_present ? 0 : PIN_OUT); + } + if (spec->board_config == CS420X_MBP55) { + unsigned int gpio = hp_present ? 0x02 : 0x08; + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, gpio); + } +} + +static void cs_automic(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + unsigned int caps, present; + + nid = cfg->input_pins[spec->automic_idx]; + caps = snd_hda_query_pin_caps(codec, nid); + if (caps & AC_PINCAP_TRIG_REQ) + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); + if (present & AC_PINSENSE_PRESENCE) + change_cur_input(codec, spec->automic_idx, 0); + else { + unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ? + AUTO_PIN_FRONT_MIC : AUTO_PIN_MIC; + change_cur_input(codec, imic, 0); + } +} + +/* + */ + +static void init_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + /* mute first */ + for (i = 0; i < spec->multiout.num_dacs; i++) + snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (spec->multiout.hp_nid) + snd_hda_codec_write(codec, spec->multiout.hp_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { + if (!spec->multiout.extra_out_nid[i]) + break; + snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + } + + /* set appropriate pin controls */ + for (i = 0; i < cfg->line_outs; i++) + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | HP_EVENT); + spec->hp_detect = 1; + } + } + for (i = 0; i < cfg->speaker_outs; i++) + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + if (spec->hp_detect) + cs_automute(codec); +} + +static void init_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int coef; + int i; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + unsigned int ctl; + hda_nid_t pin = cfg->input_pins[i]; + if (!pin || !spec->adc_nid[i]) + continue; + /* set appropriate pin control and mute first */ + ctl = PIN_IN; + if (i <= AUTO_PIN_FRONT_MIC) { + unsigned int caps = snd_hda_query_pin_caps(codec, pin); + caps >>= AC_PINCAP_VREF_SHIFT; + if (caps & AC_PINCAP_VREF_80) + ctl = PIN_VREF80; + } + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + snd_hda_codec_write(codec, spec->adc_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(spec->adc_idx[i])); + if (spec->mic_detect && spec->automic_idx == i) + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | MIC_EVENT); + } + change_cur_input(codec, spec->cur_input, 1); + if (spec->mic_detect) + cs_automic(codec); + + coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + if (is_active_pin(codec, CS_DMIC2_PIN_NID)) + coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ + if (is_active_pin(codec, CS_DMIC1_PIN_NID)) + coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 + * No effect if SPDIF_OUT2 is slected in + * IDX_SPDIF_CTL. + */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); +} + +static struct hda_verb cs_coef_init_verbs[] = { + {0x11, AC_VERB_SET_PROC_STATE, 1}, + {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_PROC_COEF, + (0x002a /* DAC1/2/3 SZCMode Soft Ramp */ + | 0x0040 /* Mute DACs on FIFO error */ + | 0x1000 /* Enable DACs High Pass Filter */ + | 0x0400 /* Disable Coefficient Auto increment */ + )}, + /* Beep */ + {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */ + + {} /* terminator */ +}; + +/* SPDIF setup */ +static void init_digital(struct hda_codec *codec) +{ + unsigned int coef; + + coef = 0x0002; /* SRC_MUTE soft-mute on SPDIF (if no lock) */ + coef |= 0x0008; /* Replace with mute on error */ + if (is_active_pin(codec, CS_DIG_OUT2_PIN_NID)) + coef |= 0x4000; /* RX to TX1 or TX2 Loopthru / SPDIF2 + * SPDIF_OUT2 is shared with GPIO1 and + * DMIC_SDA2. + */ + cs_vendor_coef_set(codec, IDX_SPDIF_CTL, coef); +} + +static int cs_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs_coef_init_verbs); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_output(codec); + init_input(codec); + init_digital(codec); + return 0; +} + +static int cs_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_output(codec); + if (err < 0) + return err; + err = build_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + err = build_digital_input(codec); + if (err < 0) + return err; + return cs_init(codec); +} + +static void cs_free(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + kfree(spec->capture_bind[0]); + kfree(spec->capture_bind[1]); + kfree(codec->spec); +} + +static void cs_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x7f) { + case HP_EVENT: + cs_automute(codec); + break; + case MIC_EVENT: + cs_automic(codec); + break; + } +} + +static struct hda_codec_ops cs_patch_ops = { + .build_controls = cs_build_controls, + .build_pcms = cs_build_pcms, + .init = cs_init, + .free = cs_free, + .unsol_event = cs_unsol_event, +}; + +static int cs_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + err = parse_output(codec); + if (err < 0) + return err; + err = parse_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + err = parse_digital_input(codec); + if (err < 0) + return err; + return 0; +} + +static const char *cs420x_models[CS420X_MODELS] = { + [CS420X_MBP55] = "mbp55", + [CS420X_AUTO] = "auto", +}; + + +static struct snd_pci_quirk cs420x_cfg_tbl[] = { + SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), + {} /* terminator */ +}; + +struct cs_pincfg { + hda_nid_t nid; + u32 val; +}; + +static struct cs_pincfg mbp55_pincfgs[] = { + { 0x09, 0x012b4030 }, + { 0x0a, 0x90100121 }, + { 0x0b, 0x90100120 }, + { 0x0c, 0x400000f0 }, + { 0x0d, 0x90a00110 }, + { 0x0e, 0x400000f0 }, + { 0x0f, 0x400000f0 }, + { 0x10, 0x014be040 }, + { 0x12, 0x400000f0 }, + { 0x15, 0x400000f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { + [CS420X_MBP55] = mbp55_pincfgs, +}; + +static void fix_pincfg(struct hda_codec *codec, int model) +{ + const struct cs_pincfg *cfg = cs_pincfgs[model]; + if (!cfg) + return; + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} + + +static int patch_cs420x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->board_config = + snd_hda_check_board_config(codec, CS420X_MODELS, + cs420x_models, cs420x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config); + + switch (spec->board_config) { + case CS420X_MBP55: + /* GPIO1 = headphones */ + /* GPIO3 = speakers */ + spec->gpio_mask = 0x0a; + spec->gpio_dir = 0x0a; + break; + } + + err = cs_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + + +/* + * patch entries + */ +static struct hda_codec_preset snd_hda_preset_cirrus[] = { + { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, + { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + {} /* terminator */ +}; + +MODULE_ALIAS("snd-hda-codec-id:10134206"); +MODULE_ALIAS("snd-hda-codec-id:10134207"); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); + +static struct hda_codec_preset_list cirrus_list = { + .preset = snd_hda_preset_cirrus, + .owner = THIS_MODULE, +}; + +static int __init patch_cirrus_init(void) +{ + return snd_hda_add_codec_preset(&cirrus_list); +} + +static void __exit patch_cirrus_exit(void) +{ + snd_hda_delete_codec_preset(&cirrus_list); +} + +module_init(patch_cirrus_init) +module_exit(patch_cirrus_exit) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index c921264bbd71..780e1a72114a 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -635,7 +635,8 @@ static int patch_cmi9880(struct hda_codec *codec) cmi9880_models, cmi9880_cfg_tbl); if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for CMI9880\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); spec->board_config = CMI_AUTO; /* try everything */ } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ac868c59f9e3..9d899eda44d7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -108,6 +108,8 @@ struct conexant_spec { struct hda_input_mux private_imux; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + unsigned int dell_automute; + unsigned int port_d_mode; }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -1908,6 +1910,480 @@ static int patch_cxt5051(struct hda_codec *codec) return 0; } +/* Conexant 5066 specific */ + +static hda_nid_t cxt5066_dac_nids[1] = { 0x10 }; +static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 }; +static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 }; +#define CXT5066_SPDIF_OUT 0x21 + +static struct hda_channel_mode cxt5066_modes[1] = { + { 2, NULL }, +}; + +static void cxt5066_update_speaker(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int pinctl; + + snd_printdd("CXT5066: update speaker, hp_present=%d\n", + spec->hp_present); + + /* Port A (HP) */ + pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0; + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* Port D (HP/LO) */ + pinctl = ((spec->hp_present & 2) && spec->cur_eapd) + ? spec->port_d_mode : 0; + snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + /* CLASS_D AMP */ + pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); + + if (spec->dell_automute) { + /* DELL AIO Port Rule: PortA > PortD > IntSpk */ + pinctl = (!(spec->hp_present & 1) && spec->cur_eapd) + ? PIN_OUT : 0; + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + } +} + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + if (!cxt_eapd_put(kcontrol, ucontrol)) + return 0; + + cxt5066_update_speaker(codec); + return 1; +} + +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_automic(struct hda_codec *codec) +{ + static struct hda_verb ext_mic_present[] = { + /* enable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to external mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + + /* disable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + /* enable internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* switch to internal mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + +/* mute internal speaker if HP is plugged */ +static void cxt5066_hp_automute(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int portA, portD; + + /* Port A */ + portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + /* Port D */ + portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE) << 1; + + spec->hp_present = !!(portA | portD); + snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", + portA, portD, spec->hp_present); + cxt5066_update_speaker(codec); +} + +/* unsolicited event for jack sensing */ +static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_automic(codec); + break; + } +} + +static const struct hda_input_mux cxt5066_analog_mic_boost = { + .num_items = 5, + .items = { + { "0dB", 0 }, + { "10dB", 1 }, + { "20dB", 2 }, + { "30dB", 3 }, + { "40dB", 4 }, + }, +}; + +static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo); +} + +static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = snd_hda_codec_read(codec, 0x17, 0, + AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT); + + ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + return 0; +} + +static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + + if (!imux->num_items) + return 0; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + imux->items[idx].index); + + return 1; +} + +static struct hda_input_mux cxt5066_capture_source = { + .num_items = 4, + .items = { + { "Mic B", 0 }, + { "Mic C", 1 }, + { "Mic E", 2 }, + { "Mic F", 3 }, + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_vol_others = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls cxt5066_bind_capture_sw_others = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new cxt5066_mixer_master[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = snd_hda_mixer_amp_volume_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + /* offset by 28 volume steps to limit minimum gain to -46dB */ + .private_value = + HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28), + }, + {} +}; + +static struct snd_kcontrol_new cxt5066_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5066_hp_master_sw_put, + .private_value = 0x1d, + }, + + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Mic Boost Capture Enum", + .info = cxt5066_mic_boost_mux_enum_info, + .get = cxt5066_mic_boost_mux_enum_get, + .put = cxt5066_mic_boost_mux_enum_put, + }, + + HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), + HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others), + {} +}; + +static struct hda_verb cxt5066_init_verbs[] = { + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* no digital microphone support yet */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* not handling these yet */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_olpc[] = { + /* Port A: headphones */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: external microphone */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port C: internal microphone */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + + /* Port D: unused */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port E: unused, but has primary EAPD */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* Port F: unused */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Audio input selectors */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable unsolicited events for Port A and B */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + +static struct hda_verb cxt5066_init_verbs_portd_lo[] = { + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int cxt5066_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + if (codec->patch_ops.unsol_event) { + cxt5066_hp_automute(codec); + cxt5066_automic(codec); + } + return 0; +} + +enum { + CXT5066_LAPTOP, /* Laptops w/ EAPD support */ + CXT5066_DELL_LAPTOP, /* Dell Laptop */ + CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ + CXT5066_MODELS +}; + +static const char *cxt5066_models[CXT5066_MODELS] = { + [CXT5066_LAPTOP] = "laptop", + [CXT5066_DELL_LAPTOP] = "dell-laptop", + [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", +}; + +static struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", + CXT5066_LAPTOP), + SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", + CXT5066_DELL_LAPTOP), + {} +}; + +static int patch_cxt5066(struct hda_codec *codec) +{ + struct conexant_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + codec->patch_ops = conexant_patch_ops; + codec->patch_ops.init = cxt5066_init; + + spec->dell_automute = 0; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids); + spec->multiout.dac_nids = cxt5066_dac_nids; + spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT; + spec->num_adc_nids = 1; + spec->adc_nids = cxt5066_adc_nids; + spec->capsrc_nids = cxt5066_capsrc_nids; + spec->input_mux = &cxt5066_capture_source; + + spec->port_d_mode = PIN_HP; + + spec->num_init_verbs = 1; + spec->init_verbs[0] = cxt5066_init_verbs; + spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes); + spec->channel_mode = cxt5066_modes; + spec->cur_adc = 0; + spec->cur_adc_idx = 0; + + board_config = snd_hda_check_board_config(codec, CXT5066_MODELS, + cxt5066_models, cxt5066_cfg_tbl); + switch (board_config) { + default: + case CXT5066_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + break; + case CXT5066_DELL_LAPTOP: + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + + spec->port_d_mode = PIN_OUT; + spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo; + spec->num_init_verbs++; + spec->dell_automute = 1; + break; + case CXT5066_OLPC_XO_1_5: + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[0] = cxt5066_init_verbs_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->port_d_mode = 0; + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; + } + + return 0; +} /* */ @@ -1919,12 +2395,15 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5047 }, { .id = 0x14f15051, .name = "CX20561 (Hermosa)", .patch = patch_cxt5051 }, + { .id = 0x14f15066, .name = "CX20582 (Pebble)", + .patch = patch_cxt5066 }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:14f15045"); MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); +MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index fcc77fec4487..01a18ed475ac 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -33,8 +33,8 @@ #include "hda_codec.h" #include "hda_local.h" -#define CVT_NID 0x02 /* audio converter */ -#define PIN_NID 0x03 /* HDMI output pin */ +static hda_nid_t cvt_nid; /* audio converter */ +static hda_nid_t pin_nid; /* HDMI output pin */ #define INTEL_HDMI_EVENT_TAG 0x08 @@ -44,30 +44,6 @@ struct intel_hdmi_spec { struct hdmi_eld sink_eld; }; -static struct hda_verb pinout_enable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {} /* terminator */ -}; - -static struct hda_verb unsolicited_response_verb[] = { - {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | - INTEL_HDMI_EVENT_TAG}, - {} -}; - -static struct hda_verb def_chan_map[] = { - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66}, - {CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77}, - {} -}; - - struct hdmi_audio_infoframe { u8 type; /* 0x84 */ u8 ver; /* 0x01 */ @@ -244,11 +220,12 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, static void hdmi_enable_output(struct hda_codec *codec) { /* Unmute */ - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); /* Enable pin out */ - snd_hda_sequence_write(codec, pinout_enable_verb); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); } /* @@ -256,8 +233,8 @@ static void hdmi_enable_output(struct hda_codec *codec) */ static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_BEST); } @@ -266,20 +243,20 @@ static void hdmi_start_infoframe_trans(struct hda_codec *codec) */ static void hdmi_stop_infoframe_trans(struct hda_codec *codec) { - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) { - return 1 + snd_hda_codec_read(codec, CVT_NID, 0, + return 1 + snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } static void hdmi_set_channel_count(struct hda_codec *codec, int chs) { - snd_hda_codec_write(codec, CVT_NID, 0, + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); if (chs != hdmi_get_channel_count(codec)) @@ -294,7 +271,7 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec) int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, CVT_NID, 0, + slot = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", slot >> 4, slot & 0x7); @@ -307,7 +284,7 @@ static void hdmi_parse_eld(struct hda_codec *codec) struct intel_hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->sink_eld; - if (!snd_hdmi_get_eld(eld, codec, PIN_NID)) + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) snd_hdmi_show_eld(eld); } @@ -322,11 +299,11 @@ static void hdmi_debug_dip_size(struct hda_codec *codec) int i; int size; - size = snd_hdmi_get_eld_size(codec, PIN_NID); + size = snd_hdmi_get_eld_size(codec, pin_nid); printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, PIN_NID, 0, + size = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, i); printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); } @@ -340,15 +317,15 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec) int size; int pi, bi; for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, PIN_NID, 0, + size = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_SIZE, i); if (size == 0) continue; - hdmi_set_dip_index(codec, PIN_NID, i, 0x0); + hdmi_set_dip_index(codec, pin_nid, i, 0x0); for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, PIN_NID, 0x0); - hdmi_get_dip_index(codec, PIN_NID, &pi, &bi); + hdmi_write_dip_byte(codec, pin_nid, 0x0); + hdmi_get_dip_index(codec, pin_nid, &pi, &bi); if (pi != i) snd_printd(KERN_INFO "dip index %d: %d != %d\n", bi, pi, i); @@ -376,9 +353,9 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec, sum += params[i]; ai->checksum = - sum; - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < sizeof(ai); i++) - hdmi_write_dip_byte(codec, PIN_NID, params[i]); + hdmi_write_dip_byte(codec, pin_nid, params[i]); } /* @@ -465,6 +442,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, static void hdmi_setup_channel_mapping(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { + int i; + if (!ai->CA) return; @@ -473,7 +452,11 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, * ALSA sequence is front/surr/clfe/side? */ - snd_hda_sequence_write(codec, def_chan_map); + for (i = 0; i < 8; i++) + snd_hda_codec_write(codec, cvt_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + (i << 4) | i); + hdmi_debug_channel_mapping(codec); } @@ -597,7 +580,6 @@ static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, .channels_max = 8, - .nid = CVT_NID, /* NID to query formats and rates and setup streams */ .ops = { .open = intel_hdmi_playback_pcm_open, .close = intel_hdmi_playback_pcm_close, @@ -613,6 +595,9 @@ static int intel_hdmi_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; + /* NID to query formats and rates and setup streams */ + intel_hdmi_pcm_playback.nid = cvt_nid; + info->name = "INTEL HDMI"; info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; @@ -636,8 +621,9 @@ static int intel_hdmi_init(struct hda_codec *codec) { hdmi_enable_output(codec); - snd_hda_sequence_write(codec, unsolicited_response_verb); - + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | INTEL_HDMI_EVENT_TAG); return 0; } @@ -657,7 +643,7 @@ static struct hda_codec_ops intel_hdmi_patch_ops = { .unsol_event = intel_hdmi_unsol_event, }; -static int patch_intel_hdmi(struct hda_codec *codec) +static int do_patch_intel_hdmi(struct hda_codec *codec) { struct intel_hdmi_spec *spec; @@ -667,7 +653,7 @@ static int patch_intel_hdmi(struct hda_codec *codec) spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = CVT_NID; + spec->multiout.dig_out_nid = cvt_nid; codec->spec = spec; codec->patch_ops = intel_hdmi_patch_ops; @@ -679,12 +665,27 @@ static int patch_intel_hdmi(struct hda_codec *codec) return 0; } +static int patch_intel_hdmi(struct hda_codec *codec) +{ + cvt_nid = 0x02; + pin_nid = 0x03; + return do_patch_intel_hdmi(codec); +} + +static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec) +{ + cvt_nid = 0x02; + pin_nid = 0x04; + return do_patch_intel_hdmi(codec); +} + static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi }, { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, + { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; @@ -694,6 +695,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862801"); MODULE_ALIAS("snd-hda-codec-id:80862802"); MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); +MODULE_ALIAS("snd-hda-codec-id:80860054"); MODULE_ALIAS("snd-hda-codec-id:10951392"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index f5792e2eea82..c8435c9a97f9 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -377,6 +377,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, + { .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, @@ -385,6 +386,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { }; MODULE_ALIAS("snd-hda-codec-id:10de0002"); +MODULE_ALIAS("snd-hda-codec-id:10de0003"); MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 30eeb304351c..7ed47f66ddd1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -208,12 +208,6 @@ enum { ALC885_MBP3, ALC885_MB5, ALC885_IMAC24, - ALC882_AUTO, - ALC882_MODEL_LAST, -}; - -/* ALC883 models */ -enum { ALC883_3ST_2ch_DIG, ALC883_3ST_6ch_DIG, ALC883_3ST_6ch, @@ -226,6 +220,7 @@ enum { ALC888_ACER_ASPIRE_4930G, ALC888_ACER_ASPIRE_6530G, ALC888_ACER_ASPIRE_8930G, + ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, @@ -237,17 +232,20 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, + ALC883_CLEVO_M540R, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, ALC888_FUJITSU_XA3530, ALC883_3ST_6ch_INTEL, + ALC889A_INTEL, + ALC889_INTEL, ALC888_ASUS_M90V, ALC888_ASUS_EEE1601, ALC889A_MB31, ALC1200_ASUS_P5Q, ALC883_SONY_VAIO_TT, - ALC883_AUTO, - ALC883_MODEL_LAST, + ALC882_AUTO, + ALC882_MODEL_LAST, }; /* for GPIO Poll */ @@ -262,6 +260,14 @@ enum { ALC_INIT_GPIO3, }; +struct alc_mic_route { + hda_nid_t pin; + unsigned char mux_idx; + unsigned char amix_idx; +}; + +#define MUX_IDX_UNDEF ((unsigned char)-1) + struct alc_spec { /* codec parameterization */ struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -304,6 +310,8 @@ struct alc_spec { unsigned int num_mux_defs; const struct hda_input_mux *input_mux; unsigned int cur_mux[3]; + struct alc_mic_route ext_mic; + struct alc_mic_route int_mic; /* channel model */ const struct hda_channel_mode *channel_mode; @@ -320,6 +328,8 @@ struct alc_spec { struct snd_array kctls; struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS]; + hda_nid_t private_capsrc_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ void (*init_hook)(struct hda_codec *codec); @@ -329,6 +339,7 @@ struct alc_spec { unsigned int sense_updated: 1; unsigned int jack_present: 1; unsigned int master_sw: 1; + unsigned int auto_mic:1; /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -370,6 +381,7 @@ struct alc_config_preset { unsigned int num_mux_defs; const struct hda_input_mux *input_mux; void (*unsol_event)(struct hda_codec *, unsigned int); + void (*setup)(struct hda_codec *); void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; @@ -417,7 +429,7 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_AUD_MIX) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; @@ -842,9 +854,10 @@ static void print_realtek_coef(struct snd_info_buffer *buffer, /* * set up from the preset table */ -static void setup_preset(struct alc_spec *spec, +static void setup_preset(struct hda_codec *codec, const struct alc_config_preset *preset) { + struct alc_spec *spec = codec->spec; int i; for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) @@ -886,6 +899,9 @@ static void setup_preset(struct alc_spec *spec, #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = preset->loopbacks; #endif + + if (preset->setup) + preset->setup(codec); } /* Enable GPIO mask and set output */ @@ -965,30 +981,64 @@ static void alc_automute_pin(struct hda_codec *codec) } } -#if 0 /* it's broken in some cases -- temporarily disabled */ +static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, + hda_nid_t nid) +{ + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int i, nums; + + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); + for (i = 0; i < nums; i++) + if (conn[i] == nid) + return i; + return -1; +} + static void alc_mic_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - unsigned int mic_nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; - unsigned int fmic_nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; - unsigned int mix_nid = spec->capsrc_nids[0]; - unsigned int capsrc_idx_mic, capsrc_idx_fmic; - - capsrc_idx_mic = mic_nid - 0x18; - capsrc_idx_fmic = fmic_nid - 0x18; - present = snd_hda_codec_read(codec, mic_nid, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (capsrc_idx_mic << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (capsrc_idx_fmic << 8) | (present ? 0x80 : 0)); - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + struct alc_mic_route *dead, *alive; + unsigned int present, type; + hda_nid_t cap_nid; + + if (!spec->auto_mic) + return; + if (!spec->int_mic.pin || !spec->ext_mic.pin) + return; + if (snd_BUG_ON(!spec->adc_nids)) + return; + + cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0]; + + present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0, + AC_VERB_GET_PIN_SENSE, 0); + present &= AC_PINSENSE_PRESENCE; + if (present) { + alive = &spec->ext_mic; + dead = &spec->int_mic; + } else { + alive = &spec->int_mic; + dead = &spec->ext_mic; + } + + type = get_wcaps_type(get_wcaps(codec, cap_nid)); + if (type == AC_WID_AUD_MIX) { + /* Matrix-mixer style (e.g. ALC882) */ + snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT, + alive->mux_idx, + HDA_AMP_MUTE, 0); + snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT, + dead->mux_idx, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* MUX style (e.g. ALC880) */ + snd_hda_codec_write_cache(codec, cap_nid, 0, + AC_VERB_SET_CONNECT_SEL, + alive->mux_idx); + } + + /* FIXME: analog mixer */ } -#else -#define alc_mic_automute(codec) do {} while(0) /* NOP */ -#endif /* disabled */ /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) @@ -1031,6 +1081,16 @@ static void alc888_coef_init(struct hda_codec *codec) AC_VERB_SET_PROC_COEF, 0x3030); } +static void alc889_coef_init(struct hda_codec *codec) +{ + unsigned int tmp; + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); + tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010); +} + static void alc_auto_init_amp(struct hda_codec *codec, int type) { unsigned int tmp; @@ -1088,15 +1148,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0885: case 0x10ec0887: case 0x10ec0889: - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 7); - tmp = snd_hda_codec_read(codec, 0x20, 0, - AC_VERB_GET_PROC_COEF, 0); - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 7); - snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_PROC_COEF, - tmp | 0x2010); + alc889_coef_init(codec); break; case 0x10ec0888: alc888_coef_init(codec); @@ -1142,6 +1194,55 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->unsol_event = alc_sku_unsol_event; } +static void alc_init_auto_mic(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t fixed, ext; + int i; + + /* there must be only two mic inputs exclusively */ + for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++) + if (cfg->input_pins[i]) + return; + + fixed = ext = 0; + for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) { + hda_nid_t nid = cfg->input_pins[i]; + unsigned int defcfg; + if (!nid) + return; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + switch (get_defcfg_connect(defcfg)) { + case AC_JACK_PORT_FIXED: + if (fixed) + return; /* already occupied */ + fixed = nid; + break; + case AC_JACK_PORT_COMPLEX: + if (ext) + return; /* already occupied */ + ext = nid; + break; + default: + return; /* invalid entry */ + } + } + if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) + return; /* no unsol support */ + snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n", + ext, fixed); + spec->ext_mic.pin = ext; + spec->int_mic.pin = fixed; + spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */ + spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */ + spec->auto_mic = 1; + snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_MIC_EVENT); + spec->unsol_event = alc_sku_unsol_event; +} + /* check subsystem ID and set up device-specific initialization; * return 1 if initialized, 0 if invalid SSID */ @@ -1243,6 +1344,7 @@ do_sku: } alc_init_auto_hp(codec); + alc_init_auto_mic(codec); return 1; } @@ -1255,6 +1357,7 @@ static void alc_ssid_check(struct hda_codec *codec, "Enable default setup for auto mode as fallback\n"); spec->init_amp = ALC_INIT_DEFAULT; alc_init_auto_hp(codec); + alc_init_auto_mic(codec); } } @@ -1436,7 +1539,25 @@ static void alc_automute_amp_unsol_event(struct hda_codec *codec, alc_automute_amp(codec); } -static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) +static void alc889_automute_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; + spec->autocfg.speaker_pins[3] = 0x19; + spec->autocfg.speaker_pins[4] = 0x1a; +} + +static void alc889_intel_init_hook(struct hda_codec *codec) +{ + alc889_coef_init(codec); + alc_automute_amp(codec); +} + +static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1444,7 +1565,6 @@ static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - alc_automute_amp(codec); } /* @@ -1643,16 +1763,15 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { { } /* end */ }; -static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) +static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_amp(codec); } -static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec) +static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1660,10 +1779,9 @@ static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - alc_automute_amp(codec); } -static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) +static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1671,7 +1789,6 @@ static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - alc_automute_amp(codec); } /* @@ -2651,13 +2768,17 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec) snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } -static void alc880_uniwill_init_hook(struct hda_codec *codec) +static void alc880_uniwill_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; +} + +static void alc880_uniwill_init_hook(struct hda_codec *codec) +{ alc_automute_amp(codec); alc880_uniwill_mic_automute(codec); } @@ -2678,13 +2799,12 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, } } -static void alc880_uniwill_p53_init_hook(struct hda_codec *codec) +static void alc880_uniwill_p53_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - alc_automute_amp(codec); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -2947,13 +3067,12 @@ static struct hda_verb alc880_lg_init_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_init_hook(struct hda_codec *codec) +static void alc880_lg_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - alc_automute_amp(codec); } /* @@ -3032,13 +3151,12 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_lw_init_hook(struct hda_codec *codec) +static void alc880_lg_lw_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_amp(codec); } static struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -3104,13 +3222,12 @@ static void alc880_medion_rim_unsol_event(struct hda_codec *codec, alc880_medion_rim_automute(codec); } -static void alc880_medion_rim_init_hook(struct hda_codec *codec) +static void alc880_medion_rim_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - alc880_medion_rim_automute(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -3977,7 +4094,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_f1734_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_init_hook, + .setup = alc880_uniwill_p53_setup, + .init_hook = alc_automute_amp, }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, @@ -4054,6 +4172,7 @@ static struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_unsol_event, + .setup = alc880_uniwill_setup, .init_hook = alc880_uniwill_init_hook, }, [ALC880_UNIWILL_P53] = { @@ -4066,7 +4185,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_init_hook, + .setup = alc880_uniwill_p53_setup, + .init_hook = alc_automute_amp, }, [ALC880_FUJITSU] = { .mixers = { alc880_fujitsu_mixer }, @@ -4080,7 +4200,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, - .init_hook = alc880_uniwill_p53_init_hook, + .setup = alc880_uniwill_p53_setup, + .init_hook = alc_automute_amp, }, [ALC880_CLEVO] = { .mixers = { alc880_three_stack_mixer }, @@ -4106,7 +4227,8 @@ static struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_lg_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc880_lg_init_hook, + .setup = alc880_lg_setup, + .init_hook = alc_automute_amp, #ifdef CONFIG_SND_HDA_POWER_SAVE .loopbacks = alc880_lg_loopbacks, #endif @@ -4122,7 +4244,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_lg_lw_modes, .input_mux = &alc880_lg_lw_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc880_lg_lw_init_hook, + .setup = alc880_lg_lw_setup, + .init_hook = alc_automute_amp, }, [ALC880_MEDION_RIM] = { .mixers = { alc880_medion_rim_mixer }, @@ -4136,7 +4259,8 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_medion_rim_capture_source, .unsol_event = alc880_medion_rim_unsol_event, - .init_hook = alc880_medion_rim_init_hook, + .setup = alc880_medion_rim_setup, + .init_hook = alc880_medion_rim_automute, }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { @@ -4189,8 +4313,6 @@ static int add_control(struct alc_spec *spec, int type, const char *name, #define alc880_fixed_pin_idx(nid) ((nid) - 0x14) #define alc880_is_multi_pin(nid) ((nid) >= 0x18) #define alc880_multi_pin_idx(nid) ((nid) - 0x18) -#define alc880_is_input_pin(nid) ((nid) >= 0x18) -#define alc880_input_pin_idx(nid) ((nid) - 0x18) #define alc880_idx_to_dac(nid) ((nid) + 0x02) #define alc880_dac_to_idx(nid) ((nid) - 0x02) #define alc880_idx_to_mixer(nid) ((nid) + 0x0c) @@ -4278,13 +4400,19 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } else { - sprintf(name, "%s Playback Volume", chname[i]); + const char *pfx; + if (cfg->line_outs == 1 && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + else + pfx = chname[i]; + sprintf(name, "%s Playback Volume", pfx); err = add_control(spec, ALC_CTL_WIDGET_VOL, name, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", chname[i]); + sprintf(name, "%s Playback Switch", pfx); err = add_control(spec, ALC_CTL_BIND_MUTE, name, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); @@ -4358,31 +4486,61 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, return 0; } +static int alc_is_input_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); + return (pincap & AC_PINCAP_IN) != 0; +} + /* create playback/capture controls for input pins */ -static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, - const struct auto_pin_cfg *cfg) +static int alc_auto_create_input_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg, + hda_nid_t mixer, + hda_nid_t cap1, hda_nid_t cap2) { + struct alc_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { - if (alc880_is_input_pin(cfg->input_pins[i])) { - idx = alc880_input_pin_idx(cfg->input_pins[i]); - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], - idx, 0x0b); - if (err < 0) - return err; + hda_nid_t pin; + + pin = cfg->input_pins[i]; + if (!alc_is_input_pin(codec, pin)) + continue; + + if (mixer) { + idx = get_connection_index(codec, mixer, pin); + if (idx >= 0) { + err = new_analog_input(spec, pin, + auto_pin_cfg_labels[i], + idx, mixer); + if (err < 0) + return err; + } + } + + if (!cap1) + continue; + idx = get_connection_index(codec, cap1, pin); + if (idx < 0 && cap2) + idx = get_connection_index(codec, cap2, pin); + if (idx >= 0) { imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = - alc880_input_pin_idx(cfg->input_pins[i]); + imux->items[imux->num_items].index = idx; imux->num_items++; } } return 0; } +static int alc880_auto_create_input_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x08, 0x09); +} + static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid, unsigned int pin_type) { @@ -4448,7 +4606,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc880_is_input_pin(nid)) { + if (alc_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC880_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) @@ -4491,7 +4649,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc880_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -4505,12 +4663,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec) &dig_nid, 1); if (err < 0) continue; - if (dig_nid > 0x7f) { - printk(KERN_ERR "alc880_auto: invalid dig_nid " - "connection 0x%x for NID 0x%x\n", dig_nid, - spec->autocfg.dig_out_pins[i]); - continue; - } if (!i) spec->multiout.dig_out_nid = dig_nid; else { @@ -4547,8 +4699,42 @@ static void alc880_auto_init(struct hda_codec *codec) alc_inithook(codec); } -static void set_capture_mixer(struct alc_spec *spec) +/* check the ADC/MUX contains all input pins; some ADC/MUX contains only + * one of two digital mic pins, e.g. on ALC272 + */ +static void fixup_automic_adc(struct hda_codec *codec) { + struct alc_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t cap = spec->capsrc_nids ? + spec->capsrc_nids[i] : spec->adc_nids[i]; + int iidx, eidx; + + iidx = get_connection_index(codec, cap, spec->int_mic.pin); + if (iidx < 0) + continue; + eidx = get_connection_index(codec, cap, spec->ext_mic.pin); + if (eidx < 0) + continue; + spec->int_mic.mux_idx = iidx; + spec->ext_mic.mux_idx = eidx; + if (spec->capsrc_nids) + spec->capsrc_nids += i; + spec->adc_nids += i; + spec->num_adc_nids = 1; + return; + } + snd_printd(KERN_INFO "hda_codec: %s: " + "No ADC/MUX containing both 0x%x and 0x%x pins\n", + codec->chip_name, spec->int_mic.pin, spec->ext_mic.pin); + spec->auto_mic = 0; /* disable auto-mic to be sure */ +} + +static void set_capture_mixer(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; static struct snd_kcontrol_new *caps[2][3] = { { alc_capture_mixer_nosrc1, alc_capture_mixer_nosrc2, @@ -4559,7 +4745,10 @@ static void set_capture_mixer(struct alc_spec *spec) }; if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { int mux; - if (spec->input_mux && spec->input_mux->num_items > 1) + if (spec->auto_mic) { + mux = 0; + fixup_automic_adc(codec); + } else if (spec->input_mux && spec->input_mux->num_items > 1) mux = 1; else mux = 0; @@ -4590,8 +4779,8 @@ static int patch_alc880(struct hda_codec *codec) alc880_models, alc880_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC880_AUTO; } @@ -4616,7 +4805,7 @@ static int patch_alc880(struct hda_codec *codec) } if (board_config != ALC880_AUTO) - setup_preset(spec, &alc880_presets[board_config]); + setup_preset(codec, &alc880_presets[board_config]); spec->stream_analog_playback = &alc880_pcm_analog_playback; spec->stream_analog_capture = &alc880_pcm_analog_capture; @@ -4629,7 +4818,7 @@ static int patch_alc880(struct hda_codec *codec) /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]); /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + wcap = get_wcaps_type(wcap); if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc880_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt); @@ -4638,7 +4827,7 @@ static int patch_alc880(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids); } } - set_capture_mixer(spec); + set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -5830,7 +6019,14 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[0]; if (nid) { - err = alc260_add_playback_controls(spec, nid, "Front", &vols); + const char *pfx; + if (!cfg->speaker_pins[0] && !cfg->hp_pins[0]) + pfx = "Master"; + else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + else + pfx = "Front"; + err = alc260_add_playback_controls(spec, nid, pfx, &vols); if (err < 0) return err; } @@ -5853,39 +6049,10 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, } /* create playback/capture controls for input pins */ -static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, +static int alc260_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (cfg->input_pins[i] >= 0x12) { - idx = cfg->input_pins[i] - 0x12; - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], idx, - 0x07); - if (err < 0) - return err; - imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } - if (cfg->input_pins[i] >= 0x0f && cfg->input_pins[i] <= 0x10){ - idx = cfg->input_pins[i] - 0x09; - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], idx, - 0x07); - if (err < 0) - return err; - imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } - } - return 0; + return alc_auto_create_input_ctls(codec, cfg, 0x07, 0x04, 0x05); } static void alc260_auto_set_output_and_unmute(struct hda_codec *codec, @@ -5999,7 +6166,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) return err; if (!spec->kctls.list) return 0; /* can't find valid BIOS pin config */ - err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc260_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -6234,8 +6401,7 @@ static int patch_alc260(struct hda_codec *codec) alc260_models, alc260_cfg_tbl); if (board_config < 0) { - snd_printd(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", + snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", codec->chip_name); board_config = ALC260_AUTO; } @@ -6261,7 +6427,7 @@ static int patch_alc260(struct hda_codec *codec) } if (board_config != ALC260_AUTO) - setup_preset(spec, &alc260_presets[board_config]); + setup_preset(codec, &alc260_presets[board_config]); spec->stream_analog_playback = &alc260_pcm_analog_playback; spec->stream_analog_capture = &alc260_pcm_analog_capture; @@ -6272,7 +6438,7 @@ static int patch_alc260(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x04 is valid */ unsigned int wcap = get_wcaps(codec, 0x04); - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + wcap = get_wcaps_type(wcap); /* get type */ if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc260_adc_nids_alt; @@ -6282,7 +6448,7 @@ static int patch_alc260(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids); } } - set_capture_mixer(spec); + set_capture_mixer(codec); set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); spec->vmaster_nid = 0x08; @@ -6301,7 +6467,7 @@ static int patch_alc260(struct hda_codec *codec) /* - * ALC882 support + * ALC882/883/885/888/889 support * * ALC882 is almost identical with ALC880 but has cleaner and more flexible * configuration. Each pin widget can choose any input DACs and a mixer. @@ -6313,22 +6479,35 @@ static int patch_alc260(struct hda_codec *codec) */ #define ALC882_DIGOUT_NID 0x06 #define ALC882_DIGIN_NID 0x0a +#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID +#define ALC883_DIGIN_NID ALC882_DIGIN_NID +#define ALC1200_DIGOUT_NID 0x10 + static struct hda_channel_mode alc882_ch_modes[1] = { { 8, NULL } }; +/* DACs */ static hda_nid_t alc882_dac_nids[4] = { /* front, rear, clfe, rear_surr */ 0x02, 0x03, 0x04, 0x05 }; +#define alc883_dac_nids alc882_dac_nids -/* identical with ALC880 */ +/* ADCs */ #define alc882_adc_nids alc880_adc_nids #define alc882_adc_nids_alt alc880_adc_nids_alt +#define alc883_adc_nids alc882_adc_nids_alt +static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 }; +static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 }; static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; +#define alc883_capsrc_nids alc882_capsrc_nids_alt +static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -6343,6 +6522,17 @@ static struct hda_input_mux alc882_capture_source = { }, }; +#define alc883_capture_source alc882_capture_source + +static struct hda_input_mux alc889_capture_source = { + .num_items = 3, + .items = { + { "Front Mic", 0x0 }, + { "Mic", 0x3 }, + { "Line", 0x2 }, + }, +}; + static struct hda_input_mux mb5_capture_source = { .num_items = 3, .items = { @@ -6352,6 +6542,77 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux alc883_3stack_6ch_intel = { + .num_items = 4, + .items = { + { "Mic", 0x1 }, + { "Front Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_lenovo_101e_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x1 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "iMic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Int Mic", 0x1 }, + }, +}; + +static struct hda_input_mux alc883_lenovo_sky_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_asus_eee1601_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + }, +}; + +static struct hda_input_mux alc889A_mb31_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + /* Front Mic (0x01) unused */ + { "Line", 0x2 }, + /* Line 2 (0x03) unused */ + /* CD (0x04) unsused? */ + }, +}; + +/* + * 2ch mode + */ +static struct hda_channel_mode alc883_3ST_2ch_modes[1] = { + { 2, NULL } +}; + /* * 2ch mode */ @@ -6364,6 +6625,18 @@ static struct hda_verb alc882_3ST_ch2_init[] = { }; /* + * 4ch mode + */ +static struct hda_verb alc882_3ST_ch4_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* * 6ch mode */ static struct hda_verb alc882_3ST_ch6_init[] = { @@ -6376,11 +6649,60 @@ static struct hda_verb alc882_3ST_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc882_3ST_6ch_modes[2] = { +static struct hda_channel_mode alc882_3ST_6ch_modes[3] = { { 2, alc882_3ST_ch2_init }, + { 4, alc882_3ST_ch4_init }, { 6, alc882_3ST_ch6_init }, }; +#define alc883_3ST_6ch_modes alc882_3ST_6ch_modes + +/* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_clevo_init[] = { + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_clevo_init[] = { + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_clevo_init[] = { + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = { + { 2, alc883_3ST_ch2_clevo_init }, + { 4, alc883_3ST_ch4_clevo_init }, + { 6, alc883_3ST_ch6_clevo_init }, +}; + + /* * 6ch mode */ @@ -6468,6 +6790,189 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; + +/* + * 2ch mode + */ +static struct hda_verb alc883_4ST_ch2_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_4ST_ch4_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_4ST_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc883_4ST_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { + { 2, alc883_4ST_ch2_init }, + { 4, alc883_4ST_ch4_init }, + { 6, alc883_4ST_ch6_init }, + { 8, alc883_4ST_ch8_init }, +}; + + +/* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_intel_init[] = { + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { + { 2, alc883_3ST_ch2_intel_init }, + { 4, alc883_3ST_ch4_intel_init }, + { 6, alc883_3ST_ch6_intel_init }, +}; + +/* + * 2ch mode + */ +static struct hda_verb alc889_ch2_intel_init[] = { + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc889_ch6_intel_init[] = { + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc889_ch8_intel_init[] = { + { 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } /* end */ +}; + +static struct hda_channel_mode alc889_8ch_intel_modes[3] = { + { 2, alc889_ch2_intel_init }, + { 6, alc889_ch6_intel_init }, + { 8, alc889_ch8_intel_init }, +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_sixstack_ch6_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { } /* end */ +}; + +/* + * 8ch mode + */ +static struct hda_verb alc883_sixstack_ch8_init[] = { + { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_sixstack_modes[2] = { + { 6, alc883_sixstack_ch6_init }, + { 8, alc883_sixstack_ch8_init }, +}; + + /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b */ @@ -6604,7 +7109,7 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = { { } /* end */ }; -static struct hda_verb alc882_init_verbs[] = { +static struct hda_verb alc882_base_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -6622,6 +7127,13 @@ static struct hda_verb alc882_init_verbs[] = { {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* mute analog input loopbacks */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* Front Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -6656,11 +7168,6 @@ static struct hda_verb alc882_init_verbs[] = { /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, @@ -6671,9 +7178,6 @@ static struct hda_verb alc882_init_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* ADC1: mute amp left and right */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, /* ADC2: mute amp left and right */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -6684,6 +7188,18 @@ static struct hda_verb alc882_init_verbs[] = { { } }; +static struct hda_verb alc882_adc1_init_verbs[] = { + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* ADC1: mute amp left and right */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } +}; + static struct hda_verb alc882_eapd_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -6691,6 +7207,110 @@ static struct hda_verb alc882_eapd_verbs[] = { { } }; +static struct hda_verb alc889_eapd_verbs[] = { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + +static struct hda_verb alc_hp15_unsol_verbs[] = { + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + +static struct hda_verb alc885_init_verbs[] = { + /* Front mixer: unmute input/output amp left and right (volume = 0) */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Rear mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* CLFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Side mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + + /* mute analog input loopbacks */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Front HP Pin: output 0 (0x0c) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Front Pin: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Rear Pin: output 1 (0x0d) */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* CLFE Pin: output 2 (0x0e) */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* Side Pin: output 3 (0x0f) */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* Mic (rear) pin: input vref at 80% */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Front Mic pin: input vref at 80% */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Line In pin: input */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* Mixer elements: 0x18, , 0x1a, 0x1b */ + /* Input mixer1 */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + /* Input mixer3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + /* ADC2: mute amp left and right */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* ADC3: mute amp left and right */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + + { } +}; + +static struct hda_verb alc885_init_input_verbs[] = { + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + { } +}; + + +/* Unmute Selector 24h and set the default input to front mic */ +static struct hda_verb alc889_init_input_verbs[] = { + {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + { } +}; + + +#define alc883_init_verbs alc882_base_init_verbs + /* Mac Pro test */ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -6898,23 +7518,21 @@ static struct hda_verb alc885_imac24_init_verbs[] = { }; /* Toggle speaker-output according to the hp-jack state */ -static void alc885_imac24_automute_init_hook(struct hda_codec *codec) +static void alc885_imac24_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - alc_automute_amp(codec); } -static void alc885_mbp3_init_hook(struct hda_codec *codec) +static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_amp(codec); } @@ -6942,13 +7560,12 @@ static void alc882_targa_automute(struct hda_codec *codec) spec->jack_present ? 1 : 3); } -static void alc882_targa_init_hook(struct hda_codec *codec) +static void alc882_targa_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - alc882_targa_automute(codec); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) @@ -7036,18 +7653,16 @@ static void alc885_macpro_init_hook(struct hda_codec *codec) static void alc885_imac24_init_hook(struct hda_codec *codec) { alc885_macpro_init_hook(codec); - alc885_imac24_automute_init_hook(codec); + alc_automute_amp(codec); } /* * generic initialization of ADC, input mixers and output mixers */ -static struct hda_verb alc882_auto_init_verbs[] = { +static struct hda_verb alc883_auto_init_verbs[] = { /* * Unmute ADC0-2 and set the default input to mic-in */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -7088,11 +7703,6 @@ static struct hda_verb alc882_auto_init_verbs[] = { /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))}, /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))}, @@ -7107,821 +7717,6 @@ static struct hda_verb alc882_auto_init_verbs[] = { { } }; -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc882_loopbacks alc880_loopbacks -#endif - -/* pcm configuration: identical with ALC880 */ -#define alc882_pcm_analog_playback alc880_pcm_analog_playback -#define alc882_pcm_analog_capture alc880_pcm_analog_capture -#define alc882_pcm_digital_playback alc880_pcm_digital_playback -#define alc882_pcm_digital_capture alc880_pcm_digital_capture - -/* - * configuration and preset - */ -static const char *alc882_models[ALC882_MODEL_LAST] = { - [ALC882_3ST_DIG] = "3stack-dig", - [ALC882_6ST_DIG] = "6stack-dig", - [ALC882_ARIMA] = "arima", - [ALC882_W2JC] = "w2jc", - [ALC882_TARGA] = "targa", - [ALC882_ASUS_A7J] = "asus-a7j", - [ALC882_ASUS_A7M] = "asus-a7m", - [ALC885_MACPRO] = "macpro", - [ALC885_MB5] = "mb5", - [ALC885_MBP3] = "mbp3", - [ALC885_IMAC24] = "imac24", - [ALC882_AUTO] = "auto", -}; - -static struct snd_pci_quirk alc882_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), - SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J), - SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M), - SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), - SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ - SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), - {} -}; - -static struct alc_config_preset alc882_presets[] = { - [ALC882_3ST_DIG] = { - .mixers = { alc882_base_mixer }, - .init_verbs = { alc882_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, - [ALC882_6ST_DIG] = { - .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), - .channel_mode = alc882_sixstack_modes, - .input_mux = &alc882_capture_source, - }, - [ALC882_ARIMA] = { - .mixers = { alc882_base_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), - .channel_mode = alc882_sixstack_modes, - .input_mux = &alc882_capture_source, - }, - [ALC882_W2JC] = { - .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - }, - [ALC885_MBP3] = { - .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mbp3_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = 2, - .dac_nids = alc882_dac_nids, - .hp_nid = 0x04, - .channel_mode = alc885_mbp_4ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc885_mbp3_init_hook, - }, - [ALC885_MB5] = { - .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mb5_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mb5_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), - .input_mux = &mb5_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - }, - [ALC885_MACPRO] = { - .mixers = { alc882_macpro_mixer }, - .init_verbs = { alc882_macpro_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .input_mux = &alc882_capture_source, - .init_hook = alc885_macpro_init_hook, - }, - [ALC885_IMAC24] = { - .mixers = { alc885_imac24_mixer }, - .init_verbs = { alc885_imac24_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), - .channel_mode = alc882_ch_modes, - .input_mux = &alc882_capture_source, - .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc885_imac24_init_hook, - }, - [ALC882_TARGA] = { - .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs, - alc882_targa_verbs}, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, - .capsrc_nids = alc882_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), - .channel_mode = alc882_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - .unsol_event = alc882_targa_unsol_event, - .init_hook = alc882_targa_init_hook, - }, - [ALC882_ASUS_A7J] = { - .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs}, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), - .adc_nids = alc882_adc_nids, - .capsrc_nids = alc882_capsrc_nids, - .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), - .channel_mode = alc882_3ST_6ch_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, - [ALC882_ASUS_A7M] = { - .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, - alc880_gpio1_init_verbs, - alc882_asus_a7m_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .dig_out_nid = ALC882_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc882_capture_source, - }, -}; - - -/* - * Pin config fixes - */ -enum { - PINFIX_ABIT_AW9D_MAX -}; - -static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { - { 0x15, 0x01080104 }, /* side */ - { 0x16, 0x01011012 }, /* rear */ - { 0x17, 0x01016011 }, /* clfe */ - { } -}; - -static const struct alc_pincfg *alc882_pin_fixes[] = { - [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, -}; - -static struct snd_pci_quirk alc882_pinfix_tbl[] = { - SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), - {} -}; - -/* - * BIOS auto configuration - */ -static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, - hda_nid_t nid, int pin_type, - int dac_idx) -{ - /* set as output */ - struct alc_spec *spec = codec->spec; - int idx; - - alc_set_pin_output(codec, nid, pin_type); - if (spec->multiout.dac_nids[dac_idx] == 0x25) - idx = 4; - else - idx = spec->multiout.dac_nids[dac_idx] - 2; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); - -} - -static void alc882_auto_init_multi_out(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i <= HDA_SIDE; i++) { - hda_nid_t nid = spec->autocfg.line_out_pins[i]; - int pin_type = get_pin_type(spec->autocfg.line_out_type); - if (nid) - alc882_auto_set_output_and_unmute(codec, nid, pin_type, - i); - } -} - -static void alc882_auto_init_hp_out(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t pin; - - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - /* use dac 0 */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); - pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); -} - -#define alc882_is_input_pin(nid) alc880_is_input_pin(nid) -#define ALC882_PIN_CD_NID ALC880_PIN_CD_NID - -static void alc882_auto_init_analog_input(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - hda_nid_t nid = spec->autocfg.input_pins[i]; - if (!nid) - continue; - alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); - if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); - } -} - -static void alc882_auto_init_input_src(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int c; - - for (c = 0; c < spec->num_adc_nids; c++) { - hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; - hda_nid_t nid = spec->capsrc_nids[c]; - unsigned int mux_idx; - const struct hda_input_mux *imux; - int conns, mute, idx, item; - - conns = snd_hda_get_connections(codec, nid, conn_list, - ARRAY_SIZE(conn_list)); - if (conns < 0) - continue; - mux_idx = c >= spec->num_mux_defs ? 0 : c; - imux = &spec->input_mux[mux_idx]; - for (idx = 0; idx < conns; idx++) { - /* if the current connection is the selected one, - * unmute it as default - otherwise mute it - */ - mute = AMP_IN_MUTE(idx); - for (item = 0; item < imux->num_items; item++) { - if (imux->items[item].index == idx) { - if (spec->cur_mux[c] == item) - mute = AMP_IN_UNMUTE(idx); - break; - } - } - /* check if we have a selector or mixer - * we could check for the widget type instead, but - * just check for Amp-In presence (in case of mixer - * without amp-in there is something wrong, this - * function shouldn't be used or capsrc nid is wrong) - */ - if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - mute); - else if (mute != AMP_IN_MUTE(idx)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, - idx); - } - } -} - -/* add mic boosts if needed */ -static int alc_auto_add_mic_boost(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int err; - hda_nid_t nid; - - nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; - if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Mic Boost", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); - if (err < 0) - return err; - } - nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; - if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Front Mic Boost", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); - if (err < 0) - return err; - } - return 0; -} - -/* almost identical with ALC880 parser... */ -static int alc882_parse_auto_config(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int err = alc880_parse_auto_config(codec); - - if (err < 0) - return err; - else if (!err) - return 0; /* no config found */ - - err = alc_auto_add_mic_boost(codec); - if (err < 0) - return err; - - /* hack - override the init verbs */ - spec->init_verbs[0] = alc882_auto_init_verbs; - - return 1; /* config found */ -} - -/* additional initialization for auto-configuration model */ -static void alc882_auto_init(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - alc882_auto_init_multi_out(codec); - alc882_auto_init_hp_out(codec); - alc882_auto_init_analog_input(codec); - alc882_auto_init_input_src(codec); - if (spec->unsol_event) - alc_inithook(codec); -} - -static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */ - -static int patch_alc882(struct hda_codec *codec) -{ - struct alc_spec *spec; - int err, board_config; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; - - codec->spec = spec; - - board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST, - alc882_models, - alc882_cfg_tbl); - - if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { - /* Pick up systems that don't supply PCI SSID */ - switch (codec->subsystem_id) { - case 0x106b0c00: /* Mac Pro */ - board_config = ALC885_MACPRO; - break; - case 0x106b1000: /* iMac 24 */ - case 0x106b2800: /* AppleTV */ - case 0x106b3e00: /* iMac 24 Aluminium */ - board_config = ALC885_IMAC24; - break; - case 0x106b00a0: /* MacBookPro3,1 - Another revision */ - case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ - case 0x106b00a4: /* MacbookPro4,1 */ - case 0x106b2c00: /* Macbook Pro rev3 */ - /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */ - case 0x106b3800: /* MacbookPro4,1 - latter revision */ - board_config = ALC885_MBP3; - break; - case 0x106b3f00: /* Macbook 5,1 */ - case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense - * seems not working, so apparently - * no perfect solution yet - */ - board_config = ALC885_MB5; - break; - default: - /* ALC889A is handled better as ALC888-compatible */ - if (codec->revision_id == 0x100101 || - codec->revision_id == 0x100103) { - alc_free(codec); - return patch_alc883(codec); - } - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", - codec->chip_name); - board_config = ALC882_AUTO; - } - } - - alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); - - if (board_config == ALC882_AUTO) { - /* automatic parse from the BIOS config */ - err = alc882_parse_auto_config(codec); - if (err < 0) { - alc_free(codec); - return err; - } else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC882_3ST_DIG; - } - } - - err = snd_hda_attach_beep_device(codec, 0x1); - if (err < 0) { - alc_free(codec); - return err; - } - - if (board_config != ALC882_AUTO) - setup_preset(spec, &alc882_presets[board_config]); - - spec->stream_analog_playback = &alc882_pcm_analog_playback; - spec->stream_analog_capture = &alc882_pcm_analog_capture; - /* FIXME: setup DAC5 */ - /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ - spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; - - spec->stream_digital_playback = &alc882_pcm_digital_playback; - spec->stream_digital_capture = &alc882_pcm_digital_capture; - - if (!spec->adc_nids && spec->input_mux) { - /* check whether NID 0x07 is valid */ - unsigned int wcap = get_wcaps(codec, 0x07); - /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN) { - spec->adc_nids = alc882_adc_nids_alt; - spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt); - spec->capsrc_nids = alc882_capsrc_nids_alt; - } else { - spec->adc_nids = alc882_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids); - spec->capsrc_nids = alc882_capsrc_nids; - } - } - set_capture_mixer(spec); - set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); - - spec->vmaster_nid = 0x0c; - - codec->patch_ops = alc_patch_ops; - if (board_config == ALC882_AUTO) - spec->init_hook = alc882_auto_init; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc882_loopbacks; -#endif - codec->proc_widget_hook = print_realtek_coef; - - return 0; -} - -/* - * ALC883 support - * - * ALC883 is almost identical with ALC880 but has cleaner and more flexible - * configuration. Each pin widget can choose any input DACs and a mixer. - * Each ADC is connected from a mixer of all inputs. This makes possible - * 6-channel independent captures. - * - * In addition, an independent DAC for the multi-playback (not used in this - * driver yet). - */ -#define ALC883_DIGOUT_NID 0x06 -#define ALC883_DIGIN_NID 0x0a - -#define ALC1200_DIGOUT_NID 0x10 - -static hda_nid_t alc883_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x03, 0x04, 0x05 -}; - -static hda_nid_t alc883_adc_nids[2] = { - /* ADC1-2 */ - 0x08, 0x09, -}; - -static hda_nid_t alc883_adc_nids_alt[1] = { - /* ADC1 */ - 0x08, -}; - -static hda_nid_t alc883_adc_nids_rev[2] = { - /* ADC2-1 */ - 0x09, 0x08 -}; - -#define alc889_adc_nids alc880_adc_nids - -static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; - -static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; - -#define alc889_capsrc_nids alc882_capsrc_nids - -/* input MUX */ -/* FIXME: should be a matrix-type input source selection */ - -static struct hda_input_mux alc883_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_3stack_6ch_intel = { - .num_items = 4, - .items = { - { "Mic", 0x1 }, - { "Front Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_lenovo_101e_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "Line", 0x2 }, - }, -}; - -static struct hda_input_mux alc883_lenovo_nb0763_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "iMic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Int Mic", 0x1 }, - }, -}; - -static struct hda_input_mux alc883_lenovo_sky_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x4 }, - }, -}; - -static struct hda_input_mux alc883_asus_eee1601_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - }, -}; - -static struct hda_input_mux alc889A_mb31_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - /* Front Mic (0x01) unused */ - { "Line", 0x2 }, - /* Line 2 (0x03) unused */ - /* CD (0x04) unsused? */ - }, -}; - -/* - * 2ch mode - */ -static struct hda_channel_mode alc883_3ST_2ch_modes[1] = { - { 2, NULL } -}; - -/* - * 2ch mode - */ -static struct hda_verb alc883_3ST_ch2_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static struct hda_verb alc883_3ST_ch4_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_3ST_ch6_init[] = { - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -static struct hda_channel_mode alc883_3ST_6ch_modes[3] = { - { 2, alc883_3ST_ch2_init }, - { 4, alc883_3ST_ch4_init }, - { 6, alc883_3ST_ch6_init }, -}; - - -/* - * 2ch mode - */ -static struct hda_verb alc883_4ST_ch2_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static struct hda_verb alc883_4ST_ch4_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_4ST_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static struct hda_verb alc883_4ST_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 }, - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -static struct hda_channel_mode alc883_4ST_8ch_modes[4] = { - { 2, alc883_4ST_ch2_init }, - { 4, alc883_4ST_ch4_init }, - { 6, alc883_4ST_ch6_init }, - { 8, alc883_4ST_ch8_init }, -}; - - -/* - * 2ch mode - */ -static struct hda_verb alc883_3ST_ch2_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* - * 4ch mode - */ -static struct hda_verb alc883_3ST_ch4_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_3ST_ch6_intel_init[] = { - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 }, - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { } /* end */ -}; - -static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = { - { 2, alc883_3ST_ch2_intel_init }, - { 4, alc883_3ST_ch4_intel_init }, - { 6, alc883_3ST_ch6_intel_init }, -}; - -/* - * 6ch mode - */ -static struct hda_verb alc883_sixstack_ch6_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; - -/* - * 8ch mode - */ -static struct hda_verb alc883_sixstack_ch8_init[] = { - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { } /* end */ -}; - -static struct hda_channel_mode alc883_sixstack_modes[2] = { - { 6, alc883_sixstack_ch6_init }, - { 8, alc883_sixstack_ch8_init }, -}; - /* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ static struct hda_verb alc889A_mb31_ch2_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ @@ -7972,34 +7767,7 @@ static struct hda_verb alc883_medion_eapd_verbs[] = { { } }; -/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17 - * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b - */ - -static struct snd_kcontrol_new alc883_base_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; +#define alc883_base_mixer alc882_base_mixer static struct snd_kcontrol_new alc883_mitac_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -8110,6 +7878,30 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_8ch_intel_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, + HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x1b, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -8350,93 +8142,14 @@ static struct snd_kcontrol_new alc883_chmode_mixer[] = { { } /* end */ }; -static struct hda_verb alc883_init_verbs[] = { - /* ADC1: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC2: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* mute analog input loopbacks */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Rear Pin: output 1 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* CLFE Pin: output 2 (0x0e) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* Side Pin: output 3 (0x0f) */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - { } -}; - /* toggle speaker-output according to the hp-jack state */ -static void alc883_mitac_init_hook(struct hda_codec *codec) +static void alc883_mitac_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - alc_automute_amp(codec); } /* auto-toggle front mic */ @@ -8468,6 +8181,22 @@ static struct hda_verb alc883_mitac_verbs[] = { { } /* end */ }; +static struct hda_verb alc883_clevo_m540r_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Int speaker */ + /*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/ + + /* enable unsolicited event */ + /* + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + */ + + { } /* end */ +}; + static struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -8591,7 +8320,7 @@ static struct hda_verb alc883_vaiott_verbs[] = { { } /* end */ }; -static void alc888_3st_hp_init_hook(struct hda_codec *codec) +static void alc888_3st_hp_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8599,7 +8328,6 @@ static void alc888_3st_hp_init_hook(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - alc_automute_amp(codec); } static struct hda_verb alc888_3st_hp_verbs[] = { @@ -8696,13 +8424,12 @@ static struct hda_verb alc883_medion_md2_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc883_medion_md2_init_hook(struct hda_codec *codec) +static void alc883_medion_md2_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - alc_automute_amp(codec); } /* toggle speaker-output according to the hp-jack state */ @@ -8719,12 +8446,16 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } -static void alc883_clevo_m720_init_hook(struct hda_codec *codec) +static void alc883_clevo_m720_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; +} + +static void alc883_clevo_m720_init_hook(struct hda_codec *codec) +{ alc_automute_amp(codec); alc883_clevo_m720_mic_automute(codec); } @@ -8743,22 +8474,20 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec, } /* toggle speaker-output according to the hp-jack state */ -static void alc883_2ch_fujitsu_pi2515_init_hook(struct hda_codec *codec) +static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - alc_automute_amp(codec); } -static void alc883_haier_w66_init_hook(struct hda_codec *codec) +static void alc883_haier_w66_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_amp(codec); } static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) @@ -8797,14 +8526,13 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, } /* toggle speaker-output according to the hp-jack state */ -static void alc883_acer_aspire_init_hook(struct hda_codec *codec) +static void alc883_acer_aspire_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - alc_automute_amp(codec); } static struct hda_verb alc883_acer_eapd_verbs[] = { @@ -8825,7 +8553,14 @@ static struct hda_verb alc883_acer_eapd_verbs[] = { { } }; -static void alc888_6st_dell_init_hook(struct hda_codec *codec) +static struct hda_verb alc888_acer_aspire_7730G_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } /* end */ +}; + +static void alc888_6st_dell_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8834,10 +8569,9 @@ static void alc888_6st_dell_init_hook(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - alc_automute_amp(codec); } -static void alc888_lenovo_sky_init_hook(struct hda_codec *codec) +static void alc888_lenovo_sky_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8847,82 +8581,17 @@ static void alc888_lenovo_sky_init_hook(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - alc_automute_amp(codec); } -static void alc883_vaiott_init_hook(struct hda_codec *codec) +static void alc883_vaiott_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - alc_automute_amp(codec); } -/* - * generic initialization of ADC, input mixers and output mixers - */ -static struct hda_verb alc883_auto_init_verbs[] = { - /* - * Unmute ADC0-2 and set the default input to mic-in - */ - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - /* Input mixer2 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, - - { } -}; - static struct hda_verb alc888_asus_m90v_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -8933,19 +8602,7 @@ static struct hda_verb alc888_asus_m90v_verbs[] = { { } /* end */ }; -static void alc883_nb_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); -} - -static void alc883_M90V_init_hook(struct hda_codec *codec) +static void alc883_mode2_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8953,26 +8610,11 @@ static void alc883_M90V_init_hook(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; - alc_automute_pin(codec); -} - -static void alc883_mode2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_MIC_EVENT: - alc883_nb_mic_automute(codec); - break; - default: - alc_sku_unsol_event(codec, res); - break; - } -} - -static void alc883_mode2_inithook(struct hda_codec *codec) -{ - alc883_M90V_init_hook(codec); - alc883_nb_mic_automute(codec); + spec->ext_mic.pin = 0x18; + spec->int_mic.pin = 0x19; + spec->ext_mic.mux_idx = 0; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } static struct hda_verb alc888_asus_eee1601_verbs[] = { @@ -9033,25 +8675,44 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) alc889A_mb31_automute(codec); } + #ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc883_loopbacks alc880_loopbacks +#define alc882_loopbacks alc880_loopbacks #endif /* pcm configuration: identical with ALC880 */ -#define alc883_pcm_analog_playback alc880_pcm_analog_playback -#define alc883_pcm_analog_capture alc880_pcm_analog_capture -#define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture -#define alc883_pcm_digital_playback alc880_pcm_digital_playback -#define alc883_pcm_digital_capture alc880_pcm_digital_capture +#define alc882_pcm_analog_playback alc880_pcm_analog_playback +#define alc882_pcm_analog_capture alc880_pcm_analog_capture +#define alc882_pcm_digital_playback alc880_pcm_digital_playback +#define alc882_pcm_digital_capture alc880_pcm_digital_capture + +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + +static hda_nid_t alc1200_slave_dig_outs[] = { + ALC883_DIGOUT_NID, 0, +}; /* * configuration and preset */ -static const char *alc883_models[ALC883_MODEL_LAST] = { - [ALC883_3ST_2ch_DIG] = "3stack-dig", +static const char *alc882_models[ALC882_MODEL_LAST] = { + [ALC882_3ST_DIG] = "3stack-dig", + [ALC882_6ST_DIG] = "6stack-dig", + [ALC882_ARIMA] = "arima", + [ALC882_W2JC] = "w2jc", + [ALC882_TARGA] = "targa", + [ALC882_ASUS_A7J] = "asus-a7j", + [ALC882_ASUS_A7M] = "asus-a7m", + [ALC885_MACPRO] = "macpro", + [ALC885_MB5] = "mb5", + [ALC885_MBP3] = "mbp3", + [ALC885_IMAC24] = "imac24", + [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig", [ALC883_3ST_6ch_DIG] = "3stack-6ch-dig", [ALC883_3ST_6ch] = "3stack-6ch", - [ALC883_6ST_DIG] = "6stack-dig", + [ALC883_6ST_DIG] = "alc883-6stack-dig", [ALC883_TARGA_DIG] = "targa-dig", [ALC883_TARGA_2ch_DIG] = "targa-2ch-dig", [ALC883_TARGA_8ch_DIG] = "targa-8ch-dig", @@ -9060,6 +8721,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g", [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g", [ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g", + [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", [ALC883_LAPTOP_EAPD] = "laptop-eapd", @@ -9071,18 +8733,22 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", + [ALC883_CLEVO_M540R] = "clevo-m540r", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", + [ALC889A_INTEL] = "intel-alc889a", + [ALC889_INTEL] = "intel-x58", [ALC1200_ASUS_P5Q] = "asus-p5q", [ALC889A_MB31] = "mb31", [ALC883_SONY_VAIO_TT] = "sony-vaio-tt", - [ALC883_AUTO] = "auto", + [ALC882_AUTO] = "auto", }; -static struct snd_pci_quirk alc883_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), +static struct snd_pci_quirk alc882_cfg_tbl[] = { + SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE), @@ -9097,40 +8763,56 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G", ALC888_ACER_ASPIRE_8930G), - SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), - SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), + SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO), + SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_6530G), + SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G", + ALC888_ACER_ASPIRE_7730G), /* default Acer -- disabled as it causes more problems. * model=auto should work fine now */ /* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), */ + SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), + + SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), + SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), + SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), + + SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), - SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), - SND_PCI_QUIRK(0x1458, 0xa002, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG), @@ -9139,6 +8821,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), @@ -9152,11 +8835,15 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), + SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), + /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */ SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx", ALC883_FUJITSU_PI2515), @@ -9171,24 +8858,186 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), + SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), - SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL), + SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL), + SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), - SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT), - {} -}; -static hda_nid_t alc883_slave_dig_outs[] = { - ALC1200_DIGOUT_NID, 0, + {} }; -static hda_nid_t alc1200_slave_dig_outs[] = { - ALC883_DIGOUT_NID, 0, +/* codec SSID table for Intel Mac */ +static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO), + SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), + SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), + SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), + SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), + /* FIXME: HP jack sense seems not working for MBP 5,1, so apparently + * no perfect solution yet + */ + SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), + {} /* terminator */ }; -static struct alc_config_preset alc883_presets[] = { +static struct alc_config_preset alc882_presets[] = { + [ALC882_3ST_DIG] = { + .mixers = { alc882_base_mixer }, + .init_verbs = { alc882_base_init_verbs, + alc882_adc1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + }, + [ALC882_6ST_DIG] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, + alc882_adc1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, + [ALC882_ARIMA] = { + .mixers = { alc882_base_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes), + .channel_mode = alc882_sixstack_modes, + .input_mux = &alc882_capture_source, + }, + [ALC882_W2JC] = { + .mixers = { alc882_w2jc_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_eapd_verbs, alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + }, + [ALC885_MBP3] = { + .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_mbp3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .hp_nid = 0x04, + .channel_mode = alc885_mbp_4ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), + .input_mux = &alc882_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mbp3_setup, + .init_hook = alc_automute_amp, + }, + [ALC885_MB5] = { + .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_mb5_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mb5_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), + .input_mux = &mb5_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + }, + [ALC885_MACPRO] = { + .mixers = { alc882_macpro_mixer }, + .init_verbs = { alc882_macpro_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .init_hook = alc885_macpro_init_hook, + }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_imac24_setup, + .init_hook = alc885_imac24_init_hook, + }, + [ALC882_TARGA] = { + .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc880_gpio3_init_verbs, alc882_targa_verbs}, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, + .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), + .channel_mode = alc882_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + .unsol_event = alc882_targa_unsol_event, + .setup = alc882_targa_setup, + .init_hook = alc882_targa_automute, + }, + [ALC882_ASUS_A7J] = { + .mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_asus_a7j_verbs}, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, + .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), + .channel_mode = alc882_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + }, + [ALC882_ASUS_A7M] = { + .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_eapd_verbs, alc880_gpio1_init_verbs, + alc882_asus_a7m_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), + .channel_mode = alc880_threestack_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + }, [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, .init_verbs = { alc883_init_verbs }, @@ -9235,6 +9084,46 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_3stack_6ch_intel, }, + [ALC889A_INTEL] = { + .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, + .init_verbs = { alc885_init_verbs, alc885_init_input_verbs, + alc_hp15_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, + .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes), + .channel_mode = alc889_8ch_intel_modes, + .capsrc_nids = alc889_capsrc_nids, + .input_mux = &alc889_capture_source, + .setup = alc889_automute_setup, + .init_hook = alc_automute_amp, + .unsol_event = alc_automute_amp_unsol_event, + .need_dac_fix = 1, + }, + [ALC889_INTEL] = { + .mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer }, + .init_verbs = { alc885_init_verbs, alc889_init_input_verbs, + alc889_eapd_verbs, alc_hp15_unsol_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc889_adc_nids), + .adc_nids = alc889_adc_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, + .num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes), + .channel_mode = alc889_8ch_intel_modes, + .capsrc_nids = alc889_capsrc_nids, + .input_mux = &alc889_capture_source, + .setup = alc889_automute_setup, + .init_hook = alc889_intel_init_hook, + .unsol_event = alc_automute_amp_unsol_event, + .need_dac_fix = 1, + }, [ALC883_6ST_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, .init_verbs = { alc883_init_verbs }, @@ -9258,7 +9147,8 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc883_targa_unsol_event, - .init_hook = alc883_targa_init_hook, + .setup = alc882_targa_setup, + .init_hook = alc882_targa_automute, }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_targa_2ch_mixer}, @@ -9273,7 +9163,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_targa_unsol_event, - .init_hook = alc883_targa_init_hook, + .setup = alc882_targa_setup, + .init_hook = alc882_targa_automute, }, [ALC883_TARGA_8ch_DIG] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -9291,7 +9182,8 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc883_targa_unsol_event, - .init_hook = alc883_targa_init_hook, + .setup = alc882_targa_setup, + .init_hook = alc882_targa_automute, }, [ALC883_ACER] = { .mixers = { alc883_base_mixer }, @@ -9317,7 +9209,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_acer_aspire_init_hook, + .setup = alc883_acer_aspire_setup, + .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_4930G] = { .mixers = { alc888_base_mixer, @@ -9337,7 +9230,8 @@ static struct alc_config_preset alc883_presets[] = { ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_4930g_init_hook, + .setup = alc888_acer_aspire_4930g_setup, + .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_6530G] = { .mixers = { alc888_acer_aspire_6530_mixer }, @@ -9355,7 +9249,8 @@ static struct alc_config_preset alc883_presets[] = { ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_acer_aspire_6530_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_6530g_init_hook, + .setup = alc888_acer_aspire_6530g_setup, + .init_hook = alc_automute_amp, }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, @@ -9376,7 +9271,28 @@ static struct alc_config_preset alc883_presets[] = { ARRAY_SIZE(alc889_capture_sources), .input_mux = alc889_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc889_acer_aspire_8930g_init_hook, + .setup = alc889_acer_aspire_8930g_setup, + .init_hook = alc_automute_amp, + }, + [ALC888_ACER_ASPIRE_7730G] = { + .mixers = { alc883_3ST_6ch_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs, + alc888_acer_aspire_7730G_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev), + .adc_nids = alc883_adc_nids_rev, + .capsrc_nids = alc883_capsrc_nids_rev, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .const_channel_count = 6, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc888_acer_aspire_6530g_setup, + .init_hook = alc_automute_amp, }, [ALC883_MEDION] = { .mixers = { alc883_fivestack_mixer, @@ -9401,7 +9317,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_medion_md2_init_hook, + .setup = alc883_medion_md2_setup, + .init_hook = alc_automute_amp, }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, @@ -9412,6 +9329,21 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_CLEVO_M540R] = { + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes), + .channel_mode = alc883_3ST_6ch_clevo_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + /* This machine has the hardware HP auto-muting, thus + * we need no software mute via unsol event + */ + }, [ALC883_CLEVO_M720] = { .mixers = { alc883_clevo_m720_mixer }, .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, @@ -9422,6 +9354,7 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc883_clevo_m720_unsol_event, + .setup = alc883_clevo_m720_setup, .init_hook = alc883_clevo_m720_init_hook, }, [ALC883_LENOVO_101E_2ch] = { @@ -9447,7 +9380,8 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_lenovo_nb0763_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_medion_md2_init_hook, + .setup = alc883_medion_md2_setup, + .init_hook = alc_automute_amp, }, [ALC888_LENOVO_MS7195_DIG] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9472,7 +9406,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_haier_w66_init_hook, + .setup = alc883_haier_w66_setup, + .init_hook = alc_automute_amp, }, [ALC888_3ST_HP] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9484,7 +9419,8 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_3st_hp_init_hook, + .setup = alc888_3st_hp_setup, + .init_hook = alc_automute_amp, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -9497,7 +9433,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_6st_dell_init_hook, + .setup = alc888_6st_dell_setup, + .init_hook = alc_automute_amp, }, [ALC883_MITAC] = { .mixers = { alc883_mitac_mixer }, @@ -9508,7 +9445,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_mitac_init_hook, + .setup = alc883_mitac_setup, + .init_hook = alc_automute_amp, }, [ALC883_FUJITSU_PI2515] = { .mixers = { alc883_2ch_fujitsu_pi2515_mixer }, @@ -9521,7 +9459,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_fujitsu_pi2515_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_2ch_fujitsu_pi2515_init_hook, + .setup = alc883_2ch_fujitsu_pi2515_setup, + .init_hook = alc_automute_amp, }, [ALC888_FUJITSU_XA3530] = { .mixers = { alc888_base_mixer, alc883_chmode_mixer }, @@ -9539,7 +9478,8 @@ static struct alc_config_preset alc883_presets[] = { ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_2_capture_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_fujitsu_xa3530_init_hook, + .setup = alc888_fujitsu_xa3530_setup, + .init_hook = alc_automute_amp, }, [ALC888_LENOVO_SKY] = { .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, @@ -9552,7 +9492,8 @@ static struct alc_config_preset alc883_presets[] = { .need_dac_fix = 1, .input_mux = &alc883_lenovo_sky_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_lenovo_sky_init_hook, + .setup = alc888_lenovo_sky_setup, + .init_hook = alc_automute_amp, }, [ALC888_ASUS_M90V] = { .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, @@ -9565,8 +9506,9 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc883_fujitsu_pi2515_capture_source, - .unsol_event = alc883_mode2_unsol_event, - .init_hook = alc883_mode2_inithook, + .unsol_event = alc_sku_unsol_event, + .setup = alc883_mode2_setup, + .init_hook = alc_inithook, }, [ALC888_ASUS_EEE1601] = { .mixers = { alc883_asus_eee1601_mixer }, @@ -9619,15 +9561,45 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc883_vaiott_init_hook, + .setup = alc883_vaiott_setup, + .init_hook = alc_automute_amp, }, }; /* + * Pin config fixes + */ +enum { + PINFIX_ABIT_AW9D_MAX +}; + +static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { + { 0x15, 0x01080104 }, /* side */ + { 0x16, 0x01011012 }, /* rear */ + { 0x17, 0x01016011 }, /* clfe */ + { } +}; + +static const struct alc_pincfg *alc882_pin_fixes[] = { + [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +}; + +static struct snd_pci_quirk alc882_pinfix_tbl[] = { + SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), + {} +}; + +/* * BIOS auto configuration */ -static void alc883_auto_set_output_and_unmute(struct hda_codec *codec, +static int alc882_auto_create_input_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x23, 0x22); +} + +static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { @@ -9644,7 +9616,7 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec, } -static void alc883_auto_init_multi_out(struct hda_codec *codec) +static void alc882_auto_init_multi_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int i; @@ -9653,12 +9625,12 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec) hda_nid_t nid = spec->autocfg.line_out_pins[i]; int pin_type = get_pin_type(spec->autocfg.line_out_type); if (nid) - alc883_auto_set_output_and_unmute(codec, nid, pin_type, + alc882_auto_set_output_and_unmute(codec, nid, pin_type, i); } } -static void alc883_auto_init_hp_out(struct hda_codec *codec) +static void alc882_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; hda_nid_t pin; @@ -9666,91 +9638,191 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec) pin = spec->autocfg.hp_pins[0]; if (pin) /* connect to front */ /* use dac 0 */ - alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); pin = spec->autocfg.speaker_pins[0]; if (pin) - alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc883_is_input_pin(nid) alc880_is_input_pin(nid) -#define ALC883_PIN_CD_NID ALC880_PIN_CD_NID - -static void alc883_auto_init_analog_input(struct hda_codec *codec) +static void alc882_auto_init_analog_input(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int i; for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc883_is_input_pin(nid)) { - alc_set_input_pin(codec, nid, i); - if (nid != ALC883_PIN_CD_NID && - (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) + if (!nid) + continue; + alc_set_input_pin(codec, nid, i); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + } +} + +static void alc882_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; + hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; + int conns, mute, idx, item; + + conns = snd_hda_get_connections(codec, nid, conn_list, + ARRAY_SIZE(conn_list)); + if (conns < 0) + continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; + for (idx = 0; idx < conns; idx++) { + /* if the current connection is the selected one, + * unmute it as default - otherwise mute it + */ + mute = AMP_IN_MUTE(idx); + for (item = 0; item < imux->num_items; item++) { + if (imux->items[item].index == idx) { + if (spec->cur_mux[c] == item) + mute = AMP_IN_UNMUTE(idx); + break; + } + } + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_MUTE); + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } -#define alc883_auto_init_input_src alc882_auto_init_input_src +/* add mic boosts if needed */ +static int alc_auto_add_mic_boost(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + hda_nid_t nid; + + nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; + if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Mic Boost", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; + if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Front Mic Boost", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} /* almost identical with ALC880 parser... */ -static int alc883_parse_auto_config(struct hda_codec *codec) +static int alc882_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err = alc880_parse_auto_config(codec); - struct auto_pin_cfg *cfg = &spec->autocfg; - int i; + static hda_nid_t alc882_ignore[] = { 0x1d, 0 }; + int i, err; + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc882_ignore); if (err < 0) return err; - else if (!err) - return 0; /* no config found */ + if (!spec->autocfg.line_outs) + return 0; /* can't find valid BIOS pin config */ - err = alc_auto_add_mic_boost(codec); + err = alc880_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc880_auto_create_extra_out(spec, + spec->autocfg.speaker_pins[0], + "Speaker"); + if (err < 0) + return err; + err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0], + "Headphone"); + if (err < 0) + return err; + err = alc882_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; - /* hack - override the init verbs */ - spec->init_verbs[0] = alc883_auto_init_verbs; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; - /* setup input_mux for ALC889 */ - if (codec->vendor_id == 0x10ec0889) { - /* digital-mic input pin is excluded in alc880_auto_create..() - * because it's under 0x18 - */ - if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || - cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux[0]; - for (i = 1; i < 3; i++) - memcpy(&spec->private_imux[i], - &spec->private_imux[0], - sizeof(spec->private_imux[0])); - imux->items[imux->num_items].label = "Int DMic"; - imux->items[imux->num_items].index = 0x0b; - imux->num_items++; - spec->num_mux_defs = 3; - spec->input_mux = spec->private_imux; + /* check multiple SPDIF-out (for recent codecs) */ + for (i = 0; i < spec->autocfg.dig_outs; i++) { + hda_nid_t dig_nid; + err = snd_hda_get_connections(codec, + spec->autocfg.dig_out_pins[i], + &dig_nid, 1); + if (err < 0) + continue; + if (!i) + spec->multiout.dig_out_nid = dig_nid; + else { + spec->multiout.slave_dig_outs = spec->slave_dig_outs; + spec->slave_dig_outs[i - 1] = dig_nid; + if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + break; } } + if (spec->autocfg.dig_in_pin) + spec->dig_in_nid = ALC880_DIGIN_NID; + + if (spec->kctls.list) + add_mixer(spec, spec->kctls.list); + + add_verb(spec, alc883_auto_init_verbs); + /* if ADC 0x07 is available, initialize it, too */ + if (get_wcaps_type(get_wcaps(codec, 0x07)) == AC_WID_AUD_IN) + add_verb(spec, alc882_adc1_init_verbs); + + spec->num_mux_defs = 1; + spec->input_mux = &spec->private_imux[0]; + + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; return 1; /* config found */ } /* additional initialization for auto-configuration model */ -static void alc883_auto_init(struct hda_codec *codec) +static void alc882_auto_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - alc883_auto_init_multi_out(codec); - alc883_auto_init_hp_out(codec); - alc883_auto_init_analog_input(codec); - alc883_auto_init_input_src(codec); + alc882_auto_init_multi_out(codec); + alc882_auto_init_hp_out(codec); + alc882_auto_init_analog_input(codec); + alc882_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } -static int patch_alc883(struct hda_codec *codec) +static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; int err, board_config; @@ -9761,28 +9833,35 @@ static int patch_alc883(struct hda_codec *codec) codec->spec = spec; - alc_fix_pll_init(codec, 0x20, 0x0a, 10); + switch (codec->vendor_id) { + case 0x10ec0882: + case 0x10ec0885: + break; + default: + /* ALC883 and variants */ + alc_fix_pll_init(codec, 0x20, 0x0a, 10); + break; + } - board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, - alc883_models, - alc883_cfg_tbl); - if (board_config < 0 || board_config >= ALC883_MODEL_LAST) { - /* Pick up systems that don't supply PCI SSID */ - switch (codec->subsystem_id) { - case 0x106b3600: /* Macbook 3.1 */ - board_config = ALC889A_MB31; - break; - default: - printk(KERN_INFO - "hda_codec: Unknown model for %s, trying " - "auto-probe from BIOS...\n", codec->chip_name); - board_config = ALC883_AUTO; - } + board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST, + alc882_models, + alc882_cfg_tbl); + + if (board_config < 0 || board_config >= ALC882_MODEL_LAST) + board_config = snd_hda_check_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl); + + if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); + board_config = ALC882_AUTO; } - if (board_config == ALC883_AUTO) { + alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + + if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ - err = alc883_parse_auto_config(codec); + err = alc882_parse_auto_config(codec); if (err < 0) { alc_free(codec); return err; @@ -9790,7 +9869,7 @@ static int patch_alc883(struct hda_codec *codec) printk(KERN_INFO "hda_codec: Cannot set up configuration " "from BIOS. Using base mode...\n"); - board_config = ALC883_3ST_2ch_DIG; + board_config = ALC882_3ST_DIG; } } @@ -9800,63 +9879,61 @@ static int patch_alc883(struct hda_codec *codec) return err; } - if (board_config != ALC883_AUTO) - setup_preset(spec, &alc883_presets[board_config]); + if (board_config != ALC882_AUTO) + setup_preset(codec, &alc882_presets[board_config]); - switch (codec->vendor_id) { - case 0x10ec0888: - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; + spec->stream_analog_playback = &alc882_pcm_analog_playback; + spec->stream_analog_capture = &alc882_pcm_analog_capture; + /* FIXME: setup DAC5 */ + /*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/ + spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture; + + spec->stream_digital_playback = &alc882_pcm_digital_playback; + spec->stream_digital_capture = &alc882_pcm_digital_capture; + + if (codec->vendor_id == 0x10ec0888) spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ - break; - case 0x10ec0889: - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); - spec->adc_nids = alc889_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc889_capsrc_nids; - break; - default: - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; + + if (!spec->adc_nids && spec->input_mux) { + int i; + spec->num_adc_nids = 0; + for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) { + hda_nid_t cap; + hda_nid_t nid = alc882_adc_nids[i]; + unsigned int wcap = get_wcaps(codec, nid); + /* get type */ + wcap = get_wcaps_type(wcap); + if (wcap != AC_WID_AUD_IN) + continue; + spec->private_adc_nids[spec->num_adc_nids] = nid; + err = snd_hda_get_connections(codec, nid, &cap, 1); + if (err < 0) + continue; + spec->private_capsrc_nids[spec->num_adc_nids] = cap; + spec->num_adc_nids++; } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - break; + spec->adc_nids = spec->private_adc_nids; + spec->capsrc_nids = spec->private_capsrc_nids; } - spec->stream_analog_playback = &alc883_pcm_analog_playback; - spec->stream_analog_capture = &alc883_pcm_analog_capture; - spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture; - - spec->stream_digital_playback = &alc883_pcm_digital_playback; - spec->stream_digital_capture = &alc883_pcm_digital_capture; - - if (!spec->cap_mixer) - set_capture_mixer(spec); + set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; - if (board_config == ALC883_AUTO) - spec->init_hook = alc883_auto_init; - + if (board_config == ALC882_AUTO) + spec->init_hook = alc882_auto_init; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) - spec->loopback.amplist = alc883_loopbacks; + spec->loopback.amplist = alc882_loopbacks; #endif codec->proc_widget_hook = print_realtek_coef; return 0; } + /* * ALC262 support */ @@ -10032,13 +10109,12 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = { }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc262_hp_t5735_init_hook(struct hda_codec *codec) +static void alc262_hp_t5735_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */ - alc_automute_amp(codec); } static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = { @@ -10195,22 +10271,20 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res) alc262_hippo_automute(codec); } -static void alc262_hippo_init_hook(struct hda_codec *codec) +static void alc262_hippo_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc262_hippo_automute(codec); } -static void alc262_hippo1_init_hook(struct hda_codec *codec) +static void alc262_hippo1_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - alc262_hippo_automute(codec); } @@ -10267,13 +10341,12 @@ static struct hda_verb alc262_tyan_verbs[] = { }; /* unsolicited event for HP jack sensing */ -static void alc262_tyan_init_hook(struct hda_codec *codec) +static void alc262_tyan_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - alc_automute_amp(codec); } @@ -10365,12 +10438,6 @@ static struct hda_verb alc262_eapd_verbs[] = { { } }; -static struct hda_verb alc262_hippo_unsol_verbs[] = { - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {} -}; - static struct hda_verb alc262_hippo1_unsol_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -10391,14 +10458,6 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { {} }; -static struct hda_input_mux alc262_dmic_capture_source = { - .num_items = 2, - .items = { - { "Int DMic", 0x9 }, - { "Mic", 0x0 }, - }, -}; - static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -10420,35 +10479,17 @@ static struct hda_verb alc262_toshiba_s06_verbs[] = { {} }; -static void alc262_dmic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x22, 0, - AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09); -} - - -/* unsolicited event for HP jack sensing */ -static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC880_MIC_EVENT) - alc262_dmic_automute(codec); - else - alc_sku_unsol_event(codec, res); -} - -static void alc262_toshiba_s06_init_hook(struct hda_codec *codec) +static void alc262_toshiba_s06_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_pin(codec); - alc262_dmic_automute(codec); + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 9; + spec->auto_mic = 1; } /* @@ -10866,104 +10907,111 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { { } /* end */ }; +/* We use two mixers depending on the output pin; 0x16 is a mono output + * and thus it's bound with a different mixer. + * This function returns which mixer amp should be used. + */ +static int alc262_check_volbit(hda_nid_t nid) +{ + if (!nid) + return 0; + else if (nid == 0x16) + return 2; + else + return 1; +} + +static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, + const char *pfx, int *vbits) +{ + char name[32]; + unsigned long val; + int vbit; + + vbit = alc262_check_volbit(nid); + if (!vbit) + return 0; + if (*vbits & vbit) /* a volume control for this mixer already there */ + return 0; + *vbits |= vbit; + snprintf(name, sizeof(name), "%s Playback Volume", pfx); + if (vbit == 2) + val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT); + else + val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT); + return add_control(spec, ALC_CTL_WIDGET_VOL, name, val); +} + +static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid, + const char *pfx) +{ + char name[32]; + unsigned long val; + + if (!nid) + return 0; + snprintf(name, sizeof(name), "%s Playback Switch", pfx); + if (nid == 0x16) + val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT); + else + val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val); +} + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - hda_nid_t nid; + const char *pfx; + int vbits; int err; spec->multiout.num_dacs = 1; /* only use one dac */ spec->multiout.dac_nids = spec->private_dac_nids; spec->multiout.dac_nids[0] = 2; - nid = cfg->line_out_pins[0]; - if (nid) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Front Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - - nid = cfg->speaker_pins[0]; - if (nid) { - if (nid == 0x16) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Speaker Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Speaker Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } else { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Speaker Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } - } - nid = cfg->hp_pins[0]; - if (nid) { - /* spec->multiout.hp_nid = 2; */ - if (nid == 0x16) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } else { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } - } - return 0; -} - -static int alc262_auto_create_analog_input_ctls(struct alc_spec *spec, - const struct auto_pin_cfg *cfg) -{ - int err; + if (!cfg->speaker_pins[0] && !cfg->hp_pins[0]) + pfx = "Master"; + else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + else + pfx = "Front"; + err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[0], pfx); + if (err < 0) + return err; + err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[0], "Speaker"); + if (err < 0) + return err; + err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[0], "Headphone"); + if (err < 0) + return err; - err = alc880_auto_create_analog_input_ctls(spec, cfg); + vbits = alc262_check_volbit(cfg->line_out_pins[0]) | + alc262_check_volbit(cfg->speaker_pins[0]) | + alc262_check_volbit(cfg->hp_pins[0]); + if (vbits == 1 || vbits == 2) + pfx = "Master"; /* only one mixer is used */ + else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + else + pfx = "Front"; + vbits = 0; + err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[0], pfx, &vbits); + if (err < 0) + return err; + err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[0], "Speaker", + &vbits); + if (err < 0) + return err; + err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[0], "Headphone", + &vbits); if (err < 0) return err; - /* digital-mic input pin is excluded in alc880_auto_create..() - * because it's under 0x18 - */ - if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || - cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux[0]; - imux->items[imux->num_items].label = "Int Mic"; - imux->items[imux->num_items].index = 0x09; - imux->num_items++; - } return 0; } +#define alc262_auto_create_input_ctls \ + alc880_auto_create_input_ctls /* * generic initialization of ADC, input mixers and output mixers @@ -11281,7 +11329,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; - err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc262_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -11412,7 +11460,7 @@ static struct alc_config_preset alc262_presets[] = { }, [ALC262_HIPPO] = { .mixers = { alc262_hippo_mixer }, - .init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs}, + .init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs}, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, .hp_nid = 0x03, @@ -11421,7 +11469,8 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_init_hook, + .setup = alc262_hippo_setup, + .init_hook = alc262_hippo_automute, }, [ALC262_HIPPO_1] = { .mixers = { alc262_hippo1_mixer }, @@ -11434,7 +11483,8 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo1_init_hook, + .setup = alc262_hippo1_setup, + .init_hook = alc262_hippo_automute, }, [ALC262_FUJITSU] = { .mixers = { alc262_fujitsu_mixer }, @@ -11497,7 +11547,8 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc262_hp_t5735_init_hook, + .setup = alc262_hp_t5735_setup, + .init_hook = alc_automute_amp, }, [ALC262_HP_RP5700] = { .mixers = { alc262_hp_rp5700_mixer }, @@ -11528,11 +11579,13 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_init_hook, + .setup = alc262_hippo_setup, + .init_hook = alc262_hippo_automute, }, [ALC262_BENQ_T31] = { .mixers = { alc262_benq_t31_mixer }, - .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs }, + .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, + alc_hp15_unsol_verbs }, .num_dacs = ARRAY_SIZE(alc262_dac_nids), .dac_nids = alc262_dac_nids, .hp_nid = 0x03, @@ -11540,7 +11593,8 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_init_hook, + .setup = alc262_hippo_setup, + .init_hook = alc262_hippo_automute, }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer }, @@ -11592,9 +11646,9 @@ static struct alc_config_preset alc262_presets[] = { .dig_out_nid = ALC262_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, - .input_mux = &alc262_dmic_capture_source, - .unsol_event = alc262_toshiba_s06_unsol_event, - .init_hook = alc262_toshiba_s06_init_hook, + .unsol_event = alc_sku_unsol_event, + .setup = alc262_toshiba_s06_setup, + .init_hook = alc_inithook, }, [ALC262_TOSHIBA_RX1] = { .mixers = { alc262_toshiba_rx1_mixer }, @@ -11606,7 +11660,8 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, - .init_hook = alc262_hippo_init_hook, + .setup = alc262_hippo_setup, + .init_hook = alc262_hippo_automute, }, [ALC262_TYAN] = { .mixers = { alc262_tyan_mixer }, @@ -11619,7 +11674,8 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc262_tyan_init_hook, + .setup = alc262_tyan_setup, + .init_hook = alc_automute_amp, }, }; @@ -11654,8 +11710,8 @@ static int patch_alc262(struct hda_codec *codec) alc262_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC262_AUTO; } @@ -11682,7 +11738,7 @@ static int patch_alc262(struct hda_codec *codec) } if (board_config != ALC262_AUTO) - setup_preset(spec, &alc262_presets[board_config]); + setup_preset(codec, &alc262_presets[board_config]); spec->stream_analog_playback = &alc262_pcm_analog_playback; spec->stream_analog_capture = &alc262_pcm_analog_capture; @@ -11708,7 +11764,7 @@ static int patch_alc262(struct hda_codec *codec) unsigned int wcap = get_wcaps(codec, 0x07); /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + wcap = get_wcaps_type(wcap); if (wcap != AC_WID_AUD_IN) { spec->adc_nids = alc262_adc_nids_alt; spec->num_adc_nids = @@ -11723,7 +11779,7 @@ static int patch_alc262(struct hda_codec *codec) } } if (!spec->cap_mixer && !spec->no_analog) - set_capture_mixer(spec); + set_capture_mixer(codec); if (!spec->no_analog) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); @@ -11815,14 +11871,6 @@ static struct hda_verb alc268_toshiba_verbs[] = { { } /* end */ }; -static struct hda_input_mux alc268_acer_lc_capture_source = { - .num_items = 2, - .items = { - { "i-Mic", 0x6 }, - { "E-Mic", 0x0 }, - }, -}; - /* Acer specific */ /* bind volumes of both NID 0x02 and 0x03 */ static struct hda_bind_ctls alc268_acer_bind_master_vol = { @@ -11941,7 +11989,8 @@ static struct hda_verb alc268_acer_verbs[] = { /* unsolicited event for HP jack sensing */ #define alc268_toshiba_unsol_event alc262_hippo_unsol_event -#define alc268_toshiba_init_hook alc262_hippo_init_hook +#define alc268_toshiba_setup alc262_hippo_setup +#define alc268_toshiba_automute alc262_hippo_automute static void alc268_acer_unsol_event(struct hda_codec *codec, unsigned int res) @@ -11971,30 +12020,33 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) AMP_IN_MUTE(0), bits); } - -static void alc268_acer_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL, - present ? 0x0 : 0x6); -} - static void alc268_acer_lc_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) + switch (res >> 26) { + case ALC880_HP_EVENT: alc268_aspire_one_speaker_automute(codec); - if ((res >> 26) == ALC880_MIC_EVENT) - alc268_acer_mic_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +static void alc268_acer_lc_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; } static void alc268_acer_lc_init_hook(struct hda_codec *codec) { alc268_aspire_one_speaker_automute(codec); - alc268_acer_mic_automute(codec); + alc_mic_automute(codec); } static struct snd_kcontrol_new alc268_dell_mixer[] = { @@ -12012,17 +12064,22 @@ static struct hda_verb alc268_dell_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, { } }; /* mute/unmute internal speaker according to the hp jack and mute state */ -static void alc268_dell_init_hook(struct hda_codec *codec) +static void alc268_dell_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_pin(codec); + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = { @@ -12043,38 +12100,16 @@ static struct hda_verb alc267_quanta_il1_verbs[] = { { } }; -static void alc267_quanta_il1_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x23, 0, - AC_VERB_SET_CONNECT_SEL, - present ? 0x00 : 0x01); -} - -static void alc267_quanta_il1_init_hook(struct hda_codec *codec) +static void alc267_quanta_il1_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_pin(codec); - alc267_quanta_il1_mic_automute(codec); -} - -static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_MIC_EVENT: - alc267_quanta_il1_mic_automute(codec); - break; - default: - alc_sku_unsol_event(codec, res); - break; - } + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } /* @@ -12154,21 +12189,16 @@ static struct hda_verb alc268_volume_init_verbs[] = { { } }; +static struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, + _DEFINE_CAPSRC(1), { } /* end */ }; @@ -12177,18 +12207,7 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = { HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = alc_mux_enum_info, - .get = alc_mux_enum_get, - .put = alc_mux_enum_put, - }, + _DEFINE_CAPSRC(2), { } /* end */ }; @@ -12275,26 +12294,38 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) { char name[32]; + hda_nid_t dac; int err; sprintf(name, "%s Playback Volume", ctlname); - if (nid == 0x14) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x02, 3, idx, - HDA_OUTPUT)); - if (err < 0) - return err; - } else if (nid == 0x15) { + switch (nid) { + case 0x14: + case 0x16: + dac = 0x02; + break; + case 0x15: + dac = 0x03; + break; + default: + return 0; + } + if (spec->multiout.dac_nids[0] != dac && + spec->multiout.dac_nids[1] != dac) { err = add_control(spec, ALC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x03, 3, idx, + HDA_COMPOSE_AMP_VAL(dac, 3, idx, HDA_OUTPUT)); if (err < 0) return err; - } else - return -1; + spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; + } + sprintf(name, "%s Playback Switch", ctlname); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + if (nid != 0x16) + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); + else /* mono */ + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT)); if (err < 0) return err; return 0; @@ -12307,14 +12338,19 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, hda_nid_t nid; int err; - spec->multiout.num_dacs = 2; /* only use one dac */ spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.dac_nids[0] = 2; - spec->multiout.dac_nids[1] = 3; nid = cfg->line_out_pins[0]; - if (nid) - alc268_new_analog_output(spec, nid, "Front", 0); + if (nid) { + const char *name; + if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + name = "Speaker"; + else + name = "Front"; + err = alc268_new_analog_output(spec, nid, name, 0); + if (err < 0) + return err; + } nid = cfg->speaker_pins[0]; if (nid == 0x1d) { @@ -12323,16 +12359,23 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) return err; + } else { + err = alc268_new_analog_output(spec, nid, "Speaker", 0); + if (err < 0) + return err; } nid = cfg->hp_pins[0]; - if (nid) - alc268_new_analog_output(spec, nid, "Headphone", 0); + if (nid) { + err = alc268_new_analog_output(spec, nid, "Headphone", 0); + if (err < 0) + return err; + } nid = cfg->line_out_pins[1] | cfg->line_out_pins[2]; if (nid == 0x16) { err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Mono Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT)); + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT)); if (err < 0) return err; } @@ -12340,38 +12383,46 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, } /* create playback/capture controls for input pins */ -static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, +static int alc268_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, idx1; + return alc_auto_create_input_ctls(codec, cfg, 0, 0x23, 0x24); +} - for (i = 0; i < AUTO_PIN_LAST; i++) { - switch(cfg->input_pins[i]) { - case 0x18: - idx1 = 0; /* Mic 1 */ - break; - case 0x19: - idx1 = 1; /* Mic 2 */ - break; - case 0x1a: - idx1 = 2; /* Line In */ - break; - case 0x1c: - idx1 = 3; /* CD */ - break; - case 0x12: - case 0x13: - idx1 = 6; /* digital mics */ - break; - default: - continue; - } - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx1; - imux->num_items++; +static void alc268_auto_set_output_and_unmute(struct hda_codec *codec, + hda_nid_t nid, int pin_type) +{ + int idx; + + alc_set_pin_output(codec, nid, pin_type); + if (nid == 0x14 || nid == 0x16) + idx = 0; + else + idx = 1; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); +} + +static void alc268_auto_init_multi_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid = spec->autocfg.line_out_pins[0]; + if (nid) { + int pin_type = get_pin_type(spec->autocfg.line_out_type); + alc268_auto_set_output_and_unmute(codec, nid, pin_type); } - return 0; +} + +static void alc268_auto_init_hp_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t pin; + + pin = spec->autocfg.hp_pins[0]; + if (pin) + alc268_auto_set_output_and_unmute(codec, pin, PIN_HP); + pin = spec->autocfg.speaker_pins[0]; + if (pin) + alc268_auto_set_output_and_unmute(codec, pin, PIN_OUT); } static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) @@ -12382,9 +12433,10 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; unsigned int dac_vol1, dac_vol2; - if (speaker_nid) { + if (line_nid == 0x1d || speaker_nid == 0x1d) { snd_hda_codec_write(codec, speaker_nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* mute mixer inputs from 0x1d */ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)); @@ -12392,6 +12444,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)); } else { + /* unmute mixer inputs from 0x1d */ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); snd_hda_codec_write(codec, 0x10, 0, @@ -12448,7 +12501,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; - err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc268_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -12467,7 +12520,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); - spec->num_mux_defs = 1; + spec->num_mux_defs = 2; spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); @@ -12479,8 +12532,6 @@ static int alc268_parse_auto_config(struct hda_codec *codec) return 1; } -#define alc268_auto_init_multi_out alc882_auto_init_multi_out -#define alc268_auto_init_hp_out alc882_auto_init_hp_out #define alc268_auto_init_analog_input alc882_auto_init_analog_input /* init callback for auto-configuration model -- overriding the default init */ @@ -12523,8 +12574,11 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), + /* almost compatible with toshiba but with optional digital outs; + * auto-probing seems working fine + */ SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series", - ALC268_TOSHIBA), + ALC268_AUTO), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), @@ -12545,7 +12599,8 @@ static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer, + alc268_capture_nosrc_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -12555,9 +12610,9 @@ static struct alc_config_preset alc268_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .input_mux = &alc268_capture_source, - .unsol_event = alc267_quanta_il1_unsol_event, - .init_hook = alc267_quanta_il1_init_hook, + .unsol_event = alc_sku_unsol_event, + .setup = alc267_quanta_il1_setup, + .init_hook = alc_inithook, }, [ALC268_3ST] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, @@ -12589,10 +12644,11 @@ static struct alc_config_preset alc268_presets[] = { .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, .unsol_event = alc268_toshiba_unsol_event, - .init_hook = alc268_toshiba_init_hook, + .setup = alc268_toshiba_setup, + .init_hook = alc268_toshiba_automute, }, [ALC268_ACER] = { - .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer, + .mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_verbs }, @@ -12628,7 +12684,7 @@ static struct alc_config_preset alc268_presets[] = { [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, alc268_beep_mixer, - alc268_capture_alt_mixer }, + alc268_capture_nosrc_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -12639,22 +12695,26 @@ static struct alc_config_preset alc268_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, - .input_mux = &alc268_acer_lc_capture_source, .unsol_event = alc268_acer_lc_unsol_event, + .setup = alc268_acer_lc_setup, .init_hook = alc268_acer_lc_init_hook, }, [ALC268_DELL] = { - .mixers = { alc268_dell_mixer, alc268_beep_mixer }, + .mixers = { alc268_dell_mixer, alc268_beep_mixer, + alc268_capture_nosrc_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_dell_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, .unsol_event = alc_sku_unsol_event, - .init_hook = alc268_dell_init_hook, - .input_mux = &alc268_capture_source, + .setup = alc268_dell_setup, + .init_hook = alc_inithook, }, [ALC268_ZEPTO] = { .mixers = { alc268_base_mixer, alc268_capture_alt_mixer, @@ -12671,8 +12731,8 @@ static struct alc_config_preset alc268_presets[] = { .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, .input_mux = &alc268_capture_source, - .unsol_event = alc268_toshiba_unsol_event, - .init_hook = alc268_toshiba_init_hook + .setup = alc268_toshiba_setup, + .init_hook = alc268_toshiba_automute, }, #ifdef CONFIG_SND_DEBUG [ALC268_TEST] = { @@ -12714,8 +12774,8 @@ static int patch_alc268(struct hda_codec *codec) ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC268_AUTO; } @@ -12734,7 +12794,7 @@ static int patch_alc268(struct hda_codec *codec) } if (board_config != ALC268_AUTO) - setup_preset(spec, &alc268_presets[board_config]); + setup_preset(codec, &alc268_presets[board_config]); spec->stream_analog_playback = &alc268_pcm_analog_playback; spec->stream_analog_capture = &alc268_pcm_analog_capture; @@ -12771,11 +12831,15 @@ static int patch_alc268(struct hda_codec *codec) int i; /* get type */ - wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { + wcap = get_wcaps_type(wcap); + if (spec->auto_mic || + wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); - add_mixer(spec, alc268_capture_alt_mixer); + if (spec->auto_mic || spec->input_mux->num_items == 1) + add_mixer(spec, alc268_capture_nosrc_mixer); + else + add_mixer(spec, alc268_capture_alt_mixer); } else { spec->adc_nids = alc268_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); @@ -12786,6 +12850,8 @@ static int patch_alc268(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], 0, AC_VERB_SET_CONNECT_SEL, + i < spec->num_mux_defs ? + spec->input_mux[i].items[0].index : spec->input_mux->items[0].index); } @@ -12820,22 +12886,6 @@ static hda_nid_t alc269_capsrc_nids[1] = { * not a mux! */ -static struct hda_input_mux alc269_eeepc_dmic_capture_source = { - .num_items = 2, - .items = { - { "i-Mic", 0x5 }, - { "e-Mic", 0x0 }, - }, -}; - -static struct hda_input_mux alc269_eeepc_amic_capture_source = { - .num_items = 2, - .items = { - { "i-Mic", 0x1 }, - { "e-Mic", 0x0 }, - }, -}; - #define alc269_modes alc260_modes #define alc269_capture_source alc880_lg_lw_capture_source @@ -12997,16 +13047,6 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) AC_VERB_SET_PROC_COEF, 0x480); } -static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x23, 0, - AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1); -} - static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) { unsigned int present_laptop; @@ -13033,10 +13073,14 @@ static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) + switch (res >> 26) { + case ALC880_HP_EVENT: alc269_quanta_fl1_speaker_automute(codec); - if ((res >> 26) == ALC880_MIC_EVENT) - alc269_quanta_fl1_mic_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } } static void alc269_lifebook_unsol_event(struct hda_codec *codec, @@ -13048,10 +13092,20 @@ static void alc269_lifebook_unsol_event(struct hda_codec *codec, alc269_lifebook_mic_autoswitch(codec); } +static void alc269_quanta_fl1_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; +} + static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) { alc269_quanta_fl1_speaker_automute(codec); - alc269_quanta_fl1_mic_automute(codec); + alc_mic_automute(codec); } static void alc269_lifebook_init_hook(struct hda_codec *codec) @@ -13096,60 +13150,44 @@ static void alc269_speaker_automute(struct hda_codec *codec) AMP_IN_MUTE(0), bits); } -static void alc269_eeepc_dmic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x23, 0, - AC_VERB_SET_CONNECT_SEL, (present ? 0 : 5)); -} - -static void alc269_eeepc_amic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); -} - /* unsolicited event for HP jack sensing */ -static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec, +static void alc269_eeepc_unsol_event(struct hda_codec *codec, unsigned int res) { - if ((res >> 26) == ALC880_HP_EVENT) + switch (res >> 26) { + case ALC880_HP_EVENT: alc269_speaker_automute(codec); - - if ((res >> 26) == ALC880_MIC_EVENT) - alc269_eeepc_dmic_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } } -static void alc269_eeepc_dmic_inithook(struct hda_codec *codec) +static void alc269_eeepc_dmic_setup(struct hda_codec *codec) { - alc269_speaker_automute(codec); - alc269_eeepc_dmic_automute(codec); + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 5; + spec->auto_mic = 1; } -/* unsolicited event for HP jack sensing */ -static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec, - unsigned int res) +static void alc269_eeepc_amic_setup(struct hda_codec *codec) { - if ((res >> 26) == ALC880_HP_EVENT) - alc269_speaker_automute(codec); - - if ((res >> 26) == ALC880_MIC_EVENT) - alc269_eeepc_amic_automute(codec); + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; } -static void alc269_eeepc_amic_inithook(struct hda_codec *codec) +static void alc269_eeepc_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); - alc269_eeepc_amic_automute(codec); + alc_mic_automute(codec); } /* @@ -13222,89 +13260,10 @@ static struct hda_verb alc269_init_verbs[] = { { } }; -/* add playback controls from the parsed DAC table */ -static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, - const struct auto_pin_cfg *cfg) -{ - hda_nid_t nid; - int err; - - spec->multiout.num_dacs = 1; /* only use one dac */ - spec->multiout.dac_nids = spec->private_dac_nids; - spec->multiout.dac_nids[0] = 2; - - nid = cfg->line_out_pins[0]; - if (nid) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Front Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - } - - nid = cfg->speaker_pins[0]; - if (nid) { - if (!cfg->line_out_pins[0]) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Speaker Playback Volume", - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } - if (nid == 0x16) { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Speaker Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } else { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Speaker Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } - } - nid = cfg->hp_pins[0]; - if (nid) { - /* spec->multiout.hp_nid = 2; */ - if (!cfg->line_out_pins[0] && !cfg->speaker_pins[0]) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } - if (nid == 0x16) { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } else { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - HDA_OUTPUT)); - if (err < 0) - return err; - } - } - return 0; -} - -#define alc269_auto_create_analog_input_ctls \ - alc262_auto_create_analog_input_ctls +#define alc269_auto_create_multi_out_ctls \ + alc268_auto_create_multi_out_ctls +#define alc269_auto_create_input_ctls \ + alc268_auto_create_input_ctls #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc269_loopbacks alc880_loopbacks @@ -13354,7 +13313,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; - err = alc269_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc269_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -13379,15 +13338,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return err; if (!spec->cap_mixer && !spec->no_analog) - set_capture_mixer(spec); + set_capture_mixer(codec); alc_ssid_check(codec, 0x15, 0x1b, 0x14); return 1; } -#define alc269_auto_init_multi_out alc882_auto_init_multi_out -#define alc269_auto_init_hp_out alc882_auto_init_hp_out +#define alc269_auto_init_multi_out alc268_auto_init_multi_out +#define alc269_auto_init_hp_out alc268_auto_init_hp_out #define alc269_auto_init_analog_input alc882_auto_init_analog_input @@ -13455,6 +13414,7 @@ static struct alc_config_preset alc269_presets[] = { .channel_mode = alc269_modes, .input_mux = &alc269_capture_source, .unsol_event = alc269_quanta_fl1_unsol_event, + .setup = alc269_quanta_fl1_setup, .init_hook = alc269_quanta_fl1_init_hook, }, [ALC269_ASUS_EEEPC_P703] = { @@ -13467,9 +13427,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .input_mux = &alc269_eeepc_amic_capture_source, - .unsol_event = alc269_eeepc_amic_unsol_event, - .init_hook = alc269_eeepc_amic_inithook, + .unsol_event = alc269_eeepc_unsol_event, + .setup = alc269_eeepc_amic_setup, + .init_hook = alc269_eeepc_inithook, }, [ALC269_ASUS_EEEPC_P901] = { .mixers = { alc269_eeepc_mixer }, @@ -13481,9 +13441,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .input_mux = &alc269_eeepc_dmic_capture_source, - .unsol_event = alc269_eeepc_dmic_unsol_event, - .init_hook = alc269_eeepc_dmic_inithook, + .unsol_event = alc269_eeepc_unsol_event, + .setup = alc269_eeepc_dmic_setup, + .init_hook = alc269_eeepc_inithook, }, [ALC269_FUJITSU] = { .mixers = { alc269_fujitsu_mixer }, @@ -13495,9 +13455,9 @@ static struct alc_config_preset alc269_presets[] = { .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, - .input_mux = &alc269_eeepc_dmic_capture_source, - .unsol_event = alc269_eeepc_dmic_unsol_event, - .init_hook = alc269_eeepc_dmic_inithook, + .unsol_event = alc269_eeepc_unsol_event, + .setup = alc269_eeepc_dmic_setup, + .init_hook = alc269_eeepc_inithook, }, [ALC269_LIFEBOOK] = { .mixers = { alc269_lifebook_mixer }, @@ -13532,8 +13492,8 @@ static int patch_alc269(struct hda_codec *codec) alc269_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC269_AUTO; } @@ -13558,7 +13518,7 @@ static int patch_alc269(struct hda_codec *codec) } if (board_config != ALC269_AUTO) - setup_preset(spec, &alc269_presets[board_config]); + setup_preset(codec, &alc269_presets[board_config]); if (codec->subsystem_id == 0x17aa3bf8) { /* Due to a hardware problem on Lenovo Ideadpad, we need to @@ -13577,7 +13537,7 @@ static int patch_alc269(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); spec->capsrc_nids = alc269_capsrc_nids; if (!spec->cap_mixer) - set_capture_mixer(spec); + set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -14127,23 +14087,23 @@ static struct hda_verb alc861_auto_init_verbs[] = { {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* set Mic 1 */ @@ -14215,64 +14175,96 @@ static struct hda_input_mux alc861_capture_source = { }, }; +static hda_nid_t alc861_look_for_dac(struct hda_codec *codec, hda_nid_t pin) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t mix, srcs[5]; + int i, j, num; + + if (snd_hda_get_connections(codec, pin, &mix, 1) != 1) + return 0; + num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs)); + if (num < 0) + return 0; + for (i = 0; i < num; i++) { + unsigned int type; + type = get_wcaps_type(get_wcaps(codec, srcs[i])); + if (type != AC_WID_AUD_OUT) + continue; + for (j = 0; j < spec->multiout.num_dacs; j++) + if (spec->multiout.dac_nids[j] == srcs[i]) + break; + if (j >= spec->multiout.num_dacs) + return srcs[i]; + } + return 0; +} + /* fill in the dac_nids table from the parsed pin configuration */ -static int alc861_auto_fill_dac_nids(struct alc_spec *spec, +static int alc861_auto_fill_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { + struct alc_spec *spec = codec->spec; int i; - hda_nid_t nid; + hda_nid_t nid, dac; spec->multiout.dac_nids = spec->private_dac_nids; for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; - if (nid) { - if (i >= ARRAY_SIZE(alc861_dac_nids)) - continue; - spec->multiout.dac_nids[i] = alc861_dac_nids[i]; - } + dac = alc861_look_for_dac(codec, nid); + if (!dac) + continue; + spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; } - spec->multiout.num_dacs = cfg->line_outs; return 0; } +static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx, + hda_nid_t nid, unsigned int chs) +{ + char name[32]; + snprintf(name, sizeof(name), "%s Playback Switch", pfx); + return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); +} + /* add playback controls from the parsed DAC table */ -static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec, +static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - char name[32]; + struct alc_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; hda_nid_t nid; - int i, idx, err; + int i, err; + + if (cfg->line_outs == 1) { + const char *pfx = NULL; + if (!cfg->hp_outs) + pfx = "Master"; + else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + if (pfx) { + nid = spec->multiout.dac_nids[0]; + return alc861_create_out_sw(codec, pfx, nid, 3); + } + } for (i = 0; i < cfg->line_outs; i++) { nid = spec->multiout.dac_nids[i]; if (!nid) continue; - if (nid == 0x05) { + if (i == 2) { /* Center/LFE */ - err = add_control(spec, ALC_CTL_BIND_MUTE, - "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 0, - HDA_OUTPUT)); + err = alc861_create_out_sw(codec, "Center", nid, 1); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 0, - HDA_OUTPUT)); + err = alc861_create_out_sw(codec, "LFE", nid, 2); if (err < 0) return err; } else { - for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1; - idx++) - if (nid == alc861_dac_nids[idx]) - break; - sprintf(name, "%s Playback Switch", chname[idx]); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - HDA_OUTPUT)); + err = alc861_create_out_sw(codec, chname[i], nid, 3); if (err < 0) return err; } @@ -14280,8 +14272,9 @@ static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec, return 0; } -static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) +static int alc861_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin) { + struct alc_spec *spec = codec->spec; int err; hda_nid_t nid; @@ -14289,70 +14282,49 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) return 0; if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) { - nid = 0x03; - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Headphone Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; - spec->multiout.hp_nid = nid; + nid = alc861_look_for_dac(codec, pin); + if (nid) { + err = alc861_create_out_sw(codec, "Headphone", nid, 3); + if (err < 0) + return err; + spec->multiout.hp_nid = nid; + } } return 0; } /* create playback/capture controls for input pins */ -static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, +static int alc861_auto_create_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx, idx1; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - switch (cfg->input_pins[i]) { - case 0x0c: - idx1 = 1; - idx = 2; /* Line In */ - break; - case 0x0f: - idx1 = 2; - idx = 2; /* Line In */ - break; - case 0x0d: - idx1 = 0; - idx = 1; /* Mic In */ - break; - case 0x10: - idx1 = 3; - idx = 1; /* Mic In */ - break; - case 0x11: - idx1 = 4; - idx = 0; /* CD */ - break; - default: - continue; - } - - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], idx, 0x15); - if (err < 0) - return err; - - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx1; - imux->num_items++; - } - return 0; + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x08, 0); } static void alc861_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, - int pin_type, int dac_idx) + int pin_type, hda_nid_t dac) { + hda_nid_t mix, srcs[5]; + int i, num; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); - snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE, + snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + if (snd_hda_get_connections(codec, nid, &mix, 1) != 1) + return; + num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs)); + if (num < 0) + return; + for (i = 0; i < num; i++) { + unsigned int mute; + if (srcs[i] == dac || srcs[i] == 0x15) + mute = AMP_IN_UNMUTE(i); + else + mute = AMP_IN_MUTE(i); + snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE, + mute); + } } static void alc861_auto_init_multi_out(struct hda_codec *codec) @@ -14375,12 +14347,13 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin; pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ + if (pin) alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, - spec->multiout.dac_nids[0]); + spec->multiout.hp_nid); pin = spec->autocfg.speaker_pins[0]; if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); + alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, + spec->multiout.dac_nids[0]); } static void alc861_auto_init_analog_input(struct hda_codec *codec) @@ -14412,16 +14385,16 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if (!spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ - err = alc861_auto_fill_dac_nids(spec, &spec->autocfg); + err = alc861_auto_fill_dac_nids(codec, &spec->autocfg); if (err < 0) return err; - err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg); + err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg); if (err < 0) return err; - err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + err = alc861_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]); if (err < 0) return err; - err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc861_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -14440,7 +14413,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); - set_capture_mixer(spec); + set_capture_mixer(codec); alc_ssid_check(codec, 0x0e, 0x0f, 0x0b); @@ -14633,8 +14606,8 @@ static int patch_alc861(struct hda_codec *codec) alc861_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC861_AUTO; } @@ -14659,7 +14632,7 @@ static int patch_alc861(struct hda_codec *codec) } if (board_config != ALC861_AUTO) - setup_preset(spec, &alc861_presets[board_config]); + setup_preset(codec, &alc861_presets[board_config]); spec->stream_analog_playback = &alc861_pcm_analog_playback; spec->stream_analog_capture = &alc861_pcm_analog_capture; @@ -15062,12 +15035,15 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, bits); } -static void alc861vd_lenovo_init_hook(struct hda_codec *codec) +static void alc861vd_lenovo_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; +} + +static void alc861vd_lenovo_init_hook(struct hda_codec *codec) +{ alc_automute_amp(codec); alc861vd_lenovo_mic_automute(codec); } @@ -15131,13 +15107,12 @@ static struct hda_verb alc861vd_dallas_verbs[] = { }; /* toggle speaker-output according to the hp-jack state */ -static void alc861vd_dallas_init_hook(struct hda_codec *codec) +static void alc861vd_dallas_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - alc_automute_amp(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -15251,6 +15226,7 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, .unsol_event = alc861vd_lenovo_unsol_event, + .setup = alc861vd_lenovo_setup, .init_hook = alc861vd_lenovo_init_hook, }, [ALC861VD_DALLAS] = { @@ -15262,7 +15238,8 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_dallas_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc861vd_dallas_init_hook, + .setup = alc861vd_dallas_setup, + .init_hook = alc_automute_amp, }, [ALC861VD_HP] = { .mixers = { alc861vd_hp_mixer }, @@ -15274,7 +15251,8 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_hp_capture_source, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc861vd_dallas_init_hook, + .setup = alc861vd_dallas_setup, + .init_hook = alc_automute_amp, }, [ALC660VD_ASUS_V1S] = { .mixers = { alc861vd_lenovo_mixer }, @@ -15289,6 +15267,7 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, .unsol_event = alc861vd_lenovo_unsol_event, + .setup = alc861vd_lenovo_setup, .init_hook = alc861vd_lenovo_init_hook, }, }; @@ -15296,6 +15275,13 @@ static struct alc_config_preset alc861vd_presets[] = { /* * BIOS auto configuration */ +static int alc861vd_auto_create_input_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0); +} + + static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, int dac_idx) { @@ -15330,7 +15316,6 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec) alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc861vd_is_input_pin(nid) alc880_is_input_pin(nid) #define ALC861VD_PIN_CD_NID ALC880_PIN_CD_NID static void alc861vd_auto_init_analog_input(struct hda_codec *codec) @@ -15340,7 +15325,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc861vd_is_input_pin(nid)) { + if (alc_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC861VD_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) @@ -15404,13 +15389,25 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } else { - sprintf(name, "%s Playback Volume", chname[i]); + const char *pfx; + if (cfg->line_outs == 1 && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { + if (!cfg->hp_pins) + pfx = "Speaker"; + else + pfx = "PCM"; + } else + pfx = chname[i]; + sprintf(name, "%s Playback Volume", pfx); err = add_control(spec, ALC_CTL_WIDGET_VOL, name, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", chname[i]); + if (cfg->line_outs == 1 && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + sprintf(name, "%s Playback Switch", pfx); err = add_control(spec, ALC_CTL_BIND_MUTE, name, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); @@ -15503,7 +15500,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc861vd_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -15557,8 +15554,8 @@ static int patch_alc861vd(struct hda_codec *codec) alc861vd_cfg_tbl); if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC861VD_AUTO; } @@ -15583,7 +15580,7 @@ static int patch_alc861vd(struct hda_codec *codec) } if (board_config != ALC861VD_AUTO) - setup_preset(spec, &alc861vd_presets[board_config]); + setup_preset(codec, &alc861vd_presets[board_config]); if (codec->vendor_id == 0x10ec0660) { /* always turn on EAPD */ @@ -15603,7 +15600,7 @@ static int patch_alc861vd(struct hda_codec *codec) if (!spec->capsrc_nids) spec->capsrc_nids = alc861vd_capsrc_nids; - set_capture_mixer(spec); + set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -15644,9 +15641,9 @@ static hda_nid_t alc272_dac_nids[2] = { 0x02, 0x03 }; -static hda_nid_t alc662_adc_nids[1] = { +static hda_nid_t alc662_adc_nids[2] = { /* ADC1-2 */ - 0x09, + 0x09, 0x08 }; static hda_nid_t alc272_adc_nids[1] = { @@ -15654,7 +15651,7 @@ static hda_nid_t alc272_adc_nids[1] = { 0x08, }; -static hda_nid_t alc662_capsrc_nids[1] = { 0x22 }; +static hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 }; static hda_nid_t alc272_capsrc_nids[1] = { 0x23 }; @@ -15678,14 +15675,6 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = { }, }; -static struct hda_input_mux alc662_eeepc_capture_source = { - .num_items = 2, - .items = { - { "i-Mic", 0x1 }, - { "e-Mic", 0x0 }, - }, -}; - static struct hda_input_mux alc663_capture_source = { .num_items = 3, .items = { @@ -15695,23 +15684,7 @@ static struct hda_input_mux alc663_capture_source = { }, }; -static struct hda_input_mux alc663_m51va_capture_source = { - .num_items = 2, - .items = { - { "Ext-Mic", 0x0 }, - { "D-Mic", 0x9 }, - }, -}; - -#if 1 /* set to 0 for testing other input sources below */ -static struct hda_input_mux alc272_nc10_capture_source = { - .num_items = 2, - .items = { - { "Autoselect Mic", 0x0 }, - { "Internal Mic", 0x1 }, - }, -}; -#else +#if 0 /* set to 1 for testing other input sources below */ static struct hda_input_mux alc272_nc10_capture_source = { .num_items = 16, .items = { @@ -16380,47 +16353,44 @@ static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec, alc662_lenovo_101e_ispeaker_automute(codec); } -static void alc662_eeepc_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); - snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); -} - /* unsolicited event for HP jack sensing */ static void alc662_eeepc_unsol_event(struct hda_codec *codec, unsigned int res) { if ((res >> 26) == ALC880_MIC_EVENT) - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); else alc262_hippo_unsol_event(codec, res); } +static void alc662_eeepc_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + alc262_hippo1_setup(codec); + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; +} + static void alc662_eeepc_inithook(struct hda_codec *codec) { - alc262_hippo1_init_hook(codec); - alc662_eeepc_mic_automute(codec); + alc262_hippo_automute(codec); + alc_mic_automute(codec); } -static void alc662_eeepc_ep20_inithook(struct hda_codec *codec) +static void alc662_eeepc_ep20_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - alc262_hippo_master_update(codec); } +#define alc662_eeepc_ep20_inithook alc262_hippo_master_update + static void alc663_m51va_speaker_automute(struct hda_codec *codec) { unsigned int present; @@ -16531,23 +16501,6 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) } } -static void alc663_m51va_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); - snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); - snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); -} - static void alc663_m51va_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -16556,36 +16509,32 @@ static void alc663_m51va_unsol_event(struct hda_codec *codec, alc663_m51va_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc663_m51va_mic_automute(codec); + alc_mic_automute(codec); break; } } +static void alc663_m51va_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; +} + static void alc663_m51va_inithook(struct hda_codec *codec) { alc663_m51va_speaker_automute(codec); - alc663_m51va_mic_automute(codec); + alc_mic_automute(codec); } /* ***************** Mode1 ******************************/ -static void alc663_mode1_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case ALC880_HP_EVENT: - alc663_m51va_speaker_automute(codec); - break; - case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); - break; - } -} +#define alc663_mode1_unsol_event alc663_m51va_unsol_event +#define alc663_mode1_setup alc663_m51va_setup +#define alc663_mode1_inithook alc663_m51va_inithook -static void alc663_mode1_inithook(struct hda_codec *codec) -{ - alc663_m51va_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); -} /* ***************** Mode2 ******************************/ static void alc662_mode2_unsol_event(struct hda_codec *codec, unsigned int res) @@ -16595,15 +16544,17 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec, alc662_f5z_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc662_mode2_setup alc663_m51va_setup + static void alc662_mode2_inithook(struct hda_codec *codec) { alc662_f5z_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } /* ***************** Mode3 ******************************/ static void alc663_mode3_unsol_event(struct hda_codec *codec, @@ -16614,15 +16565,17 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec, alc663_two_hp_m1_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc663_mode3_setup alc663_m51va_setup + static void alc663_mode3_inithook(struct hda_codec *codec) { alc663_two_hp_m1_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } /* ***************** Mode4 ******************************/ static void alc663_mode4_unsol_event(struct hda_codec *codec, @@ -16633,15 +16586,17 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec, alc663_21jd_two_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc663_mode4_setup alc663_m51va_setup + static void alc663_mode4_inithook(struct hda_codec *codec) { alc663_21jd_two_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } /* ***************** Mode5 ******************************/ static void alc663_mode5_unsol_event(struct hda_codec *codec, @@ -16652,15 +16607,17 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec, alc663_15jd_two_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc663_mode5_setup alc663_m51va_setup + static void alc663_mode5_inithook(struct hda_codec *codec) { alc663_15jd_two_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } /* ***************** Mode6 ******************************/ static void alc663_mode6_unsol_event(struct hda_codec *codec, @@ -16671,15 +16628,17 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec, alc663_two_hp_m2_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc663_mode6_setup alc663_m51va_setup + static void alc663_mode6_inithook(struct hda_codec *codec) { alc663_two_hp_m2_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } static void alc663_g71v_hp_automute(struct hda_codec *codec) @@ -16721,16 +16680,18 @@ static void alc663_g71v_unsol_event(struct hda_codec *codec, alc663_g71v_front_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc663_g71v_setup alc663_m51va_setup + static void alc663_g71v_inithook(struct hda_codec *codec) { alc663_g71v_front_automute(codec); alc663_g71v_hp_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } static void alc663_g50v_unsol_event(struct hda_codec *codec, @@ -16741,15 +16702,17 @@ static void alc663_g50v_unsol_event(struct hda_codec *codec, alc663_m51va_speaker_automute(codec); break; case ALC880_MIC_EVENT: - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); break; } } +#define alc663_g50v_setup alc663_m51va_setup + static void alc663_g50v_inithook(struct hda_codec *codec) { alc663_m51va_speaker_automute(codec); - alc662_eeepc_mic_automute(codec); + alc_mic_automute(codec); } static struct snd_kcontrol_new alc662_ecs_mixer[] = { @@ -16953,8 +16916,8 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc662_eeepc_unsol_event, + .setup = alc662_eeepc_setup, .init_hook = alc662_eeepc_inithook, }, [ALC662_ASUS_EEEPC_EP20] = { @@ -16968,6 +16931,7 @@ static struct alc_config_preset alc662_presets[] = { .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc662_lenovo_101e_capture_source, .unsol_event = alc662_eeepc_unsol_event, + .setup = alc662_eeepc_ep20_setup, .init_hook = alc662_eeepc_ep20_inithook, }, [ALC662_ECS] = { @@ -16978,8 +16942,8 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc662_eeepc_unsol_event, + .setup = alc662_eeepc_setup, .init_hook = alc662_eeepc_inithook, }, [ALC663_ASUS_M51VA] = { @@ -16990,8 +16954,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc663_m51va_capture_source, .unsol_event = alc663_m51va_unsol_event, + .setup = alc663_m51va_setup, .init_hook = alc663_m51va_inithook, }, [ALC663_ASUS_G71V] = { @@ -17002,8 +16966,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc663_g71v_unsol_event, + .setup = alc663_g71v_setup, .init_hook = alc663_g71v_inithook, }, [ALC663_ASUS_H13] = { @@ -17013,7 +16977,6 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc663_m51va_capture_source, .unsol_event = alc663_m51va_unsol_event, .init_hook = alc663_m51va_inithook, }, @@ -17027,6 +16990,7 @@ static struct alc_config_preset alc662_presets[] = { .channel_mode = alc662_3ST_6ch_modes, .input_mux = &alc663_capture_source, .unsol_event = alc663_g50v_unsol_event, + .setup = alc663_g50v_setup, .init_hook = alc663_g50v_inithook, }, [ALC663_ASUS_MODE1] = { @@ -17040,8 +17004,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc663_mode1_unsol_event, + .setup = alc663_mode1_setup, .init_hook = alc663_mode1_inithook, }, [ALC662_ASUS_MODE2] = { @@ -17054,8 +17018,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc662_mode2_unsol_event, + .setup = alc662_mode2_setup, .init_hook = alc662_mode2_inithook, }, [ALC663_ASUS_MODE3] = { @@ -17069,8 +17033,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc663_mode3_unsol_event, + .setup = alc663_mode3_setup, .init_hook = alc663_mode3_inithook, }, [ALC663_ASUS_MODE4] = { @@ -17084,8 +17048,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc663_mode4_unsol_event, + .setup = alc663_mode4_setup, .init_hook = alc663_mode4_inithook, }, [ALC663_ASUS_MODE5] = { @@ -17099,8 +17063,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc663_mode5_unsol_event, + .setup = alc663_mode5_setup, .init_hook = alc663_mode5_inithook, }, [ALC663_ASUS_MODE6] = { @@ -17114,8 +17078,8 @@ static struct alc_config_preset alc662_presets[] = { .dig_out_nid = ALC662_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc662_eeepc_capture_source, .unsol_event = alc663_mode6_unsol_event, + .setup = alc663_mode6_setup, .init_hook = alc663_mode6_inithook, }, [ALC272_DELL] = { @@ -17129,8 +17093,8 @@ static struct alc_config_preset alc662_presets[] = { .num_adc_nids = ARRAY_SIZE(alc272_adc_nids), .capsrc_nids = alc272_capsrc_nids, .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc663_m51va_capture_source, .unsol_event = alc663_m51va_unsol_event, + .setup = alc663_m51va_setup, .init_hook = alc663_m51va_inithook, }, [ALC272_DELL_ZM1] = { @@ -17141,11 +17105,11 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc662_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .adc_nids = alc662_adc_nids, - .num_adc_nids = ARRAY_SIZE(alc662_adc_nids), + .num_adc_nids = 1, .capsrc_nids = alc662_capsrc_nids, .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc663_m51va_capture_source, .unsol_event = alc663_m51va_unsol_event, + .setup = alc663_m51va_setup, .init_hook = alc663_m51va_inithook, }, [ALC272_SAMSUNG_NC10] = { @@ -17156,8 +17120,9 @@ static struct alc_config_preset alc662_presets[] = { .dac_nids = alc272_dac_nids, .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), .channel_mode = alc662_3ST_2ch_modes, - .input_mux = &alc272_nc10_capture_source, + /*.input_mux = &alc272_nc10_capture_source,*/ .unsol_event = alc663_mode4_unsol_event, + .setup = alc663_mode4_setup, .init_hook = alc663_mode4_inithook, }, }; @@ -17209,13 +17174,25 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } else { - sprintf(name, "%s Playback Volume", chname[i]); + const char *pfx; + if (cfg->line_outs == 1 && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { + if (!cfg->hp_pins) + pfx = "Speaker"; + else + pfx = "PCM"; + } else + pfx = chname[i]; + sprintf(name, "%s Playback Volume", pfx); err = add_control(spec, ALC_CTL_WIDGET_VOL, name, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", chname[i]); + if (cfg->line_outs == 1 && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + pfx = "Speaker"; + sprintf(name, "%s Playback Switch", pfx); err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i), 3, 0, HDA_INPUT)); @@ -17277,62 +17254,9 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return 0; } -/* return the index of the src widget from the connection list of the nid. - * return -1 if not found - */ -static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid, - hda_nid_t src) -{ - hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; - int i, conns; - - conns = snd_hda_get_connections(codec, nid, conn_list, - ARRAY_SIZE(conn_list)); - if (conns < 0) - return -1; - for (i = 0; i < conns; i++) - if (conn_list[i] == src) - return i; - return -1; -} - -static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int pincap = snd_hda_query_pin_caps(codec, nid); - return (pincap & AC_PINCAP_IN) != 0; -} - /* create playback/capture controls for input pins */ -static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) -{ - struct alc_spec *spec = codec->spec; - struct hda_input_mux *imux = &spec->private_imux[0]; - int i, err, idx; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - if (alc662_is_input_pin(codec, cfg->input_pins[i])) { - idx = alc662_input_pin_idx(codec, 0x0b, - cfg->input_pins[i]); - if (idx >= 0) { - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], - idx, 0x0b); - if (err < 0) - return err; - } - idx = alc662_input_pin_idx(codec, 0x22, - cfg->input_pins[i]); - if (idx >= 0) { - imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx; - imux->num_items++; - } - } - } - return 0; -} +#define alc662_auto_create_input_ctls \ + alc880_auto_create_input_ctls static void alc662_auto_set_output_and_unmute(struct hda_codec *codec, hda_nid_t nid, int pin_type, @@ -17386,7 +17310,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc662_is_input_pin(codec, nid)) { + if (alc_is_input_pin(codec, nid)) { alc_set_input_pin(codec, nid, i); if (nid != ALC662_PIN_CD_NID && (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) @@ -17427,7 +17351,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg); + err = alc662_auto_create_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -17484,8 +17408,8 @@ static int patch_alc662(struct hda_codec *codec) alc662_models, alc662_cfg_tbl); if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for %s, " - "trying auto-probe from BIOS...\n", codec->chip_name); + printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); board_config = ALC662_AUTO; } @@ -17510,7 +17434,7 @@ static int patch_alc662(struct hda_codec *codec) } if (board_config != ALC662_AUTO) - setup_preset(spec, &alc662_presets[board_config]); + setup_preset(codec, &alc662_presets[board_config]); spec->stream_analog_playback = &alc662_pcm_analog_playback; spec->stream_analog_capture = &alc662_pcm_analog_capture; @@ -17526,7 +17450,7 @@ static int patch_alc662(struct hda_codec *codec) spec->capsrc_nids = alc662_capsrc_nids; if (!spec->cap_mixer) - set_capture_mixer(spec); + set_capture_mixer(codec); if (codec->vendor_id == 0x10ec0662) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); else @@ -17562,23 +17486,23 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 }, { .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd }, { .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2", - .patch = patch_alc883 }, + .patch = patch_alc882 }, { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1", .patch = patch_alc662 }, { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, - { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, + { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A", - .patch = patch_alc882 }, /* should be patch_alc883() in future */ + .patch = patch_alc882 }, { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", - .patch = patch_alc882 }, /* should be patch_alc883() in future */ + .patch = patch_alc882 }, { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, - { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, + { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", - .patch = patch_alc883 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, - { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, + .patch = patch_alc882 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 }, + { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6990cfcb6a38..e31e53dc6962 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -40,6 +40,8 @@ enum { STAC_INSERT_EVENT, STAC_PWR_EVENT, STAC_HP_EVENT, + STAC_LO_EVENT, + STAC_MIC_EVENT, }; enum { @@ -81,6 +83,7 @@ enum { STAC_DELL_M6_DMIC, STAC_DELL_M6_BOTH, STAC_DELL_EQ, + STAC_ALIENWARE_M17X, STAC_92HD73XX_MODELS }; @@ -177,6 +180,12 @@ struct sigmatel_jack { struct snd_jack *jack; }; +struct sigmatel_mic_route { + hda_nid_t pin; + unsigned char mux_idx; + unsigned char dmux_idx; +}; + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -188,6 +197,7 @@ struct sigmatel_spec { unsigned int hp_detect: 1; unsigned int spdif_mute: 1; unsigned int check_volume_offset:1; + unsigned int auto_mic:1; /* gpio lines */ unsigned int eapd_mask; @@ -219,7 +229,6 @@ struct sigmatel_spec { /* playback */ struct hda_input_mux *mono_mux; - struct hda_input_mux *amp_mux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -239,6 +248,15 @@ struct sigmatel_spec { unsigned int num_dmuxes; hda_nid_t *smux_nids; unsigned int num_smuxes; + unsigned int num_analog_muxes; + + unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */ + unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */ + unsigned int num_caps; /* number of capture volume/switch elements */ + + struct sigmatel_mic_route ext_mic; + struct sigmatel_mic_route int_mic; + const char **spdif_labels; hda_nid_t dig_in_nid; @@ -263,7 +281,6 @@ struct sigmatel_spec { unsigned int cur_smux[2]; unsigned int cur_amux; hda_nid_t *amp_nids; - unsigned int num_amps; unsigned int powerdown_adcs; /* i/o switches */ @@ -282,7 +299,6 @@ struct sigmatel_spec { struct hda_input_mux private_dimux; struct hda_input_mux private_imux; struct hda_input_mux private_smux; - struct hda_input_mux private_amp_mux; struct hda_input_mux private_mono_mux; }; @@ -311,11 +327,6 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; -#define DELL_M6_AMP 2 -static hda_nid_t stac92hd73xx_amp_nids[3] = { - 0x0b, 0x0c, 0x0e -}; - #define STAC92HD73XX_NUM_DMICS 2 static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -323,8 +334,8 @@ static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { #define STAC92HD73_DAC_COUNT 5 -static hda_nid_t stac92hd73xx_mux_nids[4] = { - 0x28, 0x29, 0x2a, 0x2b, +static hda_nid_t stac92hd73xx_mux_nids[2] = { + 0x20, 0x21, }; static hda_nid_t stac92hd73xx_dmux_nids[2] = { @@ -335,14 +346,16 @@ static hda_nid_t stac92hd73xx_smux_nids[2] = { 0x22, 0x23, }; -#define STAC92HD83XXX_NUM_DMICS 2 -static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { - 0x11, 0x12, 0 +#define STAC92HD73XX_NUM_CAPS 2 +static unsigned long stac92hd73xx_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), }; +#define stac92hd73xx_capsws stac92hd73xx_capvols #define STAC92HD83_DAC_COUNT 3 -static hda_nid_t stac92hd83xxx_dmux_nids[2] = { +static hda_nid_t stac92hd83xxx_mux_nids[2] = { 0x17, 0x18, }; @@ -362,9 +375,12 @@ static unsigned int stac92hd83xxx_pwr_mapping[4] = { 0x03, 0x0c, 0x20, 0x40, }; -static hda_nid_t stac92hd83xxx_amp_nids[1] = { - 0xc, +#define STAC92HD83XXX_NUM_CAPS 2 +static unsigned long stac92hd83xxx_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_OUTPUT), }; +#define stac92hd83xxx_capsws stac92hd83xxx_capvols static hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f @@ -395,6 +411,13 @@ static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { 0x22, 0 }; +#define STAC92HD71BXX_NUM_CAPS 2 +static unsigned long stac92hd71bxx_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), +}; +#define stac92hd71bxx_capsws stac92hd71bxx_capvols + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -416,6 +439,13 @@ static hda_nid_t stac925x_dmux_nids[1] = { 0x14, }; +static unsigned long stac925x_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT), +}; +static unsigned long stac925x_capsws[] = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), +}; + static hda_nid_t stac922x_adc_nids[2] = { 0x06, 0x07, }; @@ -424,6 +454,13 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +#define STAC922X_NUM_CAPS 2 +static unsigned long stac922x_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), +}; +#define stac922x_capsws stac922x_capvols + static hda_nid_t stac927x_slave_dig_outs[2] = { 0x1f, 0, }; @@ -453,6 +490,18 @@ static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 }; +#define STAC927X_NUM_CAPS 3 +static unsigned long stac927x_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT), +}; +static unsigned long stac927x_capsws[] = { + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), +}; + static const char *stac927x_spdif_labels[5] = { "Digital Playback", "ADAT", "Analog Mux 1", "Analog Mux 2", "Analog Mux 3" @@ -479,6 +528,16 @@ static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 }; +#define STAC9205_NUM_CAPS 2 +static unsigned long stac9205_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT), +}; +static unsigned long stac9205_capsws[] = { + HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT), +}; + static hda_nid_t stac9200_pin_nids[8] = { 0x08, 0x09, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, @@ -529,34 +588,6 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x21, 0x22, }; -#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info - -static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = spec->amp_nids[spec->cur_amux]; - - kcontrol->private_value ^= get_amp_nid(kcontrol); - kcontrol->private_value |= nid; - - return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); -} - -static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid = spec->amp_nids[spec->cur_amux]; - - kcontrol->private_value ^= get_amp_nid(kcontrol); - kcontrol->private_value |= nid; - - return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); -} - static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -693,9 +724,35 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); + const struct hda_input_mux *imux = spec->input_mux; + unsigned int idx, prev_idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + prev_idx = spec->cur_mux[adc_idx]; + if (prev_idx == idx) + return 0; + if (idx < spec->num_analog_muxes) { + snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[idx].index); + if (prev_idx >= spec->num_analog_muxes) { + imux = spec->dinput_mux; + /* 0 = analog */ + snd_hda_codec_write_cache(codec, + spec->dmux_nids[adc_idx], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } + } else { + imux = spec->dinput_mux; + snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[idx - 1].index); + } + spec->cur_mux[adc_idx] = idx; + return 1; } static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol, @@ -726,41 +783,6 @@ static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, spec->mono_nid, &spec->cur_mmux); } -static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - return snd_hda_input_mux_info(spec->amp_mux, uinfo); -} - -static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->cur_amux; - return 0; -} - -static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - struct snd_kcontrol *ctl = - snd_hda_find_mixer_ctl(codec, "Amp Capture Volume"); - if (!ctl) - return -EINVAL; - - snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | - SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); - - return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol, - 0, &spec->cur_amux); -} - #define stac92xx_aloopback_info snd_ctl_boolean_mono_info static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, @@ -828,84 +850,16 @@ static struct hda_verb stac9200_eapd_init[] = { {} }; -static struct hda_verb stac92hd73xx_6ch_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup adcs to point to mixer */ - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* setup import muxs */ - { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - static struct hda_verb dell_eq_core_init[] = { /* set master volume to max value without distortion * and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, - /* setup adcs to point to mixer */ - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - /* setup import muxs */ - { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - -static struct hda_verb dell_m6_core_init[] = { - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup adcs to point to mixer */ - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - /* setup import muxs */ - { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {} -}; - -static struct hda_verb stac92hd73xx_8ch_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* setup adcs to point to mixer */ - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* setup import muxs */ - { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, {} }; -static struct hda_verb stac92hd73xx_10ch_core_init[] = { +static struct hda_verb stac92hd73xx_core_init[] = { /* set master volume and direct control */ { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* dac3 is connected to import3 mux */ - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f}, - /* setup adcs to point to mixer */ - { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* setup import muxs */ - { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, {} }; @@ -925,19 +879,6 @@ static struct hda_verb stac92hd71bxx_core_init[] = { {} }; -#define HD_DISABLE_PORTF 1 -static struct hda_verb stac92hd71bxx_analog_core_init[] = { - /* start of config #1 */ - - /* connect port 0f to audio mixer */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* start of config #2 */ - - /* set master volume and direct control */ - { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - {} -}; - static struct hda_verb stac92hd71bxx_unmute_core_init[] = { /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -996,31 +937,6 @@ static struct hda_verb stac9205_core_init[] = { .put = stac92xx_mono_mux_enum_put, \ } -#define STAC_AMP_MUX \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Amp Selector Capture Switch", \ - .count = 1, \ - .info = stac92xx_amp_mux_enum_info, \ - .get = stac92xx_amp_mux_enum_get, \ - .put = stac92xx_amp_mux_enum_put, \ - } - -#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ - .info = stac92xx_amp_volume_info, \ - .get = stac92xx_amp_volume_get, \ - .put = stac92xx_amp_volume_put, \ - .tlv = { .c = snd_hda_mixer_amp_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -1051,34 +967,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { { } /* end */ }; -#define DELL_M6_MIXER 6 -static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { - /* start of config #1 */ - HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), - - /* start of config #2 */ - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - - { } /* end */ -}; - static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), {} @@ -1094,134 +982,14 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { {} }; -static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), - - HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), - { } /* end */ -}; - -static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), - - HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), - { } /* end */ -}; - - -static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT), - - HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT), - - /* - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT), - */ - { } /* end */ -}; - -static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - /* analog pc-beep replaced with digital beep support */ - /* - HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), - */ - - HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT), - - HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT), - - HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), - - HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT), - { } /* end */ -}; static struct snd_kcontrol_new stac92hd71bxx_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) }; -static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - { } /* end */ -}; - static struct snd_kcontrol_new stac925x_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), - { } /* end */ -}; - -static struct snd_kcontrol_new stac9205_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -1230,29 +998,6 @@ static struct snd_kcontrol_new stac9205_loopback[] = { {} }; -/* This needs to be generated dynamically based on sequence */ -static struct snd_kcontrol_new stac922x_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), - { } /* end */ -}; - - -static struct snd_kcontrol_new stac927x_mixer[] = { - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), - { } /* end */ -}; - static struct snd_kcontrol_new stac927x_loopback[] = { STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), {} @@ -1310,16 +1055,19 @@ static int stac92xx_build_controls(struct hda_codec *codec) int err; int i; - err = snd_hda_add_new_ctls(codec, spec->mixer); - if (err < 0) - return err; + if (spec->mixer) { + err = snd_hda_add_new_ctls(codec, spec->mixer); + if (err < 0) + return err; + } for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); if (err < 0) return err; } - if (spec->num_dmuxes > 0) { + if (!spec->auto_mic && spec->num_dmuxes > 0 && + snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { stac_dmux_mixer.count = spec->num_dmuxes; err = snd_hda_ctl_add(codec, snd_ctl_new1(&stac_dmux_mixer, codec)); @@ -1766,12 +1514,20 @@ static unsigned int dell_m6_pin_configs[13] = { 0x4f0000f0, }; +static unsigned int alienware_m17x_pin_configs[13] = { + 0x0321101f, 0x0321101f, 0x03a11020, 0x03014020, + 0x90170110, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, + 0x904601b0, +}; + static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, [STAC_DELL_M6_AMIC] = dell_m6_pin_configs, [STAC_DELL_M6_DMIC] = dell_m6_pin_configs, [STAC_DELL_M6_BOTH] = dell_m6_pin_configs, [STAC_DELL_EQ] = dell_m6_pin_configs, + [STAC_ALIENWARE_M17X] = alienware_m17x_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { @@ -1783,6 +1539,7 @@ static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_DELL_M6_DMIC] = "dell-m6-dmic", [STAC_DELL_M6_BOTH] = "dell-m6", [STAC_DELL_EQ] = "dell-eq", + [STAC_ALIENWARE_M17X] = "alienware", }; static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { @@ -1820,6 +1577,12 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; +static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1, + "Alienware M17x", STAC_ALIENWARE_M17X), + {} /* terminator */ +}; + static unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, @@ -1927,6 +1690,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, "HP HDX", STAC_HP_HDX), /* HDX16 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, + "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -2642,8 +2407,7 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol, return 0; } -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid, - unsigned char type); +static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid); static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -2657,7 +2421,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, /* check to be sure that the ports are upto date with * switch changes */ - stac_issue_unsol_event(codec, nid, STAC_HP_EVENT); + stac_issue_unsol_event(codec, nid); return 1; } @@ -2790,7 +2554,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ * appropriately according to the pin direction */ if (spec->hp_detect) - stac_issue_unsol_event(codec, nid, STAC_HP_EVENT); + stac_issue_unsol_event(codec, nid); return 1; } @@ -2859,8 +2623,6 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, - STAC_CTL_WIDGET_AMP_MUX, - STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH, @@ -2871,8 +2633,6 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, - STAC_AMP_MUX, - STAC_AMP_VOL(NULL, 0, 0, 0, 0), STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), @@ -2973,6 +2733,8 @@ static int stac92xx_add_input_source(struct sigmatel_spec *spec) struct snd_kcontrol_new *knew; struct hda_input_mux *imux = &spec->private_imux; + if (spec->auto_mic) + return 0; /* no need for input source */ if (!spec->num_adcs || imux->num_items <= 1) return 0; /* no need for input source control */ knew = stac_control_new(spec, &stac_input_src_temp, @@ -3066,7 +2828,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) HDA_MAX_CONNECTIONS); for (j = 0; j < conn_len; j++) { wcaps = get_wcaps(codec, conn[j]); - wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + wtype = get_wcaps_type(wcaps); /* we check only analog outputs */ if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) continue; @@ -3325,6 +3087,21 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, return 0; } +static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol, + unsigned long sw, int idx) +{ + int err; + err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx, + "Capture Volume", vol); + if (err < 0) + return err; + err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_MUTE, idx, + "Capture Switch", sw); + if (err < 0) + return err; + return 0; +} + /* add playback controls from the parsed DAC table */ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) @@ -3398,7 +3175,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) spec->mono_nid, con_lst, HDA_MAX_NUM_INPUTS); - if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_mono_labels)) + if (num_cons <= 0 || num_cons > ARRAY_SIZE(stac92xx_mono_labels)) return -EINVAL; for (i = 0; i < num_cons; i++) { @@ -3412,37 +3189,6 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) "Mono Mux", spec->mono_nid); } -/* labels for amp mux outputs */ -static const char *stac92xx_amp_labels[3] = { - "Front Microphone", "Microphone", "Line In", -}; - -/* create amp out controls mux on capable codecs */ -static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec) -{ - struct sigmatel_spec *spec = codec->spec; - struct hda_input_mux *amp_mux = &spec->private_amp_mux; - int i, err; - - for (i = 0; i < spec->num_amps; i++) { - amp_mux->items[amp_mux->num_items].label = - stac92xx_amp_labels[i]; - amp_mux->items[amp_mux->num_items].index = i; - amp_mux->num_items++; - } - - if (spec->num_amps > 1) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, - "Amp Selector Capture Switch", 0); - if (err < 0) - return err; - } - return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, - "Amp Capture Volume", - HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); -} - - /* create PC beep volume controls */ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, hda_nid_t nid) @@ -3511,19 +3257,33 @@ static int stac92xx_beep_switch_ctl(struct hda_codec *codec) static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int wcaps, nid, i, err = 0; + int i, j, err = 0; for (i = 0; i < spec->num_muxes; i++) { + hda_nid_t nid; + unsigned int wcaps; + unsigned long val; + nid = spec->mux_nids[i]; wcaps = get_wcaps(codec, nid); + if (!(wcaps & AC_WCAP_OUT_AMP)) + continue; - if (wcaps & AC_WCAP_OUT_AMP) { - err = stac92xx_add_control_idx(spec, - STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume", - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); - if (err < 0) - return err; + /* check whether already the same control was created as + * normal Capture Volume. + */ + val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + for (j = 0; j < spec->num_caps; j++) { + if (spec->capvols[j] == val) + break; } + if (j < spec->num_caps) + continue; + + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, i, + "Mux Capture Volume", val); + if (err < 0) + return err; } return 0; }; @@ -3544,7 +3304,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) spec->smux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); - if (!num_cons) + if (num_cons <= 0) return -EINVAL; if (!labels) @@ -3565,101 +3325,231 @@ static const char *stac92xx_dmic_labels[5] = { "Digital Mic 3", "Digital Mic 4" }; +static int get_connection_index(struct hda_codec *codec, hda_nid_t mux, + hda_nid_t nid) +{ + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int i, nums; + + nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn)); + for (i = 0; i < nums; i++) + if (conn[i] == nid) + return i; + return -1; +} + +/* create a volume assigned to the given pin (only if supported) */ +/* return 1 if the volume control is created */ +static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid, + const char *label, int direction) +{ + unsigned int caps, nums; + char name[32]; + int err; + + if (direction == HDA_OUTPUT) + caps = AC_WCAP_OUT_AMP; + else + caps = AC_WCAP_IN_AMP; + if (!(get_wcaps(codec, nid) & caps)) + return 0; + caps = query_amp_caps(codec, nid, direction); + nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (!nums) + return 0; + snprintf(name, sizeof(name), "%s Capture Volume", label); + err = stac92xx_add_control(codec->spec, STAC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction)); + if (err < 0) + return err; + return 1; +} + /* create playback/capture controls for input pins on dmic capable codecs */ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *imux = &spec->private_imux; struct hda_input_mux *dimux = &spec->private_dimux; - hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int err, i, j; - char name[32]; + int err, i, active_mics; + unsigned int def_conf; dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; dimux->items[dimux->num_items].index = 0; dimux->num_items++; + active_mics = 0; + for (i = 0; i < spec->num_dmics; i++) { + /* check the validity: sometimes it's a dead vendor-spec node */ + if (get_wcaps_type(get_wcaps(codec, spec->dmic_nids[i])) + != AC_WID_PIN) + continue; + def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); + if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE) + active_mics++; + } + for (i = 0; i < spec->num_dmics; i++) { hda_nid_t nid; int index; - int num_cons; - unsigned int wcaps; - unsigned int def_conf; + const char *label; - def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); + nid = spec->dmic_nids[i]; + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + continue; + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; - nid = spec->dmic_nids[i]; - num_cons = snd_hda_get_connections(codec, - spec->dmux_nids[0], - con_lst, - HDA_MAX_NUM_INPUTS); - for (j = 0; j < num_cons; j++) - if (con_lst[j] == nid) { - index = j; - goto found; - } - continue; -found: - wcaps = get_wcaps(codec, nid) & - (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); + index = get_connection_index(codec, spec->dmux_nids[0], nid); + if (index < 0) + continue; - if (wcaps) { - sprintf(name, "%s Capture Volume", - stac92xx_dmic_labels[dimux->num_items]); + if (active_mics == 1) + label = "Digital Mic"; + else + label = stac92xx_dmic_labels[dimux->num_items]; - err = stac92xx_add_control(spec, - STAC_CTL_WIDGET_VOL, - name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, - (wcaps & AC_WCAP_OUT_AMP) ? - HDA_OUTPUT : HDA_INPUT)); + err = create_elem_capture_vol(codec, nid, label, HDA_INPUT); + if (err < 0) + return err; + if (!err) { + err = create_elem_capture_vol(codec, nid, label, + HDA_OUTPUT); if (err < 0) return err; } - dimux->items[dimux->num_items].label = - stac92xx_dmic_labels[dimux->num_items]; + dimux->items[dimux->num_items].label = label; dimux->items[dimux->num_items].index = index; dimux->num_items++; + if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) { + imux->items[imux->num_items].label = label; + imux->items[imux->num_items].index = index; + imux->num_items++; + } } return 0; } +static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t *fixed, hda_nid_t *ext) +{ + unsigned int cfg; + + if (!nid) + return 0; + cfg = snd_hda_codec_get_pincfg(codec, nid); + switch (get_defcfg_connect(cfg)) { + case AC_JACK_PORT_FIXED: + if (*fixed) + return 1; /* already occupied */ + *fixed = nid; + break; + case AC_JACK_PORT_COMPLEX: + if (*ext) + return 1; /* already occupied */ + *ext = nid; + break; + } + return 0; +} + +static int set_mic_route(struct hda_codec *codec, + struct sigmatel_mic_route *mic, + hda_nid_t pin) +{ + struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + mic->pin = pin; + for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) + if (pin == cfg->input_pins[i]) + break; + if (i <= AUTO_PIN_FRONT_MIC) { + /* analog pin */ + mic->dmux_idx = 0; + i = get_connection_index(codec, spec->mux_nids[0], pin); + if (i < 0) + return -1; + mic->mux_idx = i; + } else if (spec->dmux_nids) { + /* digital pin */ + mic->mux_idx = 0; + i = get_connection_index(codec, spec->dmux_nids[0], pin); + if (i < 0) + return -1; + mic->dmux_idx = i; + } + return 0; +} + +/* return non-zero if the device is for automatic mic switch */ +static int stac_check_auto_mic(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t fixed, ext; + int i; + + for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++) { + if (cfg->input_pins[i]) + return 0; /* must be exclusively mics */ + } + fixed = ext = 0; + for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) + if (check_mic_pin(codec, cfg->input_pins[i], &fixed, &ext)) + return 0; + for (i = 0; i < spec->num_dmics; i++) + if (check_mic_pin(codec, spec->dmic_nids[i], &fixed, &ext)) + return 0; + if (!fixed || !ext) + return 0; + if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP)) + return 0; /* no unsol support */ + if (set_mic_route(codec, &spec->ext_mic, ext) || + set_mic_route(codec, &spec->int_mic, fixed)) + return 0; /* something is wrong */ + return 1; +} + /* create playback/capture controls for input pins */ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *imux = &spec->private_imux; - hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int i, j, k; + int i, j; for (i = 0; i < AUTO_PIN_LAST; i++) { - int index; + hda_nid_t nid = cfg->input_pins[i]; + int index, err; - if (!cfg->input_pins[i]) + if (!nid) continue; index = -1; for (j = 0; j < spec->num_muxes; j++) { - int num_cons; - num_cons = snd_hda_get_connections(codec, - spec->mux_nids[j], - con_lst, - HDA_MAX_NUM_INPUTS); - for (k = 0; k < num_cons; k++) - if (con_lst[k] == cfg->input_pins[i]) { - index = k; - goto found; - } + index = get_connection_index(codec, spec->mux_nids[j], + nid); + if (index >= 0) + break; } - continue; - found: + if (index < 0) + continue; + + err = create_elem_capture_vol(codec, nid, + auto_pin_cfg_labels[i], + HDA_INPUT); + if (err < 0) + return err; + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = index; imux->num_items++; } + spec->num_analog_muxes = imux->num_items; if (imux->num_items) { /* @@ -3711,7 +3601,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out { struct sigmatel_spec *spec = codec->spec; int hp_swap = 0; - int err; + int i, err; if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -3751,11 +3641,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (snd_hda_get_connections(codec, spec->autocfg.mono_out_pin, conn_list, 1) && snd_hda_get_connections(codec, conn_list[0], - conn_list, 1)) { + conn_list, 1) > 0) { int wcaps = get_wcaps(codec, conn_list[0]); - int wid_type = (wcaps & AC_WCAP_TYPE) - >> AC_WCAP_TYPE_SHIFT; + int wid_type = get_wcaps_type(wcaps); /* LR swap check, some stac925x have a mux that * changes the DACs output path instead of the * mono-mux path. @@ -3846,6 +3735,21 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_outs = 0; } + if (stac_check_auto_mic(codec)) { + spec->auto_mic = 1; + /* only one capture for auto-mic */ + spec->num_adcs = 1; + spec->num_caps = 1; + spec->num_muxes = 1; + } + + for (i = 0; i < spec->num_caps; i++) { + err = stac92xx_add_capvol_ctls(codec, spec->capvols[i], + spec->capsws[i], i); + if (err < 0) + return err; + } + err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -3855,11 +3759,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - if (spec->num_amps > 0) { - err = stac92xx_auto_create_amp_output_ctls(codec); - if (err < 0) - return err; - } if (spec->num_dmics > 0 && !spec->dinput_mux) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) @@ -3896,7 +3795,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; - spec->amp_mux = &spec->private_amp_mux; return 1; } @@ -4108,14 +4006,14 @@ static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid, } static struct sigmatel_event *stac_get_event(struct hda_codec *codec, - hda_nid_t nid, unsigned char type) + hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; struct sigmatel_event *event = spec->events.list; int i; for (i = 0; i < spec->events.used; i++, event++) { - if (event->nid == nid && event->type == type) + if (event->nid == nid) return event; } return NULL; @@ -4135,24 +4033,32 @@ static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec, return NULL; } -static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int type) +/* check if given nid is a valid pin and no other events are assigned + * to it. If OK, assign the event, set the unsol flag, and returns 1. + * Otherwise, returns zero. + */ +static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, + unsigned int type) { struct sigmatel_event *event; int tag; if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) - return; - event = stac_get_event(codec, nid, type); - if (event) + return 0; + event = stac_get_event(codec, nid); + if (event) { + if (event->type != type) + return 0; tag = event->tag; - else + } else { tag = stac_add_event(codec->spec, nid, type, 0); - if (tag < 0) - return; + if (tag < 0) + return 0; + } snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | tag); + return 1; } static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) @@ -4245,20 +4151,36 @@ static int stac92xx_init(struct hda_codec *codec) hda_nid_t nid = cfg->hp_pins[i]; enable_pin_detect(codec, nid, STAC_HP_EVENT); } + if (cfg->line_out_type == AUTO_PIN_LINE_OUT && + cfg->speaker_outs > 0) { + /* enable pin-detect for line-outs as well */ + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + enable_pin_detect(codec, nid, STAC_LO_EVENT); + } + } + /* force to enable the first line-out; the others are set up * in unsol_event */ stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], AC_PINCTL_OUT_EN); /* fake event to set up pins */ - stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0], - STAC_HP_EVENT); + stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); for (i = 0; i < cfg->hp_outs; i++) stac_toggle_power_map(codec, cfg->hp_pins[i], 1); } + if (spec->auto_mic) { + /* initialize connection to analog input */ + if (spec->dmux_nids) + snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, + AC_VERB_SET_CONNECT_SEL, 0); + if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT)) + stac_issue_unsol_event(codec, spec->ext_mic.pin); + } for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { @@ -4285,10 +4207,9 @@ static int stac92xx_init(struct hda_codec *codec) } conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { - enable_pin_detect(codec, nid, - STAC_INSERT_EVENT); - stac_issue_unsol_event(codec, nid, - STAC_INSERT_EVENT); + if (enable_pin_detect(codec, nid, + STAC_INSERT_EVENT)) + stac_issue_unsol_event(codec, nid); } } } @@ -4333,10 +4254,8 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - if (!stac_get_event(codec, nid, STAC_INSERT_EVENT)) { - enable_pin_detect(codec, nid, STAC_PWR_EVENT); - stac_issue_unsol_event(codec, nid, STAC_PWR_EVENT); - } + if (enable_pin_detect(codec, nid, STAC_PWR_EVENT)) + stac_issue_unsol_event(codec, nid); } if (spec->dac_list) stac92xx_power_down(codec); @@ -4440,6 +4359,48 @@ static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) return 0; } +static void stac92xx_line_out_detect(struct hda_codec *codec, + int presence) +{ + struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->line_outs; i++) { + if (presence) + break; + presence = get_pin_presence(codec, cfg->line_out_pins[i]); + if (presence) { + unsigned int pinctl; + pinctl = snd_hda_codec_read(codec, + cfg->line_out_pins[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (pinctl & AC_PINCTL_IN_EN) + presence = 0; /* mic- or line-input */ + } + } + + if (presence) { + /* disable speakers */ + for (i = 0; i < cfg->speaker_outs; i++) + stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], + AC_PINCTL_OUT_EN); + if (spec->eapd_mask && spec->eapd_switch) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); + } else { + /* enable speakers */ + for (i = 0; i < cfg->speaker_outs; i++) + stac92xx_set_pinctl(codec, cfg->speaker_pins[i], + AC_PINCTL_OUT_EN); + if (spec->eapd_mask && spec->eapd_switch) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data | + spec->eapd_mask); + } +} + /* return non-zero if the hp-pin of the given array index isn't * a jack-detection target */ @@ -4492,13 +4453,6 @@ static void stac92xx_hp_detect(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->speaker_outs; i++) - stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], - AC_PINCTL_OUT_EN); - if (spec->eapd_mask && spec->eapd_switch) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); } else { /* enable lineouts */ if (spec->hp_switch) @@ -4507,14 +4461,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) stac92xx_set_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->speaker_outs; i++) - stac92xx_set_pinctl(codec, cfg->speaker_pins[i], - AC_PINCTL_OUT_EN); - if (spec->eapd_mask && spec->eapd_switch) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data | - spec->eapd_mask); } + stac92xx_line_out_detect(codec, presence); /* toggle hp outs */ for (i = 0; i < cfg->hp_outs; i++) { unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; @@ -4599,10 +4547,28 @@ static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) } } -static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid, - unsigned char type) +static void stac92xx_mic_detect(struct hda_codec *codec) { - struct sigmatel_event *event = stac_get_event(codec, nid, type); + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_mic_route *mic; + + if (get_pin_presence(codec, spec->ext_mic.pin)) + mic = &spec->ext_mic; + else + mic = &spec->int_mic; + if (mic->dmux_idx) + snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0, + AC_VERB_SET_CONNECT_SEL, + mic->dmux_idx); + else + snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0, + AC_VERB_SET_CONNECT_SEL, + mic->mux_idx); +} + +static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_event *event = stac_get_event(codec, nid); if (!event) return; codec->patch_ops.unsol_event(codec, (unsigned)event->tag << 26); @@ -4621,8 +4587,18 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) switch (event->type) { case STAC_HP_EVENT: + case STAC_LO_EVENT: stac92xx_hp_detect(codec); - /* fallthru */ + break; + case STAC_MIC_EVENT: + stac92xx_mic_detect(codec); + break; + } + + switch (event->type) { + case STAC_HP_EVENT: + case STAC_LO_EVENT: + case STAC_MIC_EVENT: case STAC_INSERT_EVENT: case STAC_PWR_EVENT: if (spec->num_pwrs > 0) @@ -4713,8 +4689,7 @@ static int stac92xx_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); /* fake event to set up pins again to override cached values */ if (spec->hp_detect) - stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0], - STAC_HP_EVENT); + stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]); return 0; } @@ -4754,6 +4729,19 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; + int i; + hda_nid_t nid; + + /* reset each pin before powering down DAC/ADC to avoid click noise */ + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wcaps); + if (wid_type == AC_WID_PIN) + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } + if (spec->eapd_mask) stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data & @@ -4790,7 +4778,8 @@ static int patch_stac9200(struct hda_codec *codec) stac9200_models, stac9200_cfg_tbl); if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac9200_brd_tbl[spec->board_config]); @@ -4862,8 +4851,8 @@ static int patch_stac925x(struct hda_codec *codec) stac925x_cfg_tbl); again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," - "using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac925x_brd_tbl[spec->board_config]); @@ -4893,6 +4882,9 @@ static int patch_stac925x(struct hda_codec *codec) spec->init = stac925x_core_init; spec->mixer = stac925x_mixer; + spec->num_caps = 1; + spec->capvols = stac925x_capvols; + spec->capsws = stac925x_capsws; err = stac92xx_parse_auto_config(codec, 0x8, 0x7); if (!err) { @@ -4914,16 +4906,6 @@ static int patch_stac925x(struct hda_codec *codec) return 0; } -static struct hda_input_mux stac92hd73xx_dmux = { - .num_items = 4, - .items = { - { "Analog Inputs", 0x0b }, - { "Digital Mic 1", 0x09 }, - { "Digital Mic 2", 0x0a }, - { "CD", 0x08 }, - } -}; - static int patch_stac92hd73xx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -4943,10 +4925,16 @@ static int patch_stac92hd73xx(struct hda_codec *codec) STAC_92HD73XX_MODELS, stac92hd73xx_models, stac92hd73xx_cfg_tbl); + /* check codec subsystem id if not found */ + if (spec->board_config < 0) + spec->board_config = + snd_hda_check_board_codec_sid_config(codec, + STAC_92HD73XX_MODELS, stac92hd73xx_models, + stac92hd73xx_codec_id_cfg_tbl); again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - " STAC92HD73XX, using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac92hd73xx_brd_tbl[spec->board_config]); @@ -4959,20 +4947,15 @@ again: "number of channels defaulting to DAC count\n"); num_dacs = STAC92HD73_DAC_COUNT; } + spec->init = stac92hd73xx_core_init; switch (num_dacs) { case 0x3: /* 6 Channel */ - spec->mixer = stac92hd73xx_6ch_mixer; - spec->init = stac92hd73xx_6ch_core_init; spec->aloopback_ctl = stac92hd73xx_6ch_loopback; break; case 0x4: /* 8 Channel */ - spec->mixer = stac92hd73xx_8ch_mixer; - spec->init = stac92hd73xx_8ch_core_init; spec->aloopback_ctl = stac92hd73xx_8ch_loopback; break; case 0x5: /* 10 Channel */ - spec->mixer = stac92hd73xx_10ch_mixer; - spec->init = stac92hd73xx_10ch_core_init; spec->aloopback_ctl = stac92hd73xx_10ch_loopback; break; } @@ -4987,14 +4970,14 @@ again: spec->dmic_nids = stac92hd73xx_dmic_nids; spec->dmux_nids = stac92hd73xx_dmux_nids; spec->smux_nids = stac92hd73xx_smux_nids; - spec->amp_nids = stac92hd73xx_amp_nids; - spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids); spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - memcpy(&spec->private_dimux, &stac92hd73xx_dmux, - sizeof(stac92hd73xx_dmux)); + + spec->num_caps = STAC92HD73XX_NUM_CAPS; + spec->capvols = stac92hd73xx_capvols; + spec->capsws = stac92hd73xx_capsws; switch (spec->board_config) { case STAC_DELL_EQ: @@ -5004,43 +4987,40 @@ again: case STAC_DELL_M6_DMIC: case STAC_DELL_M6_BOTH: spec->num_smuxes = 0; - spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; - spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; spec->eapd_switch = 0; - spec->num_amps = 1; - if (spec->board_config != STAC_DELL_EQ) - spec->init = dell_m6_core_init; switch (spec->board_config) { case STAC_DELL_M6_AMIC: /* Analog Mics */ snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; - spec->private_dimux.num_items = 1; break; case STAC_DELL_M6_DMIC: /* Digital Mics */ snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; - spec->private_dimux.num_items = 2; break; case STAC_DELL_M6_BOTH: /* Both */ snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; - spec->private_dimux.num_items = 2; break; } break; + case STAC_ALIENWARE_M17X: + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); + spec->eapd_switch = 0; + break; default: spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); spec->eapd_switch = 1; + break; } if (spec->board_config > STAC_92HD73XX_REF) { /* GPIO0 High = Enable EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; } - spec->dinput_mux = &spec->private_dimux; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; @@ -5072,15 +5052,6 @@ again: return 0; } -static struct hda_input_mux stac92hd83xxx_dmux = { - .num_items = 3, - .items = { - { "Analog Inputs", 0x03 }, - { "Digital Mic 1", 0x04 }, - { "Digital Mic 2", 0x05 }, - } -}; - static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5097,32 +5068,30 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; spec->mono_nid = 0x19; spec->digbeep_nid = 0x21; - spec->dmic_nids = stac92hd83xxx_dmic_nids; - spec->dmux_nids = stac92hd83xxx_dmux_nids; + spec->mux_nids = stac92hd83xxx_mux_nids; + spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids); spec->adc_nids = stac92hd83xxx_adc_nids; + spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); spec->pwr_nids = stac92hd83xxx_pwr_nids; - spec->amp_nids = stac92hd83xxx_amp_nids; spec->pwr_mapping = stac92hd83xxx_pwr_mapping; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; spec->init = stac92hd83xxx_core_init; - spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); - spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); - spec->num_amps = ARRAY_SIZE(stac92hd83xxx_amp_nids); - spec->num_dmics = STAC92HD83XXX_NUM_DMICS; - spec->dinput_mux = &stac92hd83xxx_dmux; spec->pin_nids = stac92hd83xxx_pin_nids; + spec->num_caps = STAC92HD83XXX_NUM_CAPS; + spec->capvols = stac92hd83xxx_capvols; + spec->capsws = stac92hd83xxx_capsws; + spec->board_config = snd_hda_check_board_config(codec, STAC_92HD83XXX_MODELS, stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - " STAC92HD83XXX, using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac92hd83xxx_brd_tbl[spec->board_config]); @@ -5164,6 +5133,8 @@ again: num_dacs = snd_hda_get_connections(codec, nid, conn, STAC92HD83_DAC_COUNT + 1) - 1; + if (num_dacs < 0) + num_dacs = STAC92HD83_DAC_COUNT; /* set port X to select the last DAC */ @@ -5177,25 +5148,6 @@ again: return 0; } -static struct hda_input_mux stac92hd71bxx_dmux_nomixer = { - .num_items = 3, - .items = { - { "Analog Inputs", 0x00 }, - { "Digital Mic 1", 0x02 }, - { "Digital Mic 2", 0x03 }, - } -}; - -static struct hda_input_mux stac92hd71bxx_dmux_amixer = { - .num_items = 4, - .items = { - { "Analog Inputs", 0x00 }, - { "Mixer", 0x01 }, - { "Digital Mic 1", 0x02 }, - { "Digital Mic 2", 0x03 }, - } -}; - /* get the pin connection (fixed, none, etc) */ static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) { @@ -5256,7 +5208,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) struct sigmatel_spec *spec; struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; - unsigned int ndmic_nids = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5285,8 +5236,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) stac92hd71bxx_cfg_tbl); again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - " STAC92HD71BXX, using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac92hd71bxx_brd_tbl[spec->board_config]); @@ -5301,6 +5252,10 @@ again: spec->dmic_nids = stac92hd71bxx_dmic_nids; spec->dmux_nids = stac92hd71bxx_dmux_nids; + spec->num_caps = STAC92HD71BXX_NUM_CAPS; + spec->capvols = stac92hd71bxx_capvols; + spec->capsws = stac92hd71bxx_capsws; + switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: @@ -5308,24 +5263,13 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, - sizeof(stac92hd71bxx_dmux_nomixer)); - spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92hd71bxx_connected_ports(codec, stac92hd71bxx_dmic_nids, STAC92HD71BXX_NUM_DMICS); - if (spec->num_dmics) { - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->dinput_mux = &spec->private_dimux; - ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; - } break; case 0x111d7608: /* 5 Port with Analog Mixer */ - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, - sizeof(stac92hd71bxx_dmux_amixer)); - spec->private_dimux.num_items--; switch (spec->board_config) { case STAC_HP_M4: /* Enable VREF power saving on GPIO1 detect */ @@ -5347,11 +5291,8 @@ again: /* no output amps */ spec->num_pwrs = 0; - spec->mixer = stac92hd71bxx_analog_mixer; - spec->dinput_mux = &spec->private_dimux; - /* disable VSW */ - spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; + spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5359,8 +5300,6 @@ again: spec->num_dmics = stac92hd71bxx_connected_ports(codec, stac92hd71bxx_dmic_nids, STAC92HD71BXX_NUM_DMICS - 1); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2; break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) @@ -5370,17 +5309,12 @@ again: spec->num_pwrs = 0; /* fallthru */ default: - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, - sizeof(stac92hd71bxx_dmux_amixer)); - spec->dinput_mux = &spec->private_dimux; - spec->mixer = stac92hd71bxx_analog_mixer; - spec->init = stac92hd71bxx_analog_core_init; + spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92hd71bxx_connected_ports(codec, stac92hd71bxx_dmic_nids, STAC92HD71BXX_NUM_DMICS); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; + break; } if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) @@ -5408,6 +5342,7 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); switch (spec->board_config) { @@ -5462,8 +5397,6 @@ again: #endif spec->multiout.dac_nids = spec->dac_nids; - if (spec->dinput_mux) - spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; err = stac92xx_parse_auto_config(codec, 0x21, 0); if (!err) { @@ -5541,8 +5474,8 @@ static int patch_stac922x(struct hda_codec *codec) again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " - "using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac922x_brd_tbl[spec->board_config]); @@ -5555,7 +5488,10 @@ static int patch_stac922x(struct hda_codec *codec) spec->num_pwrs = 0; spec->init = stac922x_core_init; - spec->mixer = stac922x_mixer; + + spec->num_caps = STAC922X_NUM_CAPS; + spec->capvols = stac922x_capvols; + spec->capsws = stac922x_capsws; spec->multiout.dac_nids = spec->dac_nids; @@ -5604,8 +5540,8 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_cfg_tbl); again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - "STAC927x, using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac927x_brd_tbl[spec->board_config]); @@ -5630,7 +5566,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_dmics = 0; spec->init = d965_core_init; - spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: switch (codec->subsystem_id) { @@ -5662,7 +5597,6 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_dmics = STAC927X_NUM_DMICS; spec->init = d965_core_init; - spec->mixer = stac927x_mixer; spec->dmux_nids = stac927x_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; @@ -5675,9 +5609,12 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_dmics = 0; spec->init = stac927x_core_init; - spec->mixer = stac927x_mixer; } + spec->num_caps = STAC927X_NUM_CAPS; + spec->capvols = stac927x_capvols; + spec->capsws = stac927x_capsws; + spec->num_pwrs = 0; spec->aloopback_ctl = stac927x_loopback; spec->aloopback_mask = 0x40; @@ -5739,7 +5676,8 @@ static int patch_stac9205(struct hda_codec *codec) stac9205_cfg_tbl); again: if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac9205_brd_tbl[spec->board_config]); @@ -5758,9 +5696,12 @@ static int patch_stac9205(struct hda_codec *codec) spec->num_pwrs = 0; spec->init = stac9205_core_init; - spec->mixer = stac9205_mixer; spec->aloopback_ctl = stac9205_loopback; + spec->num_caps = STAC9205_NUM_CAPS; + spec->capvols = stac9205_capvols; + spec->capsws = stac9205_capsws; + spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; /* Turn on/off EAPD per HP plugging */ @@ -5835,12 +5776,6 @@ static struct hda_verb stac9872_core_init[] = { {} }; -static struct snd_kcontrol_new stac9872_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - { } /* end */ -}; - static hda_nid_t stac9872_pin_nids[] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x11, 0x13, 0x14, @@ -5854,6 +5789,11 @@ static hda_nid_t stac9872_mux_nids[] = { 0x15 }; +static unsigned long stac9872_capvols[] = { + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), +}; +#define stac9872_capsws stac9872_capvols + static unsigned int stac9872_vaio_pin_configs[9] = { 0x03211020, 0x411111f0, 0x411111f0, 0x03a15030, 0x411111f0, 0x90170110, 0x411111f0, 0x411111f0, @@ -5891,8 +5831,8 @@ static int patch_stac9872(struct hda_codec *codec) stac9872_models, stac9872_cfg_tbl); if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, " - "using BIOS defaults\n"); + snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", + codec->chip_name); else stac92xx_set_config_regs(codec, stac9872_brd_tbl[spec->board_config]); @@ -5902,8 +5842,10 @@ static int patch_stac9872(struct hda_codec *codec) spec->adc_nids = stac9872_adc_nids; spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); spec->mux_nids = stac9872_mux_nids; - spec->mixer = stac9872_mixer; spec->init = stac9872_core_init; + spec->num_caps = 1; + spec->capvols = stac9872_capvols; + spec->capsws = stac9872_capsws; err = stac92xx_parse_auto_config(codec, 0x10, 0x12); if (err < 0) { diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e8f10b10cceb..ee89db90c9b6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1339,8 +1339,7 @@ static int get_mux_nids(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) { nid = spec->adc_nids[i]; while (nid) { - type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) - >> AC_WCAP_TYPE_SHIFT; + type = get_wcaps_type(get_wcaps(codec, nid)); if (type == AC_WID_PIN) break; n = snd_hda_get_connections(codec, nid, conn, diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index adc909ec125c..9da2dae64c5b 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -379,6 +379,15 @@ struct snd_ice1712 { unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate); void (*set_spdif_clock)(struct snd_ice1712 *ice); +#ifdef CONFIG_PM + int (*pm_suspend)(struct snd_ice1712 *); + int (*pm_resume)(struct snd_ice1712 *); + int pm_suspend_enabled:1; + int pm_saved_is_spdif_master:1; + unsigned int pm_saved_spdif_ctrl; + unsigned char pm_saved_spdif_cfg; + unsigned int pm_saved_route; +#endif }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index cc84a831eb21..af6e00148621 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -560,6 +560,7 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&ice->reg_lock); old = inb(ICEMT1724(ice, DMA_CONTROL)); if (cmd == SNDRV_PCM_TRIGGER_START) @@ -570,6 +571,10 @@ static int snd_vt1724_pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_unlock(&ice->reg_lock); break; + case SNDRV_PCM_TRIGGER_RESUME: + /* apps will have to restart stream */ + break; + default: return -EINVAL; } @@ -2262,7 +2267,7 @@ static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, -static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) +static void snd_vt1724_chip_reset(struct snd_ice1712 *ice) { outb(VT1724_RESET , ICEREG1724(ice, CONTROL)); inb(ICEREG1724(ice, CONTROL)); /* pci posting flush */ @@ -2272,7 +2277,7 @@ static void __devinit snd_vt1724_chip_reset(struct snd_ice1712 *ice) msleep(10); } -static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) +static int snd_vt1724_chip_init(struct snd_ice1712 *ice) { outb(ice->eeprom.data[ICE_EEP2_SYSCONF], ICEREG1724(ice, SYS_CFG)); outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); @@ -2287,6 +2292,14 @@ static int __devinit snd_vt1724_chip_init(struct snd_ice1712 *ice) outb(0, ICEREG1724(ice, POWERDOWN)); + /* MPU_RX and TX irq masks are cleared later dynamically */ + outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK)); + + /* don't handle FIFO overrun/underruns (just yet), + * since they cause machine lockups + */ + outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); + return 0; } @@ -2431,6 +2444,8 @@ static int __devinit snd_vt1724_create(struct snd_card *card, snd_vt1724_proc_init(ice); synchronize_irq(pci->irq); + card->private_data = ice; + err = pci_request_regions(pci, "ICE1724"); if (err < 0) { kfree(ice); @@ -2459,14 +2474,6 @@ static int __devinit snd_vt1724_create(struct snd_card *card, return -EIO; } - /* MPU_RX and TX irq masks are cleared later dynamically */ - outb(VT1724_IRQ_MPU_RX | VT1724_IRQ_MPU_TX , ICEREG1724(ice, IRQMASK)); - - /* don't handle FIFO overrun/underruns (just yet), - * since they cause machine lockups - */ - outb(VT1724_MULTI_FIFO_ERR, ICEMT1724(ice, DMA_INT_MASK)); - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); if (err < 0) { snd_vt1724_free(ice); @@ -2650,11 +2657,96 @@ static void __devexit snd_vt1724_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } +#ifdef CONFIG_PM +static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(ice->pcm); + snd_pcm_suspend_all(ice->pcm_pro); + snd_pcm_suspend_all(ice->pcm_ds); + snd_ac97_suspend(ice->ac97); + + spin_lock_irq(&ice->reg_lock); + ice->pm_saved_is_spdif_master = ice->is_spdif_master(ice); + ice->pm_saved_spdif_ctrl = inw(ICEMT1724(ice, SPDIF_CTRL)); + ice->pm_saved_spdif_cfg = inb(ICEREG1724(ice, SPDIF_CFG)); + ice->pm_saved_route = inl(ICEMT1724(ice, ROUTE_PLAYBACK)); + spin_unlock_irq(&ice->reg_lock); + + if (ice->pm_suspend) + ice->pm_suspend(ice); + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int snd_vt1724_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + pci_set_master(pci); + + snd_vt1724_chip_reset(ice); + + if (snd_vt1724_chip_init(ice) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + if (ice->pm_resume) + ice->pm_resume(ice); + + if (ice->pm_saved_is_spdif_master) { + /* switching to external clock via SPDIF */ + ice->set_spdif_clock(ice); + } else { + /* internal on-card clock */ + snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); + } + + update_spdif_bits(ice, ice->pm_saved_spdif_ctrl); + + outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG)); + outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK)); + + if (ice->ac97) + snd_ac97_resume(ice->ac97); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + static struct pci_driver driver = { .name = "ICE1724", .id_table = snd_vt1724_ids, .probe = snd_vt1724_probe, .remove = __devexit_p(snd_vt1724_remove), +#ifdef CONFIG_PM + .suspend = snd_vt1724_suspend, + .resume = snd_vt1724_resume, +#endif }; static int __init alsa_card_ice1724_init(void) diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 043a93879bd5..c75515f5be6f 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1077,7 +1077,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) /* * initialize the chip */ -static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) +static void ak4396_init(struct snd_ice1712 *ice) { static unsigned short ak4396_inits[] = { AK4396_CTRL1, 0x87, /* I2S Normal Mode, 24 bit */ @@ -1087,9 +1087,37 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) AK4396_RCH_ATT, 0x00, }; - struct prodigy_hifi_spec *spec; unsigned int i; + /* initialize ak4396 codec */ + /* reset codec */ + ak4396_write(ice, AK4396_CTRL1, 0x86); + msleep(100); + ak4396_write(ice, AK4396_CTRL1, 0x87); + + for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2) + ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); +} + +#ifdef CONFIG_PM +static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice) +{ + /* initialize ak4396 codec and restore previous mixer volumes */ + struct prodigy_hifi_spec *spec = ice->spec; + int i; + mutex_lock(&ice->gpio_mutex); + ak4396_init(ice); + for (i = 0; i < 2; i++) + ak4396_write(ice, AK4396_LCH_ATT + i, spec->vol[i] & 0xff); + mutex_unlock(&ice->gpio_mutex); + return 0; +} +#endif + +static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) +{ + struct prodigy_hifi_spec *spec; + ice->vt1720 = 0; ice->vt1724 = 1; @@ -1112,14 +1140,12 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) return -ENOMEM; ice->spec = spec; - /* initialize ak4396 codec */ - /* reset codec */ - ak4396_write(ice, AK4396_CTRL1, 0x86); - msleep(100); - ak4396_write(ice, AK4396_CTRL1, 0x87); - - for (i = 0; i < ARRAY_SIZE(ak4396_inits); i += 2) - ak4396_write(ice, ak4396_inits[i], ak4396_inits[i+1]); +#ifdef CONFIG_PM + ice->pm_resume = &prodigy_hd2_resume; + ice->pm_suspend_enabled = 1; +#endif + + ak4396_init(ice); return 0; } diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index c1eb923f2ac9..09b2b2a36df5 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -215,17 +215,8 @@ EXPORT_SYMBOL(oxygen_write_spi); void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data) { - unsigned long timeout; - /* should not need more than about 300 us */ - timeout = jiffies + msecs_to_jiffies(1); - do { - if (!(oxygen_read16(chip, OXYGEN_2WIRE_BUS_STATUS) - & OXYGEN_2WIRE_BUSY)) - break; - udelay(1); - cond_resched(); - } while (time_after_eq(timeout, jiffies)); + msleep(1); oxygen_write8(chip, OXYGEN_2WIRE_MAP, map); oxygen_write8(chip, OXYGEN_2WIRE_DATA, data); diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 3da5c029f93b..7bb827c7d806 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3294,15 +3294,33 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) char *clock_source; int x; - if (hdsp_check_for_iobox (hdsp)) { - snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n"); + status = hdsp_read(hdsp, HDSP_statusRegister); + status2 = hdsp_read(hdsp, HDSP_status2Register); + + snd_iprintf(buffer, "%s (Card #%d)\n", hdsp->card_name, + hdsp->card->number + 1); + snd_iprintf(buffer, "Buffers: capture %p playback %p\n", + hdsp->capture_buffer, hdsp->playback_buffer); + snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", + hdsp->irq, hdsp->port, (unsigned long)hdsp->iobase); + snd_iprintf(buffer, "Control register: 0x%x\n", hdsp->control_register); + snd_iprintf(buffer, "Control2 register: 0x%x\n", + hdsp->control2_register); + snd_iprintf(buffer, "Status register: 0x%x\n", status); + snd_iprintf(buffer, "Status2 register: 0x%x\n", status2); + + if (hdsp_check_for_iobox(hdsp)) { + snd_iprintf(buffer, "No I/O box connected.\n" + "Please connect one and upload firmware.\n"); return; - } + } if (hdsp_check_for_firmware(hdsp, 0)) { if (hdsp->state & HDSP_FirmwareCached) { if (snd_hdsp_load_firmware_from_cache(hdsp) != 0) { - snd_iprintf(buffer, "Firmware loading from cache failed, please upload manually.\n"); + snd_iprintf(buffer, "Firmware loading from " + "cache failed, " + "please upload manually.\n"); return; } } else { @@ -3319,18 +3337,6 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } } - status = hdsp_read(hdsp, HDSP_statusRegister); - status2 = hdsp_read(hdsp, HDSP_status2Register); - - snd_iprintf(buffer, "%s (Card #%d)\n", hdsp->card_name, hdsp->card->number + 1); - snd_iprintf(buffer, "Buffers: capture %p playback %p\n", - hdsp->capture_buffer, hdsp->playback_buffer); - snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", - hdsp->irq, hdsp->port, (unsigned long)hdsp->iobase); - snd_iprintf(buffer, "Control register: 0x%x\n", hdsp->control_register); - snd_iprintf(buffer, "Control2 register: 0x%x\n", hdsp->control2_register); - snd_iprintf(buffer, "Status register: 0x%x\n", status); - snd_iprintf(buffer, "Status2 register: 0x%x\n", status2); snd_iprintf(buffer, "FIFO status: %d\n", hdsp_read(hdsp, HDSP_fifoStatus) & 0xff); snd_iprintf(buffer, "MIDI1 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut0)); snd_iprintf(buffer, "MIDI1 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn0)); @@ -3351,7 +3357,6 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "\n"); - switch (hdsp_clock_source(hdsp)) { case HDSP_CLOCK_SOURCE_AUTOSYNC: clock_source = "AutoSync"; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 2f0925236a1b..5518371db13f 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -834,7 +834,7 @@ static irqreturn_t snd_ymfpci_interrupt(int irq, void *dev_id) status = snd_ymfpci_readw(chip, YDSXGR_INTFLAG); if (status & 1) { if (chip->timer) - snd_timer_interrupt(chip->timer, chip->timer->sticks); + snd_timer_interrupt(chip->timer, chip->timer_ticks); } snd_ymfpci_writew(chip, YDSXGR_INTFLAG, status); @@ -1885,8 +1885,18 @@ static int snd_ymfpci_timer_start(struct snd_timer *timer) unsigned int count; chip = snd_timer_chip(timer); - count = (timer->sticks << 1) - 1; spin_lock_irqsave(&chip->reg_lock, flags); + if (timer->sticks > 1) { + chip->timer_ticks = timer->sticks; + count = timer->sticks - 1; + } else { + /* + * Divisor 1 is not allowed; fake it by using divisor 2 and + * counting two ticks for each interrupt. + */ + chip->timer_ticks = 2; + count = 2 - 1; + } snd_ymfpci_writew(chip, YDSXGR_TIMERCOUNT, count); snd_ymfpci_writeb(chip, YDSXGR_TIMERCTRL, 0x03); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -1909,14 +1919,14 @@ static int snd_ymfpci_timer_precise_resolution(struct snd_timer *timer, unsigned long *num, unsigned long *den) { *num = 1; - *den = 48000; + *den = 96000; return 0; } static struct snd_timer_hardware snd_ymfpci_timer_hw = { .flags = SNDRV_TIMER_HW_AUTO, - .resolution = 20833, /* 1/fs = 20.8333...us */ - .ticks = 0x8000, + .resolution = 10417, /* 1 / 96 kHz = 10.41666...us */ + .ticks = 0x10000, .start = snd_ymfpci_timer_start, .stop = snd_ymfpci_timer_stop, .precise_resolution = snd_ymfpci_timer_precise_resolution, diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index d3e786a9a0a7..b1749bc67979 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -29,6 +29,7 @@ source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/fsl/Kconfig" +source "sound/soc/imx/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 6f1e28de23cf..0c5eac01bf2e 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ @@ -7,6 +7,7 @@ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += fsl/ +obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 173a239a541c..130b12118d4f 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -56,30 +56,14 @@ #define MCLK_RATE 12000000 -static struct clk *mclk; - -static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - MCLK_RATE, SND_SOC_CLOCK_IN); - if (ret < 0) { - clk_disable(mclk); - return ret; - } - - return 0; -} - -static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); +/* + * As shipped the board does not have inputs. However, it is relatively + * straightforward to modify the board to hook them up so support is left + * in the driver. + */ +#undef ENABLE_MIC_INPUT - dev_dbg(rtd->socdev->dev, "shutdown"); -} +static struct clk *mclk; static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -87,102 +71,17 @@ static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct atmel_ssc_info *ssc_p = cpu_dai->private_data; - struct ssc_device *ssc = ssc_p->ssc; int ret; - unsigned int rate; - int cmr_div, period; - - if (ssc == NULL) { - printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n"); - return -EINVAL; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* - * The SSC clock dividers depend on the sample rate. The CMR.DIV - * field divides the system master clock MCK to drive the SSC TK - * signal which provides the codec BCLK. The TCMR.PERIOD and - * RCMR.PERIOD fields further divide the BCLK signal to drive - * the SSC TF and RF signals which provide the codec DACLRC and - * ADCLRC clocks. - * - * The dividers were determined through trial and error, where a - * CMR.DIV value is chosen such that the resulting BCLK value is - * divisible, or almost divisible, by (2 * sample rate), and then - * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1. - */ - rate = params_rate(params); - - switch (rate) { - case 8000: - cmr_div = 55; /* BCLK = 133MHz/(2*55) = 1.209MHz */ - period = 74; /* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */ - break; - case 11025: - cmr_div = 67; /* BCLK = 133MHz/(2*60) = 1.108MHz */ - period = 45; /* LRC = BCLK/(2*(49+1)) = 11083,3Hz */ - break; - case 16000: - cmr_div = 63; /* BCLK = 133MHz/(2*63) = 1.055MHz */ - period = 32; /* LRC = BCLK/(2*(32+1)) = 15993,2Hz */ - break; - case 22050: - cmr_div = 52; /* BCLK = 133MHz/(2*52) = 1.278MHz */ - period = 28; /* LRC = BCLK/(2*(28+1)) = 22049Hz */ - break; - case 32000: - cmr_div = 66; /* BCLK = 133MHz/(2*66) = 1.007MHz */ - period = 15; /* LRC = BCLK/(2*(15+1)) = 31486,742Hz */ - break; - case 44100: - cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ - period = 25; /* LRC = BCLK/(2*(25+1)) = 44098Hz */ - break; - case 48000: - cmr_div = 33; /* BCLK = 133MHz/(2*33) = 2.015MHz */ - period = 20; /* LRC = BCLK/(2*(20+1)) = 47979,79Hz */ - break; - case 88200: - cmr_div = 29; /* BCLK = 133MHz/(2*29) = 2.293MHz */ - period = 12; /* LRC = BCLK/(2*(12+1)) = 88196Hz */ - break; - case 96000: - cmr_div = 23; /* BCLK = 133MHz/(2*23) = 2.891MHz */ - period = 14; /* LRC = BCLK/(2*(14+1)) = 96376Hz */ - break; - default: - printk(KERN_WARNING "unsupported rate %d" - " on at91sam9g20ek board\n", rate); - return -EINVAL; - } - - /* set the MCK divider for BCLK */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div); - if (ret < 0) - return ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* set the BCLK divider for DACLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - ATMEL_SSC_TCMR_PERIOD, period); - } else { - /* set the BCLK divider for ADCLRC */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - ATMEL_SSC_RCMR_PERIOD, period); - } + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; @@ -190,9 +89,7 @@ static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops at91sam9g20ek_ops = { - .startup = at91sam9g20ek_startup, .hw_params = at91sam9g20ek_hw_params, - .shutdown = at91sam9g20ek_shutdown, }; static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, @@ -241,10 +138,20 @@ static const struct snd_soc_dapm_route intercon[] = { */ static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) { + struct snd_soc_dai *codec_dai = &codec->dai[0]; + int ret; + printk(KERN_DEBUG "at91sam9g20ek_wm8731 " ": at91sam9g20ek_wm8731_init() called\n"); + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + MCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret); + return ret; + } + /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets, ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); @@ -255,8 +162,13 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_nc_pin(codec, "LLINEIN"); - /* always connected */ +#ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); +#else + snd_soc_dapm_nc_pin(codec, "Int Mic"); +#endif + + /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bdf1865..a521aa90ddee 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a04a07b..3f474e8ed4f6 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -29,6 +29,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; + struct mutex lock; }; #define PCM_TX 0 diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 811596f4c092..ac927ffdc961 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -7,6 +7,15 @@ config SND_BF5XX_I2S mode (supports single stereo In/Out). You will also need to select the audio interfaces to support below. +config SND_BF5XX_TDM + tristate "SoC I2S(TDM mode) Audio for the ADI BF5xx chip" + depends on (BLACKFIN && SND_SOC) + help + Say Y or M if you want to add support for codecs attached to + the Blackfin SPORT (synchronous serial ports) interface in TDM + mode. + You will also need to select the audio interfaces to support below. + config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio support for BF52x ezkit" depends on SND_BF5XX_I2S @@ -69,12 +78,24 @@ config SND_BF5XX_SOC_I2S tristate select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_TDM + tristate + select SND_BF5XX_SOC_SPORT + config SND_BF5XX_SOC_AC97 tristate select AC97_BUS select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 @@ -83,9 +104,17 @@ config SND_BF5XX_SOC_AD1980 help Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. +config SND_BF5XX_SOC_AD1938 + tristate "SoC AD1938 Audio support for Blackfin" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1938 + help + Say Y if you want to add support for AD1938 codec on Blackfin. + config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97) + depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 97bb37a6359c..87e30423912f 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -1,21 +1,29 @@ # Blackfin Platform Support snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o +snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o +obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o +snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1938) += snd-ad1938.o diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index b1ed423fabd5..2758b9017a7f 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -277,28 +277,24 @@ static int bf5xx_ac97_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; - ret = sport_set_multichannel(sport_handle, 16, 0x1F, 1); + ret = sport_set_multichannel(sport, 16, 0x1F, 1); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); + ret = sport_config_rx(sport, IRFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); + ret = sport_config_tx(sport, ITFS, 0xF, 0, (16*16-1)); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - if (dai->capture.active) - sport_rx_start(sport); - if (dai->playback.active) - sport_tx_start(sport); return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 000000000000..cd361e304b0f --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,128 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c new file mode 100644 index 000000000000..08269e91810c --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -0,0 +1,142 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1938.c + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: Thur June 4 2009 + * Description: Board driver for ad1938 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1938.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1938; + +static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI slots, 8 channels, all channels are enabled */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1938_ops = { + .startup = bf5xx_ad1938_startup, + .hw_params = bf5xx_ad1938_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1938_dai = { + .name = "ad1938", + .stream_name = "AD1938", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1938_dai, + .ops = &bf5xx_ad1938_ops, +}; + +static struct snd_soc_card bf5xx_ad1938 = { + .name = "bf5xx_ad1938", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1938_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1938_snd_devdata = { + .card = &bf5xx_ad1938, + .codec_dev = &soc_codec_dev_ad1938, +}; + +static struct platform_device *bfxx_ad1938_snd_device; + +static int __init bf5xx_ad1938_init(void) +{ + int ret; + + bfxx_ad1938_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1938_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1938_snd_device, &bf5xx_ad1938_snd_devdata); + bf5xx_ad1938_snd_devdata.dev = &bfxx_ad1938_snd_device->dev; + ret = platform_device_add(bfxx_ad1938_snd_device); + + if (ret) + platform_device_put(bfxx_ad1938_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1938_exit(void) +{ + platform_device_unregister(bfxx_ad1938_snd_device); +} + +module_init(bf5xx_ad1938_init); +module_exit(bf5xx_ad1938_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1938 board driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index edfbdc024e66..9825b71d0e28 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -203,23 +203,23 @@ static struct snd_soc_device bf5xx_ad73311_snd_devdata = { .codec_dev = &soc_codec_dev_ad73311, }; -static struct platform_device *bf52x_ad73311_snd_device; +static struct platform_device *bf5xx_ad73311_snd_device; static int __init bf5xx_ad73311_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ad73311_snd_device) + bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ad73311_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); - bf5xx_ad73311_snd_devdata.dev = &bf52x_ad73311_snd_device->dev; - ret = platform_device_add(bf52x_ad73311_snd_device); + platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311_snd_devdata); + bf5xx_ad73311_snd_devdata.dev = &bf5xx_ad73311_snd_device->dev; + ret = platform_device_add(bf5xx_ad73311_snd_device); if (ret) - platform_device_put(bf52x_ad73311_snd_device); + platform_device_put(bf5xx_ad73311_snd_device); return ret; } @@ -227,7 +227,7 @@ static int __init bf5xx_ad73311_init(void) static void __exit bf5xx_ad73311_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ad73311_snd_device); + platform_device_unregister(bf5xx_ad73311_snd_device); } module_init(bf5xx_ad73311_init); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index af06904bab0f..876abade27e1 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -259,22 +259,18 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) if (!dai->active) return 0; - ret = sport_config_rx(sport_handle, RFSR | RCKFE, RSFSE|0x1f, 0, 0); + ret = sport_config_rx(sport, RFSR | RCKFE, RSFSE|0x1f, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - ret = sport_config_tx(sport_handle, TFSR | TCKFE, TSFSE|0x1f, 0, 0); + ret = sport_config_tx(sport, TFSR | TCKFE, TSFSE|0x1f, 0, 0); if (ret) { pr_err("SPORT is busy!\n"); return -EBUSY; } - if (dai->capture.active) - sport_rx_start(sport); - if (dai->playback.active) - sport_tx_start(sport); return 0; } diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index bc0cdded7116..3a00fa4dbe6d 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -148,24 +148,24 @@ static struct snd_soc_device bf5xx_ssm2602_snd_devdata = { .codec_data = &bf5xx_ssm2602_setup, }; -static struct platform_device *bf52x_ssm2602_snd_device; +static struct platform_device *bf5xx_ssm2602_snd_device; static int __init bf5xx_ssm2602_init(void) { int ret; pr_debug("%s enter\n", __func__); - bf52x_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf52x_ssm2602_snd_device) + bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); + if (!bf5xx_ssm2602_snd_device) return -ENOMEM; - platform_set_drvdata(bf52x_ssm2602_snd_device, + platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602_snd_devdata); - bf5xx_ssm2602_snd_devdata.dev = &bf52x_ssm2602_snd_device->dev; - ret = platform_device_add(bf52x_ssm2602_snd_device); + bf5xx_ssm2602_snd_devdata.dev = &bf5xx_ssm2602_snd_device->dev; + ret = platform_device_add(bf5xx_ssm2602_snd_device); if (ret) - platform_device_put(bf52x_ssm2602_snd_device); + platform_device_put(bf5xx_ssm2602_snd_device); return ret; } @@ -173,7 +173,7 @@ static int __init bf5xx_ssm2602_init(void) static void __exit bf5xx_ssm2602_exit(void) { pr_debug("%s enter\n", __func__); - platform_device_unregister(bf52x_ssm2602_snd_device); + platform_device_unregister(bf5xx_ssm2602_snd_device); } module_init(bf5xx_ssm2602_init); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c new file mode 100644 index 000000000000..ccb5e823bd18 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -0,0 +1,330 @@ +/* + * File: sound/soc/blackfin/bf5xx-tdm-pcm.c + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: Tue June 06 2009 + * Description: DMA driver for tdm codec + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" +#include "bf5xx-sport.h" + +#define PCM_BUFFER_MAX 0x10000 +#define FRAGMENT_SIZE_MIN (4*1024) +#define FRAGMENTS_MIN 2 +#define FRAGMENTS_MAX 32 + +static void bf5xx_dma_irq(void *data) +{ + struct snd_pcm_substream *pcm = data; + snd_pcm_period_elapsed(pcm); +} + +static const struct snd_pcm_hardware bf5xx_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_RESUME), + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = PCM_BUFFER_MAX, + .period_bytes_min = FRAGMENT_SIZE_MIN, + .period_bytes_max = PCM_BUFFER_MAX/2, + .periods_min = FRAGMENTS_MIN, + .periods_max = FRAGMENTS_MAX, +}; + +static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; + snd_pcm_lib_malloc_pages(substream, size * 4); + + return 0; +} + +static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int fragsize_bytes = frames_to_bytes(runtime, runtime->period_size); + + fragsize_bytes /= runtime->channels; + /* inflate the fragsize to match the dma width of SPORT */ + fragsize_bytes *= 8; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + sport_set_tx_callback(sport, bf5xx_dma_irq, substream); + sport_config_tx_dma(sport, runtime->dma_area, + runtime->periods, fragsize_bytes); + } else { + sport_set_rx_callback(sport, bf5xx_dma_irq, substream); + sport_config_rx_dma(sport, runtime->dma_area, + runtime->periods, fragsize_bytes); + } + + return 0; +} + +static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_start(sport); + else + sport_rx_start(sport); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_tx_stop(sport); + else + sport_rx_stop(sport); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + unsigned int diff; + snd_pcm_uframes_t frames; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + diff = sport_curr_offset_tx(sport); + frames = diff / (8*4); /* 32 bytes per frame */ + } else { + diff = sport_curr_offset_rx(sport); + frames = diff / (8*4); + } + return frames; +} + +static int bf5xx_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + goto out; + + if (sport_handle != NULL) + runtime->private_data = sport_handle; + else { + pr_err("sport_handle is NULL\n"); + ret = -ENODEV; + } +out: + return ret; +} + +static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) +{ + unsigned int *src; + unsigned int *dst; + int i; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = buf; + dst = (unsigned int *)substream->runtime->dma_area; + + dst += pos * 8; + while (count--) { + for (i = 0; i < substream->runtime->channels; i++) + *(dst + i) = *src++; + dst += 8; + } + } else { + src = (unsigned int *)substream->runtime->dma_area; + dst = buf; + + src += pos * 8; + while (count--) { + for (i = 0; i < substream->runtime->channels; i++) + *dst++ = *(src+i); + src += 8; + } + } + + return 0; +} + +static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, snd_pcm_uframes_t count) +{ + unsigned char *buf = substream->runtime->dma_area; + buf += pos * 8 * 4; + memset(buf, '\0', count * 8 * 4); + + return 0; +} + + +struct snd_pcm_ops bf5xx_pcm_tdm_ops = { + .open = bf5xx_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = bf5xx_pcm_hw_params, + .hw_free = bf5xx_pcm_hw_free, + .prepare = bf5xx_pcm_prepare, + .trigger = bf5xx_pcm_trigger, + .pointer = bf5xx_pcm_pointer, + .copy = bf5xx_pcm_copy, + .silence = bf5xx_pcm_silence, +}; + +static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = bf5xx_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size * 4, + &buf->addr, GFP_KERNEL); + if (!buf->area) { + pr_err("Failed to allocate dma memory \ + Please increase uncached DMA memory region\n"); + return -ENOMEM; + } + buf->bytes = size; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + sport_handle->tx_buf = buf->area; + else + sport_handle->rx_buf = buf->area; + + return 0; +} + +static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(NULL, buf->bytes, buf->area, 0); + buf->area = NULL; + } + if (sport_handle) + sport_done(sport_handle); +} + +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); + +static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &bf5xx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->playback.channels_min) { + ret = bf5xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = bf5xx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } +out: + return ret; +} + +struct snd_soc_platform bf5xx_tdm_soc_platform = { + .name = "bf5xx-audio", + .pcm_ops = &bf5xx_pcm_tdm_ops, + .pcm_new = bf5xx_pcm_tdm_new, + .pcm_free = bf5xx_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(bf5xx_tdm_soc_platform); + +static int __init bfin_pcm_tdm_init(void) +{ + return snd_soc_register_platform(&bf5xx_tdm_soc_platform); +} +module_init(bfin_pcm_tdm_init); + +static void __exit bfin_pcm_tdm_exit(void) +{ + snd_soc_unregister_platform(&bf5xx_tdm_soc_platform); +} +module_exit(bfin_pcm_tdm_exit); + +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.h b/sound/soc/blackfin/bf5xx-tdm-pcm.h new file mode 100644 index 000000000000..ddc5047df88c --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.h @@ -0,0 +1,21 @@ +/* + * sound/soc/blackfin/bf5xx-tdm-pcm.h -- ALSA PCM interface for the Blackfin + * + * Copyright 2009 Analog Device Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _BF5XX_TDM_PCM_H +#define _BF5XX_TDM_PCM_H + +struct bf5xx_pcm_dma_params { + char *name; /* stream identifier */ +}; + +/* platform data */ +extern struct snd_soc_platform bf5xx_tdm_soc_platform; + +#endif diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c new file mode 100644 index 000000000000..3096badf09a5 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -0,0 +1,343 @@ +/* + * File: sound/soc/blackfin/bf5xx-tdm.c + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: Thurs June 04 2009 + * Description: Blackfin I2S(TDM) CPU DAI driver + * Even though TDM mode can be as part of I2S DAI, but there + * are so much difference in configuration and data flow, + * it's very ugly to integrate I2S and TDM into a module + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <asm/irq.h> +#include <asm/portmux.h> +#include <linux/mutex.h> +#include <linux/gpio.h> + +#include "bf5xx-sport.h" +#include "bf5xx-tdm.h" + +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + int configured; +}; + +static struct bf5xx_tdm_port bf5xx_tdm; +static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; + +static struct sport_param sport_params[2] = { + { + .dma_rx_chan = CH_SPORT0_RX, + .dma_tx_chan = CH_SPORT0_TX, + .err_irq = IRQ_SPORT0_ERROR, + .regs = (struct sport_register *)SPORT0_TCR1, + }, + { + .dma_rx_chan = CH_SPORT1_RX, + .dma_tx_chan = CH_SPORT1_TX, + .err_irq = IRQ_SPORT1_ERROR, + .regs = (struct sport_register *)SPORT1_TCR1, + } +}; + +/* + * Setting the TFS pin selector for SPORT 0 based on whether the selected + * port id F or G. If the port is F then no conflict should exist for the + * TFS. When Port G is selected and EMAC then there is a conflict between + * the PHY interrupt line and TFS. Current settings prevent the conflict + * by ignoring the TFS pin when Port G is selected. This allows both + * ssm2602 using Port G and EMAC concurrently. + */ +#ifdef CONFIG_BF527_SPORT0_PORTF +#define LOCAL_SPORT0_TFS (P_SPORT0_TFS) +#else +#define LOCAL_SPORT0_TFS (0) +#endif + +static u16 sport_req[][7] = { {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, + P_SPORT0_DRPRI, P_SPORT0_RSCLK, LOCAL_SPORT0_TFS, 0}, + {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, P_SPORT1_DRPRI, + P_SPORT1_RSCLK, P_SPORT1_TFS, 0} }; + +static int bf5xx_tdm_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + int ret = 0; + + /* interface format:support TDM,slave mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + break; + default: + printk(KERN_ERR "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + printk(KERN_ERR "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bf5xx_tdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int ret = 0; + + bf5xx_tdm.tcr2 &= ~0x1f; + bf5xx_tdm.rcr2 &= ~0x1f; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + bf5xx_tdm.tcr2 |= 31; + bf5xx_tdm.rcr2 |= 31; + sport_handle->wdsize = 4; + break; + /* at present, we only support 32bit transfer */ + default: + pr_err("not supported PCM format yet\n"); + return -EINVAL; + break; + } + + if (!bf5xx_tdm.configured) { + /* + * TX and RX are not independent,they are enabled at the + * same time, even if only one side is running. So, we + * need to configure both of them at the time when the first + * stream is opened. + * + * CPU DAI:slave mode. + */ + ret = sport_config_rx(sport_handle, bf5xx_tdm.rcr1, + bf5xx_tdm.rcr2, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + return -EBUSY; + } + + ret = sport_config_tx(sport_handle, bf5xx_tdm.tcr1, + bf5xx_tdm.tcr2, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + return -EBUSY; + } + + bf5xx_tdm.configured = 1; + } + + return 0; +} + +static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + /* No active stream, SPORT is allowed to be configured again. */ + if (!dai->active) + bf5xx_tdm.configured = 0; +} + +#ifdef CONFIG_PM +static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = + (struct sport_device *)dai->private_data; + + if (!dai->active) + return 0; + if (dai->capture.active) + sport_rx_stop(sport); + if (dai->playback.active) + sport_tx_stop(sport); + return 0; +} + +static int bf5xx_tdm_resume(struct snd_soc_dai *dai) +{ + int ret; + struct sport_device *sport = + (struct sport_device *)dai->private_data; + + if (!dai->active) + return 0; + + ret = sport_set_multichannel(sport, 8, 0xFF, 1); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + ret = sport_config_rx(sport, IRFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + ret = sport_config_tx(sport, ITFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + } + + return 0; +} + +#else +#define bf5xx_tdm_suspend NULL +#define bf5xx_tdm_resume NULL +#endif + +static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { + .hw_params = bf5xx_tdm_hw_params, + .set_fmt = bf5xx_tdm_set_dai_fmt, + .shutdown = bf5xx_tdm_shutdown, +}; + +struct snd_soc_dai bf5xx_tdm_dai = { + .name = "bf5xx-tdm", + .id = 0, + .suspend = bf5xx_tdm_suspend, + .resume = bf5xx_tdm_resume, + .playback = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .ops = &bf5xx_tdm_dai_ops, +}; +EXPORT_SYMBOL_GPL(bf5xx_tdm_dai); + +static int __devinit bfin_tdm_probe(struct platform_device *pdev) +{ + int ret = 0; + + if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { + pr_err("Requesting Peripherals failed\n"); + return -EFAULT; + } + + /* request DMA for SPORT */ + sport_handle = sport_init(&sport_params[sport_num], 4, \ + 8 * sizeof(u32), NULL); + if (!sport_handle) { + peripheral_free_list(&sport_req[sport_num][0]); + return -ENODEV; + } + + /* SPORT works in TDM mode */ + ret = sport_set_multichannel(sport_handle, 8, 0xFF, 1); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = sport_config_rx(sport_handle, IRFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = sport_config_tx(sport_handle, ITFS, 0x1F, 0, 0); + if (ret) { + pr_err("SPORT is busy!\n"); + ret = -EBUSY; + goto sport_config_err; + } + + ret = snd_soc_register_dai(&bf5xx_tdm_dai); + if (ret) { + pr_err("Failed to register DAI: %d\n", ret); + goto sport_config_err; + } + return 0; + +sport_config_err: + peripheral_free_list(&sport_req[sport_num][0]); + return ret; +} + +static int __devexit bfin_tdm_remove(struct platform_device *pdev) +{ + peripheral_free_list(&sport_req[sport_num][0]); + snd_soc_unregister_dai(&bf5xx_tdm_dai); + + return 0; +} + +static struct platform_driver bfin_tdm_driver = { + .probe = bfin_tdm_probe, + .remove = __devexit_p(bfin_tdm_remove), + .driver = { + .name = "bfin-tdm", + .owner = THIS_MODULE, + }, +}; + +static int __init bfin_tdm_init(void) +{ + return platform_driver_register(&bfin_tdm_driver); +} +module_init(bfin_tdm_init); + +static void __exit bfin_tdm_exit(void) +{ + platform_driver_unregister(&bfin_tdm_driver); +} +module_exit(bfin_tdm_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("TDM driver for ADI Blackfin"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h new file mode 100644 index 000000000000..618ec3d90cd4 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -0,0 +1,14 @@ +/* + * sound/soc/blackfin/bf5xx-tdm.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _BF5XX_TDM_H +#define _BF5XX_TDM_H + +extern struct snd_soc_dai bf5xx_tdm_dai; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbc97fd76648..0edca93af3b0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,11 +12,15 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AD1836 if SPI_MASTER + select SND_SOC_AD1938 if SPI_MASTER select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4642 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -30,18 +34,23 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8523 if I2C select SND_SOC_WM8580 if I2C select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8960 if I2C + select SND_SOC_WM8961 if I2C select SND_SOC_WM8971 if I2C + select SND_SOC_WM8974 if I2C select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C + select SND_SOC_WM8993 if I2C select SND_SOC_WM9081 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS @@ -57,11 +66,21 @@ config SND_SOC_ALL_CODECS If unsure select "N". +config SND_SOC_WM_HUBS + tristate + default y if SND_SOC_WM8993=y + default m if SND_SOC_WM8993=m config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AD1836 + tristate + +config SND_SOC_AD1938 + tristate + config SND_SOC_AD1980 tristate @@ -74,6 +93,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4642 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate @@ -86,6 +108,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_CX20442 + tristate + config SND_SOC_L3 tristate @@ -129,6 +154,9 @@ config SND_SOC_WM8400 config SND_SOC_WM8510 tristate +config SND_SOC_WM8523 + tristate + config SND_SOC_WM8580 tristate @@ -144,6 +172,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8776 + tristate + config SND_SOC_WM8900 tristate @@ -156,15 +187,24 @@ config SND_SOC_WM8940 config SND_SOC_WM8960 tristate +config SND_SOC_WM8961 + tristate + config SND_SOC_WM8971 tristate +config SND_SOC_WM8974 + tristate + config SND_SOC_WM8988 tristate config SND_SOC_WM8990 tristate +config SND_SOC_WM8993 + tristate + config SND_SOC_WM9081 tristate @@ -176,3 +216,7 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate + +# Amp +config SND_SOC_MAX9877 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8b7530546f4d..fb4af28486ba 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,9 +1,13 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ad1836-objs := ad1836.o +snd-soc-ad1938-objs := ad1938.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4642-objs := ak4642.o snd-soc-cs4270-objs := cs4270.o +snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-spdif-objs := spdif_transciever.o @@ -18,29 +22,42 @@ snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o +snd-soc-wm8523-objs := wm8523.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8776-objs := wm8776.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8940-objs := wm8940.o snd-soc-wm8960-objs := wm8960.o +snd-soc-wm8961-objs := wm8961.o snd-soc-wm8971-objs := wm8971.o +snd-soc-wm8974-objs := wm8974.o snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm8993-objs := wm8993.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +snd-soc-wm-hubs-objs := wm_hubs.o + +# Amp +snd-soc-max9877-objs := max9877.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o +obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o @@ -55,19 +72,28 @@ obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o +obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o +obj-$(CONFIG_SND_SOC_WM8961) += snd-soc-wm8961.o obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o +obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o + +# Amp +obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c new file mode 100644 index 000000000000..3612bb92df90 --- /dev/null +++ b/sound/soc/codecs/ad1836.c @@ -0,0 +1,446 @@ +/* + * File: sound/soc/codecs/ad1836.c + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: Aug 04 2009 + * Description: Driver for AD1836 sound chip + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-dapm.h> +#include <linux/spi/spi.h> +#include "ad1836.h" + +/* codec private data */ +struct ad1836_priv { + struct snd_soc_codec codec; + u16 reg_cache[AD1836_NUM_REGS]; +}; + +static struct snd_soc_codec *ad1836_codec; +struct snd_soc_codec_device soc_codec_dev_ad1836; +static int ad1836_register(struct ad1836_priv *ad1836); +static void ad1836_unregister(struct ad1836_priv *ad1836); + +/* + * AD1836 volume/mute/de-emphasis etc. controls + */ +static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; + +static const struct soc_enum ad1836_deemp_enum = + SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); + +static const struct snd_kcontrol_new ad1836_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL, + AD1836_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL, + AD1836_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL, + AD1836_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE, + AD1836_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE, + AD1836_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE, + AD1836_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE, + AD1836_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE, + AD1836_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1, + AD1836_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1, + AD1836_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1, + AD1836_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + /* at present, we support adc aux mode to interface with + * blackfin sport tdm mode + */ + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + /* ALCLK,ABCLK are both output, AD1836 can only be master */ + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ad1836_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, + AD1836_DAC_WORD_LEN_MASK, word_len); + + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + AD1836_ADC_WORD_LEN_MASK, word_len); + + return 0; +} + + +/* + * interface to read/write ad1836 register + */ +#define AD1836_SPI_REG_SHFT 12 +#define AD1836_SPI_READ (1 << 11) +#define AD1836_SPI_VAL_MSK 0x3FF + +/* + * write to the ad1836 register space + */ + +static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *reg_cache = codec->reg_cache; + int ret = 0; + + if (value != reg_cache[reg]) { + unsigned short buf; + struct spi_transfer t = { + .tx_buf = &buf, + .len = 2, + }; + struct spi_message m; + + buf = (reg << AD1836_SPI_REG_SHFT) | + (value & AD1836_SPI_VAL_MSK); + spi_message_init(&m); + spi_message_add_tail(&t, &m); + ret = spi_sync(codec->control_data, &m); + if (ret == 0) + reg_cache[reg] = value; + } + + return ret; +} + +/* + * read from the ad1836 register space cache + */ +static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +static int __devinit ad1836_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ad1836_priv *ad1836; + + ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL); + if (ad1836 == NULL) + return -ENOMEM; + + codec = &ad1836->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, ad1836); + + return ad1836_register(ad1836); +} + +static int __devexit ad1836_spi_remove(struct spi_device *spi) +{ + struct ad1836_priv *ad1836 = dev_get_drvdata(&spi->dev); + + ad1836_unregister(ad1836); + return 0; +} + +static struct spi_driver ad1836_spi_driver = { + .driver = { + .name = "ad1836-spi", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = ad1836_spi_probe, + .remove = __devexit_p(ad1836_spi_remove), +}; + +static struct snd_soc_dai_ops ad1836_dai_ops = { + .hw_params = ad1836_hw_params, + .set_fmt = ad1836_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad1836_dai = { + .name = "AD1836", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad1836_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad1836_dai); + +static int ad1836_register(struct ad1836_priv *ad1836) +{ + int ret; + struct snd_soc_codec *codec = &ad1836->codec; + + if (ad1836_codec) { + dev_err(codec->dev, "Another ad1836 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->private_data = ad1836; + codec->reg_cache = ad1836->reg_cache; + codec->reg_cache_size = AD1836_NUM_REGS; + codec->name = "AD1836"; + codec->owner = THIS_MODULE; + codec->dai = &ad1836_dai; + codec->num_dai = 1; + codec->write = ad1836_write_reg; + codec->read = ad1836_read_reg_cache; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ad1836_dai.dev = codec->dev; + ad1836_codec = codec; + + /* default setting for ad1836 */ + /* de-emphasis: 48kHz, power-on dac */ + codec->write(codec, AD1836_DAC_CTRL1, 0x300); + /* unmute dac channels */ + codec->write(codec, AD1836_DAC_CTRL2, 0x0); + /* high-pass filter enable, power-on adc */ + codec->write(codec, AD1836_ADC_CTRL1, 0x100); + /* unmute adc channles, adc aux mode */ + codec->write(codec, AD1836_ADC_CTRL2, 0x180); + /* left/right diff:PGA/MUX */ + codec->write(codec, AD1836_ADC_CTRL3, 0x3A); + /* volume */ + codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad1836); + return ret; + } + + ret = snd_soc_register_dai(&ad1836_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + kfree(ad1836); + return ret; + } + + return 0; +} + +static void ad1836_unregister(struct ad1836_priv *ad1836) +{ + snd_soc_unregister_dai(&ad1836_dai); + snd_soc_unregister_codec(&ad1836->codec); + kfree(ad1836); + ad1836_codec = NULL; +} + +static int ad1836_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ad1836_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ad1836_codec; + codec = ad1836_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ad1836_snd_controls, + ARRAY_SIZE(ad1836_snd_controls)); + snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, + ARRAY_SIZE(ad1836_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int ad1836_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad1836 = { + .probe = ad1836_probe, + .remove = ad1836_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); + +static int __init ad1836_init(void) +{ + int ret; + + ret = spi_register_driver(&ad1836_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", + ret); + } + + return ret; +} +module_init(ad1836_init); + +static void __exit ad1836_exit(void) +{ + spi_unregister_driver(&ad1836_spi_driver); +} +module_exit(ad1836_exit); + +MODULE_DESCRIPTION("ASoC ad1836 driver"); +MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h new file mode 100644 index 000000000000..7660ee6973c0 --- /dev/null +++ b/sound/soc/codecs/ad1836.h @@ -0,0 +1,64 @@ +/* + * File: sound/soc/codecs/ad1836.h + * Based on: + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: Aug 04, 2009 + * Description: definitions for AD1836 registers + * + * Modified: + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef __AD1836_H__ +#define __AD1836_H__ + +#define AD1836_DAC_CTRL1 0 +#define AD1836_DAC_POWERDOWN 2 +#define AD1836_DAC_SERFMT_MASK 0xE0 +#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) +#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) +#define AD1836_DAC_WORD_LEN_MASK 0x18 + +#define AD1836_DAC_CTRL2 1 +#define AD1836_DACL1_MUTE 0 +#define AD1836_DACR1_MUTE 1 +#define AD1836_DACL2_MUTE 2 +#define AD1836_DACR2_MUTE 3 +#define AD1836_DACL3_MUTE 4 +#define AD1836_DACR3_MUTE 5 + +#define AD1836_DAC_L1_VOL 2 +#define AD1836_DAC_R1_VOL 3 +#define AD1836_DAC_L2_VOL 4 +#define AD1836_DAC_R2_VOL 5 +#define AD1836_DAC_L3_VOL 6 +#define AD1836_DAC_R3_VOL 7 + +#define AD1836_ADC_CTRL1 12 +#define AD1836_ADC_POWERDOWN 7 +#define AD1836_ADC_HIGHPASS_FILTER 8 + +#define AD1836_ADC_CTRL2 13 +#define AD1836_ADCL1_MUTE 0 +#define AD1836_ADCR1_MUTE 1 +#define AD1836_ADCL2_MUTE 2 +#define AD1836_ADCR2_MUTE 3 +#define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_SERFMT_MASK (7 << 6) +#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) +#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) + +#define AD1836_ADC_CTRL3 14 + +#define AD1836_NUM_REGS 16 + +extern struct snd_soc_dai ad1836_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad1836; +#endif diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c new file mode 100644 index 000000000000..e62b27701a49 --- /dev/null +++ b/sound/soc/codecs/ad1938.c @@ -0,0 +1,682 @@ +/* + * File: sound/soc/codecs/ad1938.c + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: June 04 2009 + * Description: Driver for AD1938 sound chip + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-dapm.h> +#include <linux/spi/spi.h> +#include "ad1938.h" + +/* codec private data */ +struct ad1938_priv { + struct snd_soc_codec codec; + u8 reg_cache[AD1938_NUM_REGS]; +}; + +static struct snd_soc_codec *ad1938_codec; +struct snd_soc_codec_device soc_codec_dev_ad1938; +static int ad1938_register(struct ad1938_priv *ad1938); +static void ad1938_unregister(struct ad1938_priv *ad1938); + +/* + * AD1938 volume/mute/de-emphasis etc. controls + */ +static const char *ad1938_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; + +static const struct soc_enum ad1938_deemp_enum = + SOC_ENUM_SINGLE(AD1938_DAC_CTRL2, 1, 4, ad1938_deemp); + +static const struct snd_kcontrol_new ad1938_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD1938_DAC_L1_VOL, + AD1938_DAC_R1_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R("DAC2 Volume", AD1938_DAC_L2_VOL, + AD1938_DAC_R2_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R("DAC3 Volume", AD1938_DAC_L3_VOL, + AD1938_DAC_R3_VOL, 0, 0xFF, 1), + SOC_DOUBLE_R("DAC4 Volume", AD1938_DAC_L4_VOL, + AD1938_DAC_R4_VOL, 0, 0xFF, 1), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD1938_ADC_CTRL0, AD1938_ADCL1_MUTE, + AD1938_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1938_ADC_CTRL0, AD1938_ADCL2_MUTE, + AD1938_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL1_MUTE, + AD1938_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL2_MUTE, + AD1938_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL3_MUTE, + AD1938_DACR3_MUTE, 1, 1), + SOC_DOUBLE("DAC4 Switch", AD1938_DAC_CHNL_MUTE, AD1938_DACL4_MUTE, + AD1938_DACR4_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD1938_ADC_CTRL0, + AD1938_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad1938_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC4OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad1938_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg; + + reg = codec->read(codec, AD1938_DAC_CTRL2); + reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg & + (~AD1938_DAC_MASTER_MUTE); + codec->write(codec, AD1938_DAC_CTRL2, reg); + + return 0; +} + +static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd) +{ + int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0); + reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg | + AD1938_PLL_POWERDOWN; + codec->write(codec, AD1938_PLL_CLK_CTRL0, reg); + + return 0; +} + +static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int mask, int slots, int width) +{ + struct snd_soc_codec *codec = dai->codec; + int dac_reg = codec->read(codec, AD1938_DAC_CTRL1); + int adc_reg = codec->read(codec, AD1938_ADC_CTRL2); + + dac_reg &= ~AD1938_DAC_CHAN_MASK; + adc_reg &= ~AD1938_ADC_CHAN_MASK; + + switch (slots) { + case 2: + dac_reg |= AD1938_DAC_2_CHANNELS << AD1938_DAC_CHAN_SHFT; + adc_reg |= AD1938_ADC_2_CHANNELS << AD1938_ADC_CHAN_SHFT; + break; + case 4: + dac_reg |= AD1938_DAC_4_CHANNELS << AD1938_DAC_CHAN_SHFT; + adc_reg |= AD1938_ADC_4_CHANNELS << AD1938_ADC_CHAN_SHFT; + break; + case 8: + dac_reg |= AD1938_DAC_8_CHANNELS << AD1938_DAC_CHAN_SHFT; + adc_reg |= AD1938_ADC_8_CHANNELS << AD1938_ADC_CHAN_SHFT; + break; + case 16: + dac_reg |= AD1938_DAC_16_CHANNELS << AD1938_DAC_CHAN_SHFT; + adc_reg |= AD1938_ADC_16_CHANNELS << AD1938_ADC_CHAN_SHFT; + break; + default: + return -EINVAL; + } + + codec->write(codec, AD1938_DAC_CTRL1, dac_reg); + codec->write(codec, AD1938_ADC_CTRL2, adc_reg); + + return 0; +} + +static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int adc_reg, dac_reg; + + adc_reg = codec->read(codec, AD1938_ADC_CTRL2); + dac_reg = codec->read(codec, AD1938_DAC_CTRL1); + + /* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S + * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A) + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + adc_reg &= ~AD1938_ADC_SERFMT_MASK; + adc_reg |= AD1938_ADC_SERFMT_TDM; + break; + case SND_SOC_DAIFMT_DSP_A: + adc_reg &= ~AD1938_ADC_SERFMT_MASK; + adc_reg |= AD1938_ADC_SERFMT_AUX; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ + adc_reg &= ~AD1938_ADC_LEFT_HIGH; + adc_reg &= ~AD1938_ADC_BCLK_INV; + dac_reg &= ~AD1938_DAC_LEFT_HIGH; + dac_reg &= ~AD1938_DAC_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: /* normal bclk + invert frm */ + adc_reg |= AD1938_ADC_LEFT_HIGH; + adc_reg &= ~AD1938_ADC_BCLK_INV; + dac_reg |= AD1938_DAC_LEFT_HIGH; + dac_reg &= ~AD1938_DAC_BCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: /* invert bclk + normal frm */ + adc_reg &= ~AD1938_ADC_LEFT_HIGH; + adc_reg |= AD1938_ADC_BCLK_INV; + dac_reg &= ~AD1938_DAC_LEFT_HIGH; + dac_reg |= AD1938_DAC_BCLK_INV; + break; + + case SND_SOC_DAIFMT_IB_IF: /* invert bclk + frm */ + adc_reg |= AD1938_ADC_LEFT_HIGH; + adc_reg |= AD1938_ADC_BCLK_INV; + dac_reg |= AD1938_DAC_LEFT_HIGH; + dac_reg |= AD1938_DAC_BCLK_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ + adc_reg |= AD1938_ADC_LCR_MASTER; + adc_reg |= AD1938_ADC_BCLK_MASTER; + dac_reg |= AD1938_DAC_LCR_MASTER; + dac_reg |= AD1938_DAC_BCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ + adc_reg |= AD1938_ADC_LCR_MASTER; + adc_reg &= ~AD1938_ADC_BCLK_MASTER; + dac_reg |= AD1938_DAC_LCR_MASTER; + dac_reg &= ~AD1938_DAC_BCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + adc_reg &= ~AD1938_ADC_LCR_MASTER; + adc_reg |= AD1938_ADC_BCLK_MASTER; + dac_reg &= ~AD1938_DAC_LCR_MASTER; + dac_reg |= AD1938_DAC_BCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ + adc_reg &= ~AD1938_ADC_LCR_MASTER; + adc_reg &= ~AD1938_ADC_BCLK_MASTER; + dac_reg &= ~AD1938_DAC_LCR_MASTER; + dac_reg &= ~AD1938_DAC_BCLK_MASTER; + break; + default: + return -EINVAL; + } + + codec->write(codec, AD1938_ADC_CTRL2, adc_reg); + codec->write(codec, AD1938_DAC_CTRL1, dac_reg); + + return 0; +} + +static int ad1938_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0, reg = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + reg = codec->read(codec, AD1938_DAC_CTRL2); + reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len; + codec->write(codec, AD1938_DAC_CTRL2, reg); + + reg = codec->read(codec, AD1938_ADC_CTRL1); + reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len; + codec->write(codec, AD1938_ADC_CTRL1, reg); + + return 0; +} + +static int ad1938_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + ad1938_pll_powerctrl(codec, 1); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + ad1938_pll_powerctrl(codec, 0); + break; + } + codec->bias_level = level; + return 0; +} + +/* + * interface to read/write ad1938 register + */ + +#define AD1938_SPI_ADDR 0x4 +#define AD1938_SPI_READ 0x1 +#define AD1938_SPI_BUFLEN 3 + +/* + * write to the ad1938 register space + */ + +static int ad1938_write_reg(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *reg_cache = codec->reg_cache; + int ret = 0; + + if (value != reg_cache[reg]) { + uint8_t buf[AD1938_SPI_BUFLEN]; + struct spi_transfer t = { + .tx_buf = buf, + .len = AD1938_SPI_BUFLEN, + }; + struct spi_message m; + + buf[0] = AD1938_SPI_ADDR << 1; + buf[1] = reg; + buf[2] = value; + spi_message_init(&m); + spi_message_add_tail(&t, &m); + ret = spi_sync(codec->control_data, &m); + if (ret == 0) + reg_cache[reg] = value; + } + + return ret; +} + +/* + * read from the ad1938 register space cache + */ + +static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +/* + * read from the ad1938 register space + */ + +static unsigned int ad1938_read_reg(struct snd_soc_codec *codec, + unsigned int reg) +{ + char w_buf[AD1938_SPI_BUFLEN]; + char r_buf[AD1938_SPI_BUFLEN]; + int ret; + + struct spi_transfer t = { + .tx_buf = w_buf, + .rx_buf = r_buf, + .len = AD1938_SPI_BUFLEN, + }; + struct spi_message m; + + w_buf[0] = (AD1938_SPI_ADDR << 1) | AD1938_SPI_READ; + w_buf[1] = reg; + w_buf[2] = 0; + + spi_message_init(&m); + spi_message_add_tail(&t, &m); + ret = spi_sync(codec->control_data, &m); + if (ret == 0) + return r_buf[2]; + else + return -EIO; +} + +static int ad1938_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *reg_cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + for (i = 0; i < codec->reg_cache_size; i++) { + int ret = ad1938_read_reg(codec, i); + if (ret == -EIO) { + dev_err(&spi->dev, "AD1938 SPI read failure\n"); + return ret; + } + reg_cache[i] = ret; + } + + return 0; +} + +static int __devinit ad1938_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ad1938_priv *ad1938; + + ad1938 = kzalloc(sizeof(struct ad1938_priv), GFP_KERNEL); + if (ad1938 == NULL) + return -ENOMEM; + + codec = &ad1938->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, ad1938); + + return ad1938_register(ad1938); +} + +static int __devexit ad1938_spi_remove(struct spi_device *spi) +{ + struct ad1938_priv *ad1938 = dev_get_drvdata(&spi->dev); + + ad1938_unregister(ad1938); + return 0; +} + +static struct spi_driver ad1938_spi_driver = { + .driver = { + .name = "ad1938", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = ad1938_spi_probe, + .remove = __devexit_p(ad1938_spi_remove), +}; + +static struct snd_soc_dai_ops ad1938_dai_ops = { + .hw_params = ad1938_hw_params, + .digital_mute = ad1938_mute, + .set_tdm_slot = ad1938_set_tdm_slot, + .set_fmt = ad1938_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad1938_dai = { + .name = "AD1938", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad1938_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad1938_dai); + +static int ad1938_register(struct ad1938_priv *ad1938) +{ + int ret; + struct snd_soc_codec *codec = &ad1938->codec; + + if (ad1938_codec) { + dev_err(codec->dev, "Another ad1938 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->private_data = ad1938; + codec->reg_cache = ad1938->reg_cache; + codec->reg_cache_size = AD1938_NUM_REGS; + codec->name = "AD1938"; + codec->owner = THIS_MODULE; + codec->dai = &ad1938_dai; + codec->num_dai = 1; + codec->write = ad1938_write_reg; + codec->read = ad1938_read_reg_cache; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ad1938_dai.dev = codec->dev; + ad1938_codec = codec; + + /* default setting for ad1938 */ + + /* unmute dac channels */ + codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0); + /* de-emphasis: 48kHz, powedown dac */ + codec->write(codec, AD1938_DAC_CTRL2, 0x1A); + /* powerdown dac, dac in tdm mode */ + codec->write(codec, AD1938_DAC_CTRL0, 0x41); + /* high-pass filter enable */ + codec->write(codec, AD1938_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + codec->write(codec, AD1938_ADC_CTRL1, 0x43); + /* pll input: mclki/xi */ + codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D); + codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04); + + ad1938_fill_cache(codec); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad1938); + return ret; + } + + ret = snd_soc_register_dai(&ad1938_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + kfree(ad1938); + return ret; + } + + return 0; +} + +static void ad1938_unregister(struct ad1938_priv *ad1938) +{ + ad1938_set_bias_level(&ad1938->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ad1938_dai); + snd_soc_unregister_codec(&ad1938->codec); + kfree(ad1938); + ad1938_codec = NULL; +} + +static int ad1938_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ad1938_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ad1938_codec; + codec = ad1938_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ad1938_snd_controls, + ARRAY_SIZE(ad1938_snd_controls)); + snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets, + ARRAY_SIZE(ad1938_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int ad1938_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM +static int ad1938_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ad1938_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int ad1938_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + ad1938_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} +#else +#define ad1938_suspend NULL +#define ad1938_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_ad1938 = { + .probe = ad1938_probe, + .remove = ad1938_remove, + .suspend = ad1938_suspend, + .resume = ad1938_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938); + +static int __init ad1938_init(void) +{ + int ret; + + ret = spi_register_driver(&ad1938_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ad1938 SPI driver: %d\n", + ret); + } + + return ret; +} +module_init(ad1938_init); + +static void __exit ad1938_exit(void) +{ + spi_unregister_driver(&ad1938_spi_driver); +} +module_exit(ad1938_exit); + +MODULE_DESCRIPTION("ASoC ad1938 driver"); +MODULE_AUTHOR("Barry Song "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1938.h b/sound/soc/codecs/ad1938.h new file mode 100644 index 000000000000..fe3c48cd2d5b --- /dev/null +++ b/sound/soc/codecs/ad1938.h @@ -0,0 +1,100 @@ +/* + * File: sound/soc/codecs/ad1836.h + * Based on: + * Author: Barry Song <Barry.Song@analog.com> + * + * Created: May 25, 2009 + * Description: definitions for AD1938 registers + * + * Modified: + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, see the file COPYING, or write + * to the Free Software Foundation, Inc., + * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef __AD1938_H__ +#define __AD1938_H__ + +#define AD1938_PLL_CLK_CTRL0 0 +#define AD1938_PLL_POWERDOWN 0x01 +#define AD1938_PLL_CLK_CTRL1 1 +#define AD1938_DAC_CTRL0 2 +#define AD1938_DAC_POWERDOWN 0x01 +#define AD1938_DAC_SERFMT_MASK 0xC0 +#define AD1938_DAC_SERFMT_STEREO (0 << 6) +#define AD1938_DAC_SERFMT_TDM (1 << 6) +#define AD1938_DAC_CTRL1 3 +#define AD1938_DAC_2_CHANNELS 0 +#define AD1938_DAC_4_CHANNELS 1 +#define AD1938_DAC_8_CHANNELS 2 +#define AD1938_DAC_16_CHANNELS 3 +#define AD1938_DAC_CHAN_SHFT 1 +#define AD1938_DAC_CHAN_MASK (3 << AD1938_DAC_CHAN_SHFT) +#define AD1938_DAC_LCR_MASTER (1 << 4) +#define AD1938_DAC_BCLK_MASTER (1 << 5) +#define AD1938_DAC_LEFT_HIGH (1 << 3) +#define AD1938_DAC_BCLK_INV (1 << 7) +#define AD1938_DAC_CTRL2 4 +#define AD1938_DAC_WORD_LEN_MASK 0xC +#define AD1938_DAC_MASTER_MUTE 1 +#define AD1938_DAC_CHNL_MUTE 5 +#define AD1938_DACL1_MUTE 0 +#define AD1938_DACR1_MUTE 1 +#define AD1938_DACL2_MUTE 2 +#define AD1938_DACR2_MUTE 3 +#define AD1938_DACL3_MUTE 4 +#define AD1938_DACR3_MUTE 5 +#define AD1938_DACL4_MUTE 6 +#define AD1938_DACR4_MUTE 7 +#define AD1938_DAC_L1_VOL 6 +#define AD1938_DAC_R1_VOL 7 +#define AD1938_DAC_L2_VOL 8 +#define AD1938_DAC_R2_VOL 9 +#define AD1938_DAC_L3_VOL 10 +#define AD1938_DAC_R3_VOL 11 +#define AD1938_DAC_L4_VOL 12 +#define AD1938_DAC_R4_VOL 13 +#define AD1938_ADC_CTRL0 14 +#define AD1938_ADC_POWERDOWN 0x01 +#define AD1938_ADC_HIGHPASS_FILTER 1 +#define AD1938_ADCL1_MUTE 2 +#define AD1938_ADCR1_MUTE 3 +#define AD1938_ADCL2_MUTE 4 +#define AD1938_ADCR2_MUTE 5 +#define AD1938_ADC_CTRL1 15 +#define AD1938_ADC_SERFMT_MASK 0x60 +#define AD1938_ADC_SERFMT_STEREO (0 << 5) +#define AD1938_ADC_SERFMT_TDM (1 << 2) +#define AD1938_ADC_SERFMT_AUX (2 << 5) +#define AD1938_ADC_WORD_LEN_MASK 0x3 +#define AD1938_ADC_CTRL2 16 +#define AD1938_ADC_2_CHANNELS 0 +#define AD1938_ADC_4_CHANNELS 1 +#define AD1938_ADC_8_CHANNELS 2 +#define AD1938_ADC_16_CHANNELS 3 +#define AD1938_ADC_CHAN_SHFT 4 +#define AD1938_ADC_CHAN_MASK (3 << AD1938_ADC_CHAN_SHFT) +#define AD1938_ADC_LCR_MASTER (1 << 3) +#define AD1938_ADC_BCLK_MASTER (1 << 6) +#define AD1938_ADC_LEFT_HIGH (1 << 2) +#define AD1938_ADC_BCLK_INV (1 << 1) + +#define AD1938_NUM_REGS 17 + +extern struct snd_soc_dai ad1938_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad1938; +#endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index dd3380202766..0abec0d29a96 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -59,21 +59,6 @@ static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, return cache[reg]; } -static inline unsigned int ak4535_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 data; - data = reg; - - if (codec->hw_write(codec->control_data, &data, 1) != 1) - return -EIO; - - if (codec->hw_read(codec->control_data, &data, 1) != 1) - return -EIO; - - return data; -}; - /* * write ak4535 register cache */ @@ -635,7 +620,6 @@ static int ak4535_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { codec->hw_write = (hw_write_t)i2c_master_send; - codec->hw_read = (hw_read_t)i2c_master_recv; ret = ak4535_add_i2c_device(pdev, setup); } #endif diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c new file mode 100644 index 000000000000..e057c7b578df --- /dev/null +++ b/sound/soc/codecs/ak4642.c @@ -0,0 +1,502 @@ +/* + * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori@renesas.com> + * + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +/* ** CAUTION ** + * + * This is very simple driver. + * It can use headphone output / stereo input only + * + * AK4642 is not tested. + * AK4643 is tested. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "ak4642.h" + +#define AK4642_VERSION "0.0.1" + +#define PW_MGMT1 0x00 +#define PW_MGMT2 0x01 +#define SG_SL1 0x02 +#define SG_SL2 0x03 +#define MD_CTL1 0x04 +#define MD_CTL2 0x05 +#define TIMER 0x06 +#define ALC_CTL1 0x07 +#define ALC_CTL2 0x08 +#define L_IVC 0x09 +#define L_DVC 0x0a +#define ALC_CTL3 0x0b +#define R_IVC 0x0c +#define R_DVC 0x0d +#define MD_CTL3 0x0e +#define MD_CTL4 0x0f +#define PW_MGMT3 0x10 +#define DF_S 0x11 +#define FIL3_0 0x12 +#define FIL3_1 0x13 +#define FIL3_2 0x14 +#define FIL3_3 0x15 +#define EQ_0 0x16 +#define EQ_1 0x17 +#define EQ_2 0x18 +#define EQ_3 0x19 +#define EQ_4 0x1a +#define EQ_5 0x1b +#define FIL1_0 0x1c +#define FIL1_1 0x1d +#define FIL1_2 0x1e +#define FIL1_3 0x1f +#define PW_MGMT4 0x20 +#define MD_CTL5 0x21 +#define LO_MS 0x22 +#define HP_MS 0x23 +#define SPK_MS 0x24 + +#define AK4642_CACHEREGNUM 0x25 + +struct snd_soc_codec_device soc_codec_dev_ak4642; + +/* codec private data */ +struct ak4642_priv { + struct snd_soc_codec codec; + unsigned int sysclk; +}; + +static struct snd_soc_codec *ak4642_codec; + +/* + * ak4642 register cache + */ +static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0001, 0x0000, + 0x0002, 0x0000, 0x0000, 0x0000, + 0x00e1, 0x00e1, 0x0018, 0x0000, + 0x00e1, 0x0018, 0x0011, 0x0008, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, +}; + +/* + * read ak4642 register cache + */ +static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write ak4642 register cache + */ +static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= AK4642_CACHEREGNUM) + return; + + cache[reg] = value; +} + +/* + * write to the AK4642 register space + */ +static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D8 AK4642 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + ak4642_write_reg_cache(codec, reg, value); + return 0; + } else + return -EIO; +} + +static int ak4642_sync(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i, r = 0; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + r |= ak4642_write(codec, i, cache[i]); + + return r; +}; + +static int ak4642_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* + * start headphone output + * + * PLL, Master Mode + * Audio I/F Format :MSB justified (ADC & DAC) + * Sampling Frequency: 44.1kHz + * Digital Volume: −8dB + * Bass Boost Level : Middle + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p97. + * + * Example code use 0x39, 0x79 value for 0x01 address, + * But we need MCKO (0x02) bit now + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x0f, 0x09); + ak4642_write(codec, 0x0e, 0x19); + ak4642_write(codec, 0x09, 0x91); + ak4642_write(codec, 0x0c, 0x91); + ak4642_write(codec, 0x0a, 0x28); + ak4642_write(codec, 0x0d, 0x28); + ak4642_write(codec, 0x00, 0x64); + ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ + ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + } else { + /* + * start stereo input + * + * PLL Master Mode + * Audio I/F Format:MSB justified (ADC & DAC) + * Sampling Frequency:44.1kHz + * Pre MIC AMP:+20dB + * MIC Power On + * ALC setting:Refer to Table 35 + * ALC bit=“1” + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p94. + */ + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x02, 0x05); + ak4642_write(codec, 0x06, 0x3c); + ak4642_write(codec, 0x08, 0xe1); + ak4642_write(codec, 0x0b, 0x00); + ak4642_write(codec, 0x07, 0x21); + ak4642_write(codec, 0x00, 0x41); + ak4642_write(codec, 0x10, 0x01); + } + + return 0; +} + +static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + + if (is_play) { + /* stop headphone output */ + ak4642_write(codec, 0x01, 0x3b); + ak4642_write(codec, 0x01, 0x0b); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x0e, 0x11); + ak4642_write(codec, 0x0f, 0x08); + } else { + /* stop stereo input */ + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x10, 0x00); + ak4642_write(codec, 0x07, 0x01); + } +} + +static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct ak4642_priv *ak4642 = codec->private_data; + + ak4642->sysclk = freq; + return 0; +} + +static struct snd_soc_dai_ops ak4642_dai_ops = { + .startup = ak4642_dai_startup, + .shutdown = ak4642_dai_shutdown, + .set_sysclk = ak4642_dai_set_sysclk, +}; + +struct snd_soc_dai ak4642_dai = { + .name = "AK4642", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE }, + .ops = &ak4642_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4642_dai); + +static int ak4642_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + ak4642_sync(codec); + return 0; +} + +/* + * initialise the AK4642 driver + * register the mixer and dsp interfaces with the kernel + */ +static int ak4642_init(struct ak4642_priv *ak4642) +{ + struct snd_soc_codec *codec = &ak4642->codec; + int ret = 0; + + if (ak4642_codec) { + dev_err(codec->dev, "Another ak4642 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4642; + codec->name = "AK4642"; + codec->owner = THIS_MODULE; + codec->read = ak4642_read_reg_cache; + codec->write = ak4642_write; + codec->dai = &ak4642_dai; + codec->num_dai = 1; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); + codec->reg_cache = kmemdup(ak4642_reg, + sizeof(ak4642_reg), GFP_KERNEL); + + if (!codec->reg_cache) + return -ENOMEM; + + ak4642_dai.dev = codec->dev; + ak4642_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto reg_cache_err; + } + + ret = snd_soc_register_dai(&ak4642_dai); + if (ret) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto reg_cache_err; + } + + /* + * clock setting + * + * Audio I/F Format: MSB justified (ADC & DAC) + * BICK frequency at Master Mode: 64fs + * Input Master Clock Select at PLL Mode: 11.2896MHz + * MCKO: Enable + * Sampling Frequency: 44.1kHz + * + * This operation came from example code of + * "ASAHI KASEI AK4642" (japanese) manual p89. + * + * please fix-me + */ + ak4642_write(codec, 0x01, 0x08); + ak4642_write(codec, 0x04, 0x4a); + ak4642_write(codec, 0x05, 0x27); + ak4642_write(codec, 0x00, 0x40); + ak4642_write(codec, 0x01, 0x0b); + + return ret; + +reg_cache_err: + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + return ret; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int ak4642_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ak4642_priv *ak4642; + struct snd_soc_codec *codec; + int ret; + + ak4642 = kzalloc(sizeof(struct ak4642_priv), GFP_KERNEL); + if (!ak4642) + return -ENOMEM; + + codec = &ak4642->codec; + codec->dev = &i2c->dev; + + i2c_set_clientdata(i2c, ak4642); + codec->control_data = i2c; + + ret = ak4642_init(ak4642); + if (ret < 0) + printk(KERN_ERR "failed to initialise AK4642\n"); + + return ret; +} + +static int ak4642_i2c_remove(struct i2c_client *client) +{ + struct ak4642_priv *ak4642 = i2c_get_clientdata(client); + + snd_soc_unregister_dai(&ak4642_dai); + snd_soc_unregister_codec(&ak4642->codec); + kfree(ak4642->codec.reg_cache); + kfree(ak4642); + ak4642_codec = NULL; + + return 0; +} + +static const struct i2c_device_id ak4642_i2c_id[] = { + { "ak4642", 0 }, + { "ak4643", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id); + +static struct i2c_driver ak4642_i2c_driver = { + .driver = { + .name = "AK4642 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = ak4642_i2c_probe, + .remove = ak4642_i2c_remove, + .id_table = ak4642_i2c_id, +}; + +#endif + +static int ak4642_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + int ret; + + if (!ak4642_codec) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4642_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to create pcms\n"); + goto pcm_err; + } + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "ak4642: failed to register card\n"); + goto card_err; + } + + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; + +} + +/* power down chip */ +static int ak4642_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4642 = { + .probe = ak4642_probe, + .remove = ak4642_remove, + .resume = ak4642_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); + +static int __init ak4642_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&ak4642_i2c_driver); +#endif + return ret; + +} +module_init(ak4642_modinit); + +static void __exit ak4642_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&ak4642_i2c_driver); +#endif + +} +module_exit(ak4642_exit); + +MODULE_DESCRIPTION("Soc AK4642 driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4642.h b/sound/soc/codecs/ak4642.h new file mode 100644 index 000000000000..e476833d314e --- /dev/null +++ b/sound/soc/codecs/ak4642.h @@ -0,0 +1,20 @@ +/* + * ak4642.h -- AK4642 Soc Audio driver + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori@renesas.com> + * + * Based on ak4535.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AK4642_H +#define _AK4642_H + +extern struct snd_soc_dai ak4642_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4642; + +#endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b8226c8a4..ca1e24a8f12a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -806,15 +806,30 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + return snd_soc_suspend_device(codec->dev); } static int cs4270_i2c_resume(struct i2c_client *client) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + + return snd_soc_resume_device(codec->dev); +} + +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) +{ + struct snd_soc_codec *codec = cs4270_codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -840,6 +855,8 @@ static int cs4270_i2c_resume(struct i2c_client *client) #else #define cs4270_i2c_suspend NULL #define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* @@ -868,7 +885,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c new file mode 100644 index 000000000000..38eac9c866e1 --- /dev/null +++ b/sound/soc/codecs/cx20442.c @@ -0,0 +1,501 @@ +/* + * cx20442.c -- CX20442 ALSA Soc Audio driver + * + * Copyright 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> + * + * Initially based on sound/soc/codecs/wm8400.c + * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/tty.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "cx20442.h" + + +struct cx20442_priv { + struct snd_soc_codec codec; + u8 reg_cache[1]; +}; + +#define CX20442_PM 0x0 + +#define CX20442_TELIN 0 +#define CX20442_TELOUT 1 +#define CX20442_MIC 2 +#define CX20442_SPKOUT 3 +#define CX20442_AGC 4 + +static const struct snd_soc_dapm_widget cx20442_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("TELOUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("AGCOUT"), + + SND_SOC_DAPM_MIXER("SPKOUT Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA("TELOUT Amp", CX20442_PM, CX20442_TELOUT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKOUT Amp", CX20442_PM, CX20442_SPKOUT, 0, NULL, 0), + SND_SOC_DAPM_PGA("SPKOUT AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MICBIAS("TELIN Bias", CX20442_PM, CX20442_TELIN, 0), + SND_SOC_DAPM_MICBIAS("MIC Bias", CX20442_PM, CX20442_MIC, 0), + + SND_SOC_DAPM_PGA("MIC AGC", CX20442_PM, CX20442_AGC, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("TELIN"), + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("AGCIN"), +}; + +static const struct snd_soc_dapm_route cx20442_audio_map[] = { + {"TELOUT", NULL, "TELOUT Amp"}, + + {"SPKOUT", NULL, "SPKOUT Mixer"}, + {"SPKOUT Mixer", NULL, "SPKOUT Amp"}, + + {"TELOUT Amp", NULL, "DAC"}, + {"SPKOUT Amp", NULL, "DAC"}, + + {"SPKOUT Mixer", NULL, "SPKOUT AGC"}, + {"SPKOUT AGC", NULL, "AGCIN"}, + + {"AGCOUT", NULL, "MIC AGC"}, + {"MIC AGC", NULL, "MIC"}, + + {"MIC Bias", NULL, "MIC"}, + {"Input Mixer", NULL, "MIC Bias"}, + + {"TELIN Bias", NULL, "TELIN"}, + {"Input Mixer", NULL, "TELIN Bias"}, + + {"ADC", NULL, "Input Mixer"}, +}; + +static int cx20442_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, cx20442_dapm_widgets, + ARRAY_SIZE(cx20442_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, cx20442_audio_map, + ARRAY_SIZE(cx20442_audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int cx20442_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +enum v253_vls { + V253_VLS_NONE = 0, + V253_VLS_T, + V253_VLS_L, + V253_VLS_LT, + V253_VLS_S, + V253_VLS_ST, + V253_VLS_M, + V253_VLS_MST, + V253_VLS_S1, + V253_VLS_S1T, + V253_VLS_MS1T, + V253_VLS_M1, + V253_VLS_M1ST, + V253_VLS_M1S1T, + V253_VLS_H, + V253_VLS_HT, + V253_VLS_MS, + V253_VLS_MS1, + V253_VLS_M1S, + V253_VLS_M1S1, + V253_VLS_TEST, +}; + +static int cx20442_pm_to_v253_vls(u8 value) +{ + switch (value & ~(1 << CX20442_AGC)) { + case 0: + return V253_VLS_T; + case (1 << CX20442_SPKOUT): + case (1 << CX20442_MIC): + case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): + return V253_VLS_M1S1; + case (1 << CX20442_TELOUT): + case (1 << CX20442_TELIN): + case (1 << CX20442_TELOUT) | (1 << CX20442_TELIN): + return V253_VLS_L; + case (1 << CX20442_TELOUT) | (1 << CX20442_MIC): + return V253_VLS_NONE; + } + return -EINVAL; +} +static int cx20442_pm_to_v253_vsp(u8 value) +{ + switch (value & ~(1 << CX20442_AGC)) { + case (1 << CX20442_SPKOUT): + case (1 << CX20442_MIC): + case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): + return (bool)(value & (1 << CX20442_AGC)); + } + return (value & (1 << CX20442_AGC)) ? -EINVAL : 0; +} + +static int cx20442_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *reg_cache = codec->reg_cache; + int vls, vsp, old, len; + char buf[18]; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + /* hw_write and control_data pointers required for talking to the modem + * are expected to be set by the line discipline initialization code */ + if (!codec->hw_write || !codec->control_data) + return -EIO; + + old = reg_cache[reg]; + reg_cache[reg] = value; + + vls = cx20442_pm_to_v253_vls(value); + if (vls < 0) + return vls; + + vsp = cx20442_pm_to_v253_vsp(value); + if (vsp < 0) + return vsp; + + if ((vls == V253_VLS_T) || + (vls == cx20442_pm_to_v253_vls(old))) { + if (vsp == cx20442_pm_to_v253_vsp(old)) + return 0; + len = snprintf(buf, ARRAY_SIZE(buf), "at+vsp=%d\r", vsp); + } else if (vsp == cx20442_pm_to_v253_vsp(old)) + len = snprintf(buf, ARRAY_SIZE(buf), "at+vls=%d\r", vls); + else + len = snprintf(buf, ARRAY_SIZE(buf), + "at+vls=%d;+vsp=%d\r", vls, vsp); + + if (unlikely(len > (ARRAY_SIZE(buf) - 1))) + return -ENOMEM; + + dev_dbg(codec->dev, "%s: %s\n", __func__, buf); + if (codec->hw_write(codec->control_data, buf, len) != len) + return -EIO; + + return 0; +} + + +/* Moved up here as line discipline referres it during initialization */ +static struct snd_soc_codec *cx20442_codec; + + +/* + * Line discpline related code + * + * Any of the callback functions below can be used in two ways: + * 1) registerd by a machine driver as one of line discipline operations, + * 2) called from a machine's provided line discipline callback function + * in case when extra machine specific code must be run as well. + */ + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *v253_init = "ate0m0q0+fclass=8\r"; + +/* Line discipline .open() */ +static int v253_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = cx20442_codec; + int ret, len = strlen(v253_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass the codec structure address for use by other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, v253_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Line discipline .close() */ +static void v253_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + tty->disc_data = NULL; + + if (!codec) + return; + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; +} + +/* Line discipline .hangup() */ +static int v253_hangup(struct tty_struct *tty) +{ + v253_close(tty); + return 0; +} + +/* Line discipline .receive_buf() */ +static void v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + + if (!codec) + return; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Set up codec driver access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + } +} + +/* Line discipline .write_wakeup() */ +static void v253_wakeup(struct tty_struct *tty) +{ +} + +struct tty_ldisc_ops v253_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx20442", + .owner = THIS_MODULE, + .open = v253_open, + .close = v253_close, + .hangup = v253_hangup, + .receive_buf = v253_receive, + .write_wakeup = v253_wakeup, +}; +EXPORT_SYMBOL_GPL(v253_ops); + + +/* + * Codec DAI + */ + +struct snd_soc_dai cx20442_dai = { + .name = "CX20442", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; +EXPORT_SYMBOL_GPL(cx20442_dai); + +static int cx20442_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!cx20442_codec) { + dev_err(&pdev->dev, "cx20442 not yet discovered\n"); + return -ENODEV; + } + codec = cx20442_codec; + + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + goto pcm_err; + } + + cx20442_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int cx20442_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device cx20442_codec_dev = { + .probe = cx20442_codec_probe, + .remove = cx20442_codec_remove, +}; +EXPORT_SYMBOL_GPL(cx20442_codec_dev); + +static int cx20442_register(struct cx20442_priv *cx20442) +{ + struct snd_soc_codec *codec = &cx20442->codec; + int ret; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CX20442"; + codec->owner = THIS_MODULE; + codec->private_data = cx20442; + + codec->dai = &cx20442_dai; + codec->num_dai = 1; + + codec->reg_cache = &cx20442->reg_cache; + codec->reg_cache_size = ARRAY_SIZE(cx20442->reg_cache); + codec->read = cx20442_read_reg_cache; + codec->write = cx20442_write; + + codec->bias_level = SND_SOC_BIAS_OFF; + + cx20442_dai.dev = codec->dev; + + cx20442_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&cx20442_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + cx20442_codec = NULL; + kfree(cx20442); + return ret; +} + +static void cx20442_unregister(struct cx20442_priv *cx20442) +{ + snd_soc_unregister_dai(&cx20442_dai); + snd_soc_unregister_codec(&cx20442->codec); + + cx20442_codec = NULL; + kfree(cx20442); +} + +static int cx20442_platform_probe(struct platform_device *pdev) +{ + struct cx20442_priv *cx20442; + struct snd_soc_codec *codec; + + cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); + if (cx20442 == NULL) + return -ENOMEM; + + codec = &cx20442->codec; + + codec->control_data = NULL; + codec->hw_write = NULL; + codec->pop_time = 0; + + codec->dev = &pdev->dev; + platform_set_drvdata(pdev, cx20442); + + return cx20442_register(cx20442); +} + +static int __exit cx20442_platform_remove(struct platform_device *pdev) +{ + struct cx20442_priv *cx20442 = platform_get_drvdata(pdev); + + cx20442_unregister(cx20442); + return 0; +} + +static struct platform_driver cx20442_platform_driver = { + .driver = { + .name = "cx20442", + .owner = THIS_MODULE, + }, + .probe = cx20442_platform_probe, + .remove = __exit_p(cx20442_platform_remove), +}; + +static int __init cx20442_init(void) +{ + return platform_driver_register(&cx20442_platform_driver); +} +module_init(cx20442_init); + +static void __exit cx20442_exit(void) +{ + platform_driver_unregister(&cx20442_platform_driver); +} +module_exit(cx20442_exit); + +MODULE_DESCRIPTION("ASoC CX20442-11 voice modem codec driver"); +MODULE_AUTHOR("Janusz Krzysztofik"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:cx20442"); diff --git a/sound/soc/codecs/cx20442.h b/sound/soc/codecs/cx20442.h new file mode 100644 index 000000000000..688a5eb62e17 --- /dev/null +++ b/sound/soc/codecs/cx20442.h @@ -0,0 +1,20 @@ +/* + * cx20442.h -- audio driver for CX20442 + * + * Copyright 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _CX20442_CODEC_H +#define _CX20442_CODEC_H + +extern struct snd_soc_dai cx20442_dai; +extern struct snd_soc_codec_device cx20442_codec_dev; +extern struct tty_ldisc_ops v253_ops; + +#endif diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c new file mode 100644 index 000000000000..9e7e964a5fa3 --- /dev/null +++ b/sound/soc/codecs/max9877.c @@ -0,0 +1,308 @@ +/* + * max9877.c -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9877.h" + +static struct i2c_client *i2c; + +static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; + +static void max9877_write_regs(void) +{ + unsigned int i; + u8 data[6]; + + data[0] = MAX9877_INPUT_MODE; + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + data[i + 1] = max9877_regs[i]; + + if (i2c_master_send(i2c, data, 6) != 6) + dev_err(&i2c->dev, "i2c write failed\n"); +} + +static int max9877_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; + + return 0; +} + +static int max9877_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + + if (invert) + val = mask - val; + + if (((max9877_regs[reg] >> shift) & mask) == val) + return 0; + + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= val << shift; + max9877_write_regs(); + + return 1; +} + +static int max9877_get_2reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + + ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + + return 0; +} + +static int max9877_set_2reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val2 = (ucontrol->value.integer.value[1] & mask); + unsigned int change = 1; + + if (((max9877_regs[reg] >> shift) & mask) == val) + change = 0; + + if (((max9877_regs[reg2] >> shift) & mask) == val2) + change = 0; + + if (change) { + max9877_regs[reg] &= ~(mask << shift); + max9877_regs[reg] |= val << shift; + max9877_regs[reg2] &= ~(mask << shift); + max9877_regs[reg2] |= val2 << shift; + max9877_write_regs(); + } + + return change; +} + +static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + + if (value) + value -= 1; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + value += 1; + + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + + value = value >> MAX9877_OSC_OFFSET; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + value = value << MAX9877_OSC_OFFSET; + if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + return 0; + + max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; + max9877_regs[MAX9877_OUTPUT_MODE] |= value; + max9877_write_regs(); + return 1; +} + +static const unsigned int max9877_pgain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), + 2, 2, TLV_DB_SCALE_ITEM(2000, 0, 0), +}; + +static const unsigned int max9877_output_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const char *max9877_out_mode[] = { + "INA -> SPK", + "INA -> HP", + "INA -> SPK and HP", + "INB -> SPK", + "INB -> HP", + "INB -> SPK and HP", + "INA + INB -> SPK", + "INA + INB -> HP", + "INA + INB -> SPK and HP", +}; + +static const char *max9877_osc_mode[] = { + "1176KHz", + "1100KHz", + "700KHz", +}; + +static const struct soc_enum max9877_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), +}; + +static const struct snd_kcontrol_new max9877_controls[] = { + SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, + max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), + SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, + max9877_get_reg, max9877_set_reg, max9877_output_tlv), + SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", + MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_get_2reg, max9877_set_2reg, max9877_output_tlv), + SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0, + max9877_get_reg, max9877_set_reg), + SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1, + max9877_get_reg, max9877_set_reg), + SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], + max9877_get_out_mode, max9877_set_out_mode), + SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], + max9877_get_osc_mode, max9877_set_osc_mode), +}; + +/* This function is called from ASoC machine driver */ +int max9877_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, max9877_controls, + ARRAY_SIZE(max9877_controls)); +} +EXPORT_SYMBOL_GPL(max9877_add_controls); + +static int __devinit max9877_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + i2c = client; + + max9877_write_regs(); + + return 0; +} + +static __devexit int max9877_i2c_remove(struct i2c_client *client) +{ + i2c = NULL; + + return 0; +} + +static const struct i2c_device_id max9877_i2c_id[] = { + { "max9877", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9877_i2c_id); + +static struct i2c_driver max9877_i2c_driver = { + .driver = { + .name = "max9877", + .owner = THIS_MODULE, + }, + .probe = max9877_i2c_probe, + .remove = __devexit_p(max9877_i2c_remove), + .id_table = max9877_i2c_id, +}; + +static int __init max9877_init(void) +{ + return i2c_add_driver(&max9877_i2c_driver); +} +module_init(max9877_init); + +static void __exit max9877_exit(void) +{ + i2c_del_driver(&max9877_i2c_driver); +} +module_exit(max9877_exit); + +MODULE_DESCRIPTION("ASoC MAX9877 amp driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h new file mode 100644 index 000000000000..6da72290ac58 --- /dev/null +++ b/sound/soc/codecs/max9877.h @@ -0,0 +1,37 @@ +/* + * max9877.h -- amp driver for max9877 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _MAX9877_H +#define _MAX9877_H + +#define MAX9877_INPUT_MODE 0x00 +#define MAX9877_SPK_VOLUME 0x01 +#define MAX9877_HPL_VOLUME 0x02 +#define MAX9877_HPR_VOLUME 0x03 +#define MAX9877_OUTPUT_MODE 0x04 + +/* MAX9877_INPUT_MODE */ +#define MAX9877_INB (1 << 4) +#define MAX9877_INA (1 << 5) +#define MAX9877_ZCD (1 << 6) + +/* MAX9877_OUTPUT_MODE */ +#define MAX9877_OUTMODE_MASK (15 << 0) +#define MAX9877_OSC_MASK (3 << 4) +#define MAX9877_OSC_OFFSET 4 +#define MAX9877_BYPASS (1 << 6) +#define MAX9877_SHDN (1 << 7) + +extern int max9877_add_controls(struct snd_soc_codec *codec); + +#endif diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index 218b33adad90..a63191141052 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -21,6 +21,8 @@ #include "spdif_transciever.h" +MODULE_LICENSE("GPL"); + #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE @@ -34,6 +36,7 @@ struct snd_soc_dai dit_stub_dai = { .formats = STUB_FORMATS, }, }; +EXPORT_SYMBOL_GPL(dit_stub_dai); static int spdif_dit_probe(struct platform_device *pdev) { diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 8ad4b7b3e3ba..befc6488c39a 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -149,7 +149,7 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val); @@ -168,7 +168,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb0d1bf34b57..3395cf945d56 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -53,6 +53,7 @@ /* codec private data */ struct aic3x_priv { + struct snd_soc_codec codec; unsigned int sysclk; int master; }; @@ -145,8 +146,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, u8 *value) { *value = reg & 0xff; - if (codec->hw_read(codec->control_data, value, 1) != 1) - return -EIO; + + value[0] = i2c_smbus_read_byte_data(codec->control_data, value[0]); aic3x_write_reg_cache(codec, reg, *value); return 0; @@ -1156,11 +1157,13 @@ static int aic3x_resume(struct platform_device *pdev) * initialise the AIC3X driver * register the mixer and dsp interfaces with the kernel */ -static int aic3x_init(struct snd_soc_device *socdev) +static int aic3x_init(struct snd_soc_codec *codec) { - struct snd_soc_codec *codec = socdev->card->codec; - struct aic3x_setup_data *setup = socdev->codec_data; - int reg, ret = 0; + int reg; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); codec->name = "tlv320aic3x"; codec->owner = THIS_MODULE; @@ -1177,13 +1180,6 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); aic3x_write(codec, AIC3X_RESET, SOFT_RESET); - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "aic3x: failed to create pcms\n"); - goto pcm_err; - } - /* DAC default volume and mute */ aic3x_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); aic3x_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); @@ -1250,30 +1246,51 @@ static int aic3x_init(struct snd_soc_device *socdev) /* off, with power on */ aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* setup GPIO functions */ - aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); - aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); + return 0; +} - snd_soc_add_controls(codec, aic3x_snd_controls, - ARRAY_SIZE(aic3x_snd_controls)); - aic3x_add_widgets(codec); - ret = snd_soc_init_card(socdev); +static struct snd_soc_codec *aic3x_codec; + +static int aic3x_register(struct snd_soc_codec *codec) +{ + int ret; + + ret = aic3x_init(codec); if (ret < 0) { - printk(KERN_ERR "aic3x: failed to register card\n"); - goto card_err; + dev_err(codec->dev, "Failed to initialise device\n"); + return ret; } - return ret; + aic3x_codec = codec; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); - return ret; + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec\n"); + return ret; + } + + ret = snd_soc_register_dai(&aic3x_dai); + if (ret) { + dev_err(codec->dev, "Failed to register dai\n"); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; } -static struct snd_soc_device *aic3x_socdev; +static int aic3x_unregister(struct aic3x_priv *aic3x) +{ + aic3x_set_bias_level(&aic3x->codec, SND_SOC_BIAS_OFF); + + snd_soc_unregister_dai(&aic3x_dai); + snd_soc_unregister_codec(&aic3x->codec); + + kfree(aic3x); + aic3x_codec = NULL; + + return 0; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* @@ -1288,28 +1305,36 @@ static struct snd_soc_device *aic3x_socdev; static int aic3x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = aic3x_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct aic3x_priv *aic3x; - i2c_set_clientdata(i2c, codec); + aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); + if (aic3x == NULL) { + dev_err(&i2c->dev, "failed to create private data\n"); + return -ENOMEM; + } + + codec = &aic3x->codec; + codec->dev = &i2c->dev; + codec->private_data = aic3x; codec->control_data = i2c; + codec->hw_write = (hw_write_t) i2c_master_send; - ret = aic3x_init(socdev); - if (ret < 0) - printk(KERN_ERR "aic3x: failed to initialise AIC3X\n"); - return ret; + i2c_set_clientdata(i2c, aic3x); + + return aic3x_register(codec); } static int aic3x_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); - return 0; + struct aic3x_priv *aic3x = i2c_get_clientdata(client); + + return aic3x_unregister(aic3x); } static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", 0 }, + { "tlv320aic33", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1320,56 +1345,28 @@ static struct i2c_driver aic3x_i2c_driver = { .name = "aic3x I2C Codec", .owner = THIS_MODULE, }, - .probe = aic3x_i2c_probe, + .probe = aic3x_i2c_probe, .remove = aic3x_i2c_remove, .id_table = aic3x_i2c_id, }; -static int aic3x_i2c_read(struct i2c_client *client, u8 *value, int len) +static inline void aic3x_i2c_init(void) { - value[0] = i2c_smbus_read_byte_data(client, value[0]); - return (len == 1); -} - -static int aic3x_add_i2c_device(struct platform_device *pdev, - const struct aic3x_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; int ret; ret = i2c_add_driver(&aic3x_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "tlv320aic3x", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; + if (ret) + printk(KERN_ERR "%s: error regsitering i2c driver, %d\n", + __func__, ret); +} -err_driver: +static inline void aic3x_i2c_exit(void) +{ i2c_del_driver(&aic3x_i2c_driver); - return -ENODEV; } +#else +static inline void aic3x_i2c_init(void) { } +static inline void aic3x_i2c_exit(void) { } #endif static int aic3x_probe(struct platform_device *pdev) @@ -1377,43 +1374,51 @@ static int aic3x_probe(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct aic3x_setup_data *setup; struct snd_soc_codec *codec; - struct aic3x_priv *aic3x; int ret = 0; - printk(KERN_INFO "AIC3X Audio Codec %s\n", AIC3X_VERSION); + codec = aic3x_codec; + if (!codec) { + dev_err(&pdev->dev, "Codec not registered\n"); + return -ENODEV; + } + socdev->card->codec = codec; setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); - if (aic3x == NULL) { - kfree(codec); - return -ENOMEM; + if (setup) { + /* setup GPIO functions */ + aic3x_write(codec, AIC3X_GPIO1_REG, + (setup->gpio_func[0] & 0xf) << 4); + aic3x_write(codec, AIC3X_GPIO2_REG, + (setup->gpio_func[1] & 0xf) << 4); } - codec->private_data = aic3x; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - - aic3x_socdev = socdev; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t) i2c_master_send; - codec->hw_read = (hw_read_t) aic3x_i2c_read; - ret = aic3x_add_i2c_device(pdev, setup); + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic3x: failed to create pcms\n"); + goto pcm_err; } -#else - /* Add other interfaces here */ -#endif - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); + snd_soc_add_controls(codec, aic3x_snd_controls, + ARRAY_SIZE(aic3x_snd_controls)); + + aic3x_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "aic3x: failed to register card\n"); + goto card_err; } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + +pcm_err: + kfree(codec->reg_cache); return ret; } @@ -1428,12 +1433,8 @@ static int aic3x_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&aic3x_i2c_driver); -#endif - kfree(codec->private_data); - kfree(codec); + + kfree(codec->reg_cache); return 0; } @@ -1448,13 +1449,15 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x); static int __init aic3x_modinit(void) { - return snd_soc_register_dai(&aic3x_dai); + aic3x_i2c_init(); + + return 0; } module_init(aic3x_modinit); static void __exit aic3x_exit(void) { - snd_soc_unregister_dai(&aic3x_dai); + aic3x_i2c_exit(); } module_exit(aic3x_exit); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index ac827e578c4d..9af1c886213c 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -282,8 +282,6 @@ int aic3x_headset_detected(struct snd_soc_codec *codec); int aic3x_button_pressed(struct snd_soc_codec *codec); struct aic3x_setup_data { - int i2c_bus; - unsigned short i2c_address; unsigned int gpio_func[2]; }; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4dbb853eef5a..4df7c6c61c76 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -225,55 +225,11 @@ static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) return; if (mute) { - /* Bypass the reg_cache and mute the volumes - * Headset mute is done in it's own event handler - * Things to mute: Earpiece, PreDrivL/R, CarkitL/R - */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL); - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_EAR_GAIN), - TWL4030_REG_EAR_CTL); - - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL); - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PREDL_GAIN), - TWL4030_REG_PREDL_CTL); - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL); - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PREDR_GAIN), - TWL4030_REG_PREDL_CTL); - - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL); - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PRECKL_GAIN), - TWL4030_REG_PRECKL_CTL); - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL); - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - reg_val & (~TWL4030_PRECKR_GAIN), - TWL4030_REG_PRECKR_CTL); - /* Disable PLL */ reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); reg_val &= ~TWL4030_APLL_EN; twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); } else { - /* Restore the volumes - * Headset mute is done in it's own event handler - * Things to restore: Earpiece, PreDrivL/R, CarkitL/R - */ - twl4030_write(codec, TWL4030_REG_EAR_CTL, - twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL)); - - twl4030_write(codec, TWL4030_REG_PREDL_CTL, - twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL)); - twl4030_write(codec, TWL4030_REG_PREDR_CTL, - twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL)); - - twl4030_write(codec, TWL4030_REG_PRECKL_CTL, - twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL)); - twl4030_write(codec, TWL4030_REG_PRECKR_CTL, - twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL)); - /* Enable PLL */ reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); reg_val |= TWL4030_APLL_EN; @@ -443,16 +399,20 @@ SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); /* Left analog microphone selection */ static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { - SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0), - SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0), - SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0), - SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0), + SOC_DAPM_SINGLE("Main Mic Capture Switch", + TWL4030_REG_ANAMICL, 0, 1, 0), + SOC_DAPM_SINGLE("Headset Mic Capture Switch", + TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("AUXL Capture Switch", + TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Carkit Mic Capture Switch", + TWL4030_REG_ANAMICL, 3, 1, 0), }; /* Right analog microphone selection */ static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { - SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), - SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0), + SOC_DAPM_SINGLE("Sub Mic Capture Switch", TWL4030_REG_ANAMICR, 0, 1, 0), + SOC_DAPM_SINGLE("AUXR Capture Switch", TWL4030_REG_ANAMICR, 2, 1, 0), }; /* TX1 L/R Analog/Digital microphone selection */ @@ -560,6 +520,41 @@ static int micpath_event(struct snd_soc_dapm_widget *w, return 0; } +/* + * Output PGA builder: + * Handle the muting and unmuting of the given output (turning off the + * amplifier associated with the output pin) + * On mute bypass the reg_cache and mute the volume + * On unmute: restore the register content + * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R + */ +#define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ +static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ + struct snd_kcontrol *kcontrol, int event) \ +{ \ + u8 reg_val; \ + \ + switch (event) { \ + case SND_SOC_DAPM_POST_PMU: \ + twl4030_write(w->codec, reg, \ + twl4030_read_reg_cache(w->codec, reg)); \ + break; \ + case SND_SOC_DAPM_POST_PMD: \ + reg_val = twl4030_read_reg_cache(w->codec, reg); \ + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + reg_val & (~mask), \ + reg); \ + break; \ + } \ + return 0; \ +} + +TWL4030_OUTPUT_PGA(earpiece, TWL4030_REG_EAR_CTL, TWL4030_EAR_GAIN); +TWL4030_OUTPUT_PGA(predrivel, TWL4030_REG_PREDL_CTL, TWL4030_PREDL_GAIN); +TWL4030_OUTPUT_PGA(predriver, TWL4030_REG_PREDR_CTL, TWL4030_PREDR_GAIN); +TWL4030_OUTPUT_PGA(carkitl, TWL4030_REG_PRECKL_CTL, TWL4030_PRECKL_GAIN); +TWL4030_OUTPUT_PGA(carkitr, TWL4030_REG_PRECKR_CTL, TWL4030_PRECKR_GAIN); + static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp) { unsigned char hs_ctl; @@ -620,6 +615,9 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, static void headset_ramp(struct snd_soc_codec *codec, int ramp) { + struct snd_soc_device *socdev = codec->socdev; + struct twl4030_setup_data *setup = socdev->codec_data; + unsigned char hs_gain, hs_pop; struct twl4030_priv *twl4030 = codec->private_data; /* Base values for ramp delay calculation: 2^19 - 2^26 */ @@ -629,6 +627,17 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + /* Enable external mute control, this dramatically reduces + * the pop-noise */ + if (setup && setup->hs_extmute) { + if (setup->set_hs_extmute) { + setup->set_hs_extmute(1); + } else { + hs_pop |= TWL4030_EXTMUTE; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } + } + if (ramp) { /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; @@ -636,6 +645,9 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Wait ramp delay time + 1, so the VMID can settle */ + mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / + twl4030->sysclk) + 1); } else { /* Headset ramp-down _not_ according to * the TRM, but in a way that it is working */ @@ -652,6 +664,16 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) hs_pop &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); } + + /* Disable external mute */ + if (setup && setup->hs_extmute) { + if (setup->set_hs_extmute) { + setup->set_hs_extmute(0); + } else { + hs_pop &= ~TWL4030_EXTMUTE; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); + } + } } static int headsetlpga_event(struct snd_soc_dapm_widget *w, @@ -712,7 +734,19 @@ static int bypass_event(struct snd_soc_dapm_widget *w, reg = twl4030_read_reg_cache(w->codec, m->reg); - if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* + * bypass_state[0:3] - analog HiFi bypass + * bypass_state[4] - analog voice bypass + * bypass_state[5] - digital voice bypass + * bypass_state[6:7] - digital HiFi bypass + */ + if (m->reg == TWL4030_REG_VSTPGA) { + /* Voice digital bypass */ + if (reg) + twl4030->bypass_state |= (1 << 5); + else + twl4030->bypass_state &= ~(1 << 5); + } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { /* Analog bypass */ if (reg & (1 << m->shift)) twl4030->bypass_state |= @@ -726,12 +760,6 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state |= (1 << 4); else twl4030->bypass_state &= ~(1 << 4); - } else if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); } else { /* Digital bypass */ if (reg & (0x7 << m->shift)) @@ -924,7 +952,7 @@ static const struct soc_enum twl4030_op_modes_enum = ARRAY_SIZE(twl4030_op_modes_texts), twl4030_op_modes_texts); -int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, +static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); @@ -1005,6 +1033,16 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +/* AVADC clock priority */ +static const char *twl4030_avadc_clk_priority_texts[] = { + "Voice high priority", "HiFi high priority" +}; + +static const struct soc_enum twl4030_avadc_clk_priority_enum = + SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, + ARRAY_SIZE(twl4030_avadc_clk_priority_texts), + twl4030_avadc_clk_priority_texts); + static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", "437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms", @@ -1106,6 +1144,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN, 0, 3, 5, 0, input_gain_tlv), + SOC_ENUM("AVADC Clock Priority", twl4030_avadc_clk_priority_enum), + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), @@ -1208,13 +1248,22 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_earpiece_controls[0], ARRAY_SIZE(twl4030_dapm_earpiece_controls)), + SND_SOC_DAPM_PGA_E("Earpiece PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, earpiecepga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* PreDrivL/R */ SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_predrivel_controls[0], ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_PGA_E("PredriveL PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, predrivelpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_predriver_controls[0], ARRAY_SIZE(twl4030_dapm_predriver_controls)), + SND_SOC_DAPM_PGA_E("PredriveR PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, predriverpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* HeadsetL/R */ SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsol_controls[0], @@ -1232,22 +1281,28 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitl_controls[0], ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_PGA_E("CarkitL PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, carkitlpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitr_controls[0], ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + SND_SOC_DAPM_PGA_E("CarkitR PGA", SND_SOC_NOPM, + 0, 0, NULL, 0, carkitrpga_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeL", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreelmute_control), SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreelpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0, &twl4030_dapm_handsfreer_control), - SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("HandsfreeR", SND_SOC_NOPM, 0, 0, &twl4030_dapm_handsfreermute_control), SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM, 0, 0, NULL, 0, handsfreerpga_event, @@ -1282,11 +1337,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_REG), /* Analog input mixers for the capture amplifiers */ - SND_SOC_DAPM_MIXER("Analog Left Capture Route", + SND_SOC_DAPM_MIXER("Analog Left", TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_controls[0], ARRAY_SIZE(twl4030_dapm_analoglmic_controls)), - SND_SOC_DAPM_MIXER("Analog Right Capture Route", + SND_SOC_DAPM_MIXER("Analog Right", TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_controls[0], ARRAY_SIZE(twl4030_dapm_analogrmic_controls)), @@ -1326,16 +1381,19 @@ static const struct snd_soc_dapm_route intercon[] = { {"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"}, {"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"}, {"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"}, + {"Earpiece PGA", NULL, "Earpiece Mixer"}, /* PreDrivL */ {"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, {"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, {"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"}, + {"PredriveL PGA", NULL, "PredriveL Mixer"}, /* PreDrivR */ {"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"}, {"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, {"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, {"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"PredriveR PGA", NULL, "PredriveR Mixer"}, /* HeadsetL */ {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, @@ -1350,24 +1408,26 @@ static const struct snd_soc_dapm_route intercon[] = { {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"}, {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"}, {"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"}, + {"CarkitL PGA", NULL, "CarkitL Mixer"}, /* CarkitR */ {"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"}, {"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"}, {"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"}, + {"CarkitR PGA", NULL, "CarkitR Mixer"}, /* HandsfreeL */ {"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"}, {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"}, {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"}, - {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"}, - {"HandsfreeL PGA", NULL, "HandsfreeL Switch"}, + {"HandsfreeL", "Switch", "HandsfreeL Mux"}, + {"HandsfreeL PGA", NULL, "HandsfreeL"}, /* HandsfreeR */ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"}, {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"}, {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"}, {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"}, - {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"}, - {"HandsfreeR PGA", NULL, "HandsfreeR Switch"}, + {"HandsfreeR", "Switch", "HandsfreeR Mux"}, + {"HandsfreeR PGA", NULL, "HandsfreeR"}, /* Vibra */ {"Vibra Mux", "AudioL1", "DAC Left1"}, {"Vibra Mux", "AudioR1", "DAC Right1"}, @@ -1377,29 +1437,29 @@ static const struct snd_soc_dapm_route intercon[] = { /* outputs */ {"OUTL", NULL, "Analog L2 Playback Mixer"}, {"OUTR", NULL, "Analog R2 Playback Mixer"}, - {"EARPIECE", NULL, "Earpiece Mixer"}, - {"PREDRIVEL", NULL, "PredriveL Mixer"}, - {"PREDRIVER", NULL, "PredriveR Mixer"}, + {"EARPIECE", NULL, "Earpiece PGA"}, + {"PREDRIVEL", NULL, "PredriveL PGA"}, + {"PREDRIVER", NULL, "PredriveR PGA"}, {"HSOL", NULL, "HeadsetL PGA"}, {"HSOR", NULL, "HeadsetR PGA"}, - {"CARKITL", NULL, "CarkitL Mixer"}, - {"CARKITR", NULL, "CarkitR Mixer"}, + {"CARKITL", NULL, "CarkitL PGA"}, + {"CARKITR", NULL, "CarkitR PGA"}, {"HFL", NULL, "HandsfreeL PGA"}, {"HFR", NULL, "HandsfreeR PGA"}, {"Vibra Route", "Audio", "Vibra Mux"}, {"VIBRA", NULL, "Vibra Route"}, /* Capture path */ - {"Analog Left Capture Route", "Main mic", "MAINMIC"}, - {"Analog Left Capture Route", "Headset mic", "HSMIC"}, - {"Analog Left Capture Route", "AUXL", "AUXL"}, - {"Analog Left Capture Route", "Carkit mic", "CARKITMIC"}, + {"Analog Left", "Main Mic Capture Switch", "MAINMIC"}, + {"Analog Left", "Headset Mic Capture Switch", "HSMIC"}, + {"Analog Left", "AUXL Capture Switch", "AUXL"}, + {"Analog Left", "Carkit Mic Capture Switch", "CARKITMIC"}, - {"Analog Right Capture Route", "Sub mic", "SUBMIC"}, - {"Analog Right Capture Route", "AUXR", "AUXR"}, + {"Analog Right", "Sub Mic Capture Switch", "SUBMIC"}, + {"Analog Right", "AUXR Capture Switch", "AUXR"}, - {"ADC Physical Left", NULL, "Analog Left Capture Route"}, - {"ADC Physical Right", NULL, "Analog Right Capture Route"}, + {"ADC Physical Left", NULL, "Analog Left"}, + {"ADC Physical Right", NULL, "Analog Right"}, {"Digimic0 Enable", NULL, "DIGIMIC0"}, {"Digimic1 Enable", NULL, "DIGIMIC1"}, @@ -1423,11 +1483,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, /* Analog bypass routes */ - {"Right1 Analog Loopback", "Switch", "Analog Right Capture Route"}, - {"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"}, - {"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"}, - {"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"}, - {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"}, + {"Right1 Analog Loopback", "Switch", "Analog Right"}, + {"Left1 Analog Loopback", "Switch", "Analog Left"}, + {"Right2 Analog Loopback", "Switch", "Analog Right"}, + {"Left2 Analog Loopback", "Switch", "Analog Left"}, + {"Voice Analog Loopback", "Switch", "Analog Left"}, {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, @@ -1609,8 +1669,6 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* If the substream has 4 channel, do the necessary setup */ if (params_channels(params) == 4) { - u8 format, mode; - format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); @@ -1806,6 +1864,19 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + + if (tristate) + reg |= TWL4030_AIF_TRI_EN; + else + reg &= ~TWL4030_AIF_TRI_EN; + + return twl4030_write(codec, TWL4030_REG_AUDIO_IF, reg); +} + /* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R * (VTXL, VTXR) for uplink has to be enabled/disabled. */ static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, @@ -1948,7 +2019,7 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: @@ -1980,6 +2051,19 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + + if (tristate) + reg |= TWL4030_VIF_TRI_EN; + else + reg &= ~TWL4030_VIF_TRI_EN; + + return twl4030_write(codec, TWL4030_REG_VOICE_IF, reg); +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1989,6 +2073,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .hw_params = twl4030_hw_params, .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, + .set_tristate = twl4030_set_tristate, }; static struct snd_soc_dai_ops twl4030_dai_voice_ops = { @@ -1997,6 +2082,7 @@ static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, + .set_tristate = twl4030_voice_set_tristate, }; struct snd_soc_dai twl4030_dai[] = { diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index fe5f395d9e4f..2b4bfa23f985 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -274,6 +274,8 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; unsigned int sysclk; + unsigned int hs_extmute:1; + void (*set_hs_extmute)(int mute); }; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 269b108e1de6..c33b92edbded 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -163,7 +163,7 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) else mute_reg &= ~(1<<2); - uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + uda134x_write(codec, UDA134X_DATA010, mute_reg); return 0; } diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 5b21594e0e58..92ec03442154 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -5,9 +5,7 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. * - * Copyright (c) 2007 Philipp Zabel <philipp.zabel@gmail.com> - * Improved support for DAPM and audio routing/mixing capabilities, - * added TLV support. + * Copyright (c) 2007-2009 Philipp Zabel <philipp.zabel@gmail.com> * * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC * codec model. @@ -19,26 +17,32 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/types.h> -#include <linux/string.h> #include <linux/slab.h> #include <linux/errno.h> -#include <linux/ioctl.h> +#include <linux/gpio.h> #include <linux/delay.h> #include <linux/i2c.h> #include <linux/workqueue.h> #include <sound/core.h> #include <sound/control.h> #include <sound/initval.h> -#include <sound/info.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include <sound/uda1380.h> #include "uda1380.h" -static struct work_struct uda1380_work; static struct snd_soc_codec *uda1380_codec; +/* codec private data */ +struct uda1380_priv { + struct snd_soc_codec codec; + u16 reg_cache[UDA1380_CACHEREGNUM]; + unsigned int dac_clk; + struct work_struct work; +}; + /* * uda1380 register cache */ @@ -473,6 +477,7 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; + struct uda1380_priv *uda1380 = codec->private_data; int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); switch (cmd) { @@ -480,13 +485,13 @@ static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: uda1380_write_reg_cache(codec, UDA1380_MIXER, mixer & ~R14_SILENCE); - schedule_work(&uda1380_work); + schedule_work(&uda1380->work); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: uda1380_write_reg_cache(codec, UDA1380_MIXER, mixer | R14_SILENCE); - schedule_work(&uda1380_work); + schedule_work(&uda1380->work); break; } return 0; @@ -670,44 +675,33 @@ static int uda1380_resume(struct platform_device *pdev) return 0; } -/* - * initialise the UDA1380 driver - * register mixer and dsp interfaces with the kernel - */ -static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) +static int uda1380_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda1380_platform_data *pdata; int ret = 0; - codec->name = "UDA1380"; - codec->owner = THIS_MODULE; - codec->read = uda1380_read_reg_cache; - codec->write = uda1380_write; - codec->set_bias_level = uda1380_set_bias_level; - codec->dai = uda1380_dai; - codec->num_dai = ARRAY_SIZE(uda1380_dai); - codec->reg_cache = kmemdup(uda1380_reg, sizeof(uda1380_reg), - GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; - codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); - codec->reg_cache_step = 1; - uda1380_reset(codec); + if (uda1380_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - uda1380_codec = codec; - INIT_WORK(&uda1380_work, uda1380_flush_work); + socdev->card->codec = uda1380_codec; + codec = uda1380_codec; + pdata = codec->dev->platform_data; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - pr_err("uda1380: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } /* power on device */ uda1380_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* set clock input */ - switch (dac_clk) { + switch (pdata->dac_clk) { case UDA1380_DAC_CLK_SYSCLK: uda1380_write(codec, UDA1380_CLK, 0); break; @@ -716,13 +710,12 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) break; } - /* uda1380 init */ snd_soc_add_controls(codec, uda1380_snd_controls, ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { - pr_err("uda1380: failed to register card\n"); + dev_err(codec->dev, "failed to register card: %d\n", ret); goto card_err; } @@ -732,165 +725,201 @@ card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); pcm_err: - kfree(codec->reg_cache); return ret; } -static struct snd_soc_device *uda1380_socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -static int uda1380_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +/* power down chip */ +static int uda1380_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = uda1380_socdev; - struct uda1380_setup_data *setup = socdev->codec_data; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int ret; - - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - ret = uda1380_init(socdev, setup->dac_clk); - if (ret < 0) - pr_err("uda1380: failed to initialise UDA1380\n"); + if (codec->control_data) + uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); - return ret; -} + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); -static int uda1380_i2c_remove(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); return 0; } -static const struct i2c_device_id uda1380_i2c_id[] = { - { "uda1380", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); - -static struct i2c_driver uda1380_i2c_driver = { - .driver = { - .name = "UDA1380 I2C Codec", - .owner = THIS_MODULE, - }, - .probe = uda1380_i2c_probe, - .remove = uda1380_i2c_remove, - .id_table = uda1380_i2c_id, +struct snd_soc_codec_device soc_codec_dev_uda1380 = { + .probe = uda1380_probe, + .remove = uda1380_remove, + .suspend = uda1380_suspend, + .resume = uda1380_resume, }; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); -static int uda1380_add_i2c_device(struct platform_device *pdev, - const struct uda1380_setup_data *setup) +static int uda1380_register(struct uda1380_priv *uda1380) { - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; + int ret, i; + struct snd_soc_codec *codec = &uda1380->codec; + struct uda1380_platform_data *pdata = codec->dev->platform_data; - ret = i2c_add_driver(&uda1380_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; + if (uda1380_codec) { + dev_err(codec->dev, "Another UDA1380 is registered\n"); + return -EINVAL; + } + + if (!pdata || !pdata->gpio_power || !pdata->gpio_reset) + return -EINVAL; + + ret = gpio_request(pdata->gpio_power, "uda1380 power"); + if (ret) + goto err_out; + ret = gpio_request(pdata->gpio_reset, "uda1380 reset"); + if (ret) + goto err_gpio; + + gpio_direction_output(pdata->gpio_power, 1); + + /* we may need to have the clock running here - pH5 */ + gpio_direction_output(pdata->gpio_reset, 1); + udelay(5); + gpio_set_value(pdata->gpio_reset, 0); + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = uda1380; + codec->name = "UDA1380"; + codec->owner = THIS_MODULE; + codec->read = uda1380_read_reg_cache; + codec->write = uda1380_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = uda1380_set_bias_level; + codec->dai = uda1380_dai; + codec->num_dai = ARRAY_SIZE(uda1380_dai); + codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); + codec->reg_cache = &uda1380->reg_cache; + codec->reg_cache_step = 1; + + memcpy(codec->reg_cache, uda1380_reg, sizeof(uda1380_reg)); + + ret = uda1380_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_reset; } - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "uda1380", I2C_NAME_SIZE); + INIT_WORK(&uda1380->work, uda1380_flush_work); + + for (i = 0; i < ARRAY_SIZE(uda1380_dai); i++) + uda1380_dai[i].dev = codec->dev; - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; + uda1380_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err_reset; } - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; + ret = snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_dai; } return 0; -err_driver: - i2c_del_driver(&uda1380_i2c_driver); - return -ENODEV; +err_dai: + snd_soc_unregister_codec(codec); +err_reset: + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_reset); +err_gpio: + gpio_free(pdata->gpio_power); +err_out: + return ret; } -#endif -static int uda1380_probe(struct platform_device *pdev) +static void uda1380_unregister(struct uda1380_priv *uda1380) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct uda1380_setup_data *setup; + struct snd_soc_codec *codec = &uda1380->codec; + struct uda1380_platform_data *pdata = codec->dev->platform_data; + + snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); + snd_soc_unregister_codec(&uda1380->codec); + + gpio_set_value(pdata->gpio_power, 0); + gpio_free(pdata->gpio_reset); + gpio_free(pdata->gpio_power); + + kfree(uda1380); + uda1380_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int uda1380_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct uda1380_priv *uda1380; struct snd_soc_codec *codec; int ret; - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) + uda1380 = kzalloc(sizeof(struct uda1380_priv), GFP_KERNEL); + if (uda1380 == NULL) return -ENOMEM; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + codec = &uda1380->codec; + codec->hw_write = (hw_write_t)i2c_master_send; - uda1380_socdev = socdev; - ret = -ENODEV; + i2c_set_clientdata(i2c, uda1380); + codec->control_data = i2c; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = uda1380_add_i2c_device(pdev, setup); - } -#endif + codec->dev = &i2c->dev; + ret = uda1380_register(uda1380); if (ret != 0) - kfree(codec); + kfree(uda1380); + return ret; } -/* power down chip */ -static int uda1380_remove(struct platform_device *pdev) +static int __devexit uda1380_i2c_remove(struct i2c_client *i2c) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); - - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&uda1380_i2c_driver); -#endif - kfree(codec); - + struct uda1380_priv *uda1380 = i2c_get_clientdata(i2c); + uda1380_unregister(uda1380); return 0; } -struct snd_soc_codec_device soc_codec_dev_uda1380 = { - .probe = uda1380_probe, - .remove = uda1380_remove, - .suspend = uda1380_suspend, - .resume = uda1380_resume, +static const struct i2c_device_id uda1380_i2c_id[] = { + { "uda1380", 0 }, + { } }; -EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); +MODULE_DEVICE_TABLE(i2c, uda1380_i2c_id); + +static struct i2c_driver uda1380_i2c_driver = { + .driver = { + .name = "UDA1380 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = uda1380_i2c_probe, + .remove = __devexit_p(uda1380_i2c_remove), + .id_table = uda1380_i2c_id, +}; +#endif static int __init uda1380_modinit(void) { - return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&uda1380_i2c_driver); + if (ret != 0) + pr_err("Failed to register UDA1380 I2C driver: %d\n", ret); +#endif + return 0; } module_init(uda1380_modinit); static void __exit uda1380_exit(void) { - snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&uda1380_i2c_driver); +#endif } module_exit(uda1380_exit); diff --git a/sound/soc/codecs/uda1380.h b/sound/soc/codecs/uda1380.h index c55c17a52a12..9cefa8a54770 100644 --- a/sound/soc/codecs/uda1380.h +++ b/sound/soc/codecs/uda1380.h @@ -72,14 +72,6 @@ #define R22_SKIP_DCFIL 0x0002 #define R23_AGC_EN 0x0001 -struct uda1380_setup_data { - int i2c_bus; - unsigned short i2c_address; - int dac_clk; -#define UDA1380_DAC_CLK_SYSCLK 0 -#define UDA1380_DAC_CLK_WSPLL 1 -}; - #define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */ #define UDA1380_DAI_PLAYBACK 1 /* playback DAI */ #define UDA1380_DAI_CAPTURE 2 /* capture DAI */ diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e7348d341b76..3ff0373dff89 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -63,6 +63,8 @@ struct wm8350_data { struct wm8350_jack_data hpl; struct wm8350_jack_data hpr; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; + int fll_freq_out; + int fll_freq_in; }; static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec, @@ -406,7 +408,6 @@ static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" }; static const char *wm8350_dacmutem[] = { "Normal", "Soft" }; static const char *wm8350_dacmutes[] = { "Fast", "Slow" }; -static const char *wm8350_dacfilter[] = { "Normal", "Sloping" }; static const char *wm8350_adcfilter[] = { "None", "High Pass" }; static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" }; static const char *wm8350_lr[] = { "Left", "Right" }; @@ -416,7 +417,6 @@ static const struct soc_enum wm8350_enum[] = { SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol), SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem), SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes), - SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter), SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter), SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp), SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol), @@ -444,10 +444,9 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = { 0, 255, 0, dac_pcm_tlv), SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]), SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]), - SOC_ENUM("Playback PCM Filter", wm8350_enum[4]), - SOC_ENUM("Capture PCM Filter", wm8350_enum[5]), - SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]), - SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]), + SOC_ENUM("Capture PCM Filter", wm8350_enum[4]), + SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]), + SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]), SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume", WM8350_ADC_DIGITAL_VOLUME_L, WM8350_ADC_DIGITAL_VOLUME_R, @@ -613,7 +612,7 @@ SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1); /* Out4 Capture Mux */ static const struct snd_kcontrol_new wm8350_out4_capture_controls = -SOC_DAPM_ENUM("Route", wm8350_enum[8]); +SOC_DAPM_ENUM("Route", wm8350_enum[7]); static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = { @@ -993,6 +992,7 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai) { struct snd_soc_codec *codec = codec_dai->codec; + struct wm8350 *wm8350 = codec->control_data; u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & ~WM8350_AIF_WL_MASK; @@ -1012,6 +1012,19 @@ static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, } wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + + /* The sloping stopband filter is recommended for use with + * lower sample rates to improve performance. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (params_rate(params) < 24000) + wm8350_set_bits(wm8350, WM8350_DAC_MUTE_VOLUME, + WM8350_DAC_SB_FILT); + else + wm8350_clear_bits(wm8350, WM8350_DAC_MUTE_VOLUME, + WM8350_DAC_SB_FILT); + } + return 0; } @@ -1093,10 +1106,14 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *priv = codec->private_data; struct _fll_div fll_div; int ret = 0; u16 fll_1, fll_4; + if (freq_in == priv->fll_freq_in && freq_out == priv->fll_freq_out) + return 0; + /* power down FLL - we need to do this for reconfiguration */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA | WM8350_FLL_OSC_ENA); @@ -1131,6 +1148,9 @@ static int wm8350_set_fll(struct snd_soc_dai *codec_dai, wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA); wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA); + priv->fll_freq_out = freq_out; + priv->fll_freq_in = freq_in; + return 0; } @@ -1660,6 +1680,21 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM +static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int wm8350_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define wm8350_codec_suspend NULL +#define wm8350_codec_resume NULL +#endif + static struct platform_driver wm8350_codec_driver = { .driver = { .name = "wm8350-codec", @@ -1667,6 +1702,8 @@ static struct platform_driver wm8350_codec_driver = { }, .probe = wm8350_codec_probe, .remove = __devexit_p(wm8350_codec_remove), + .suspend = wm8350_codec_suspend, + .resume = wm8350_codec_resume, }; static __init int wm8350_init(void) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 502eefac1ecd..b9ef4d915221 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1022,10 +1022,15 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out) return 0; - if (freq_out != 0) { + if (freq_out) { ret = fll_factors(wm8400, &factors, freq_in, freq_out); if (ret != 0) return ret; + } else { + /* Bodge GCC 4.4.0 uninitialised variable warning - it + * doesn't seem capable of working out that we exit if + * freq_out is 0 before any of the uses. */ + memset(&factors, 0, sizeof(factors)); } wm8400->fll_out = freq_out; @@ -1040,7 +1045,7 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, reg &= ~WM8400_FLL_OSC_ENA; wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); - if (freq_out == 0) + if (!freq_out) return 0; reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); @@ -1553,6 +1558,21 @@ static int __exit wm8400_codec_remove(struct platform_device *dev) return 0; } +#ifdef CONFIG_PM +static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int wm8400_pdev_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define wm8400_pdev_suspend NULL +#define wm8400_pdev_resume NULL +#endif + static struct platform_driver wm8400_codec_driver = { .driver = { .name = "wm8400-codec", @@ -1560,6 +1580,8 @@ static struct platform_driver wm8400_codec_driver = { }, .probe = wm8400_codec_probe, .remove = __exit_p(wm8400_codec_remove), + .suspend = wm8400_pdev_suspend, + .resume = wm8400_pdev_resume, }; static int __init wm8400_codec_init(void) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index c8b8dba85890..060d5d06ba95 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -58,55 +58,7 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { #define WM8510_POWER1_BIASEN 0x08 #define WM8510_POWER1_BUFIOEN 0x10 -/* - * read wm8510 register cache - */ -static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8510_RESET) - return 0; - if (reg >= WM8510_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8510 register cache - */ -static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8510_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8510 register space - */ -static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8510 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8510_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) +#define wm8510_reset(c) snd_soc_write(c, WM8510_RESET, 0) static const char *wm8510_companding[] = { "Off", "NC", "u-law", "A-law" }; static const char *wm8510_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; @@ -327,27 +279,27 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, if (freq_in == 0 || freq_out == 0) { /* Clock CODEC directly from MCLK */ - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); - wm8510_write(codec, WM8510_CLOCK, reg & 0x0ff); + reg = snd_soc_read(codec, WM8510_CLOCK); + snd_soc_write(codec, WM8510_CLOCK, reg & 0x0ff); /* Turn off PLL */ - reg = wm8510_read_reg_cache(codec, WM8510_POWER1); - wm8510_write(codec, WM8510_POWER1, reg & 0x1df); + reg = snd_soc_read(codec, WM8510_POWER1); + snd_soc_write(codec, WM8510_POWER1, reg & 0x1df); return 0; } pll_factors(freq_out*4, freq_in); - wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); - wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); - wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); - wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); - reg = wm8510_read_reg_cache(codec, WM8510_POWER1); - wm8510_write(codec, WM8510_POWER1, reg | 0x020); + snd_soc_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); + snd_soc_write(codec, WM8510_PLLK1, pll_div.k >> 18); + snd_soc_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); + reg = snd_soc_read(codec, WM8510_POWER1); + snd_soc_write(codec, WM8510_POWER1, reg | 0x020); /* Run CODEC from PLL instead of MCLK */ - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK); - wm8510_write(codec, WM8510_CLOCK, reg | 0x100); + reg = snd_soc_read(codec, WM8510_CLOCK); + snd_soc_write(codec, WM8510_CLOCK, reg | 0x100); return 0; } @@ -363,24 +315,24 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8510_OPCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; - wm8510_write(codec, WM8510_GPIO, reg | div); + reg = snd_soc_read(codec, WM8510_GPIO) & 0x1cf; + snd_soc_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f; - wm8510_write(codec, WM8510_CLOCK, reg | div); + reg = snd_soc_read(codec, WM8510_CLOCK) & 0x11f; + snd_soc_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: - reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; - wm8510_write(codec, WM8510_ADC, reg | div); + reg = snd_soc_read(codec, WM8510_ADC) & 0x1f7; + snd_soc_write(codec, WM8510_ADC, reg | div); break; case WM8510_DACCLK: - reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; - wm8510_write(codec, WM8510_DAC, reg | div); + reg = snd_soc_read(codec, WM8510_DAC) & 0x1f7; + snd_soc_write(codec, WM8510_DAC, reg | div); break; case WM8510_BCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; - wm8510_write(codec, WM8510_CLOCK, reg | div); + reg = snd_soc_read(codec, WM8510_CLOCK) & 0x1e3; + snd_soc_write(codec, WM8510_CLOCK, reg | div); break; default: return -EINVAL; @@ -394,7 +346,7 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; - u16 clk = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1fe; + u16 clk = snd_soc_read(codec, WM8510_CLOCK) & 0x1fe; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -441,8 +393,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8510_write(codec, WM8510_IFACE, iface); - wm8510_write(codec, WM8510_CLOCK, clk); + snd_soc_write(codec, WM8510_IFACE, iface); + snd_soc_write(codec, WM8510_CLOCK, clk); return 0; } @@ -453,8 +405,8 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f; - u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; + u16 iface = snd_soc_read(codec, WM8510_IFACE) & 0x19f; + u16 adn = snd_soc_read(codec, WM8510_ADD) & 0x1f1; /* bit size */ switch (params_format(params)) { @@ -493,20 +445,20 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, break; } - wm8510_write(codec, WM8510_IFACE, iface); - wm8510_write(codec, WM8510_ADD, adn); + snd_soc_write(codec, WM8510_IFACE, iface); + snd_soc_write(codec, WM8510_ADD, adn); return 0; } static int wm8510_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; + u16 mute_reg = snd_soc_read(codec, WM8510_DAC) & 0xffbf; if (mute) - wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); + snd_soc_write(codec, WM8510_DAC, mute_reg | 0x40); else - wm8510_write(codec, WM8510_DAC, mute_reg); + snd_soc_write(codec, WM8510_DAC, mute_reg); return 0; } @@ -514,13 +466,13 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute) static int wm8510_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3; + u16 power1 = snd_soc_read(codec, WM8510_POWER1) & ~0x3; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: power1 |= 0x1; /* VMID 50k */ - wm8510_write(codec, WM8510_POWER1, power1); + snd_soc_write(codec, WM8510_POWER1, power1); break; case SND_SOC_BIAS_STANDBY: @@ -528,18 +480,18 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Initial cap charge at VMID 5k */ - wm8510_write(codec, WM8510_POWER1, power1 | 0x3); + snd_soc_write(codec, WM8510_POWER1, power1 | 0x3); mdelay(100); } power1 |= 0x2; /* VMID 500k */ - wm8510_write(codec, WM8510_POWER1, power1); + snd_soc_write(codec, WM8510_POWER1, power1); break; case SND_SOC_BIAS_OFF: - wm8510_write(codec, WM8510_POWER1, 0); - wm8510_write(codec, WM8510_POWER2, 0); - wm8510_write(codec, WM8510_POWER3, 0); + snd_soc_write(codec, WM8510_POWER1, 0); + snd_soc_write(codec, WM8510_POWER2, 0); + snd_soc_write(codec, WM8510_POWER3, 0); break; } @@ -577,6 +529,7 @@ struct snd_soc_dai wm8510_dai = { .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, .ops = &wm8510_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(wm8510_dai); @@ -612,15 +565,14 @@ static int wm8510_resume(struct platform_device *pdev) * initialise the WM8510 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8510_init(struct snd_soc_device *socdev) +static int wm8510_init(struct snd_soc_device *socdev, + enum snd_soc_control_type control) { struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8510"; codec->owner = THIS_MODULE; - codec->read = wm8510_read_reg_cache; - codec->write = wm8510_write; codec->set_bias_level = wm8510_set_bias_level; codec->dai = &wm8510_dai; codec->num_dai = 1; @@ -630,13 +582,20 @@ static int wm8510_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", + ret); + goto err; + } + wm8510_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8510: failed to create pcms\n"); - goto pcm_err; + goto err; } /* power on device */ @@ -655,7 +614,7 @@ static int wm8510_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } @@ -678,7 +637,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = wm8510_init(socdev); + ret = wm8510_init(socdev, SND_SOC_I2C); if (ret < 0) pr_err("failed to initialise WM8510\n"); @@ -758,7 +717,7 @@ static int __devinit wm8510_spi_probe(struct spi_device *spi) codec->control_data = spi; - ret = wm8510_init(socdev); + ret = wm8510_init(socdev, SND_SOC_SPI); if (ret < 0) dev_err(&spi->dev, "failed to initialise WM8510\n"); @@ -779,30 +738,6 @@ static struct spi_driver wm8510_spi_driver = { .probe = wm8510_spi_probe, .remove = __devexit_p(wm8510_spi_remove), }; - -static int wm8510_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} #endif /* CONFIG_SPI_MASTER */ static int wm8510_probe(struct platform_device *pdev) @@ -827,13 +762,11 @@ static int wm8510_probe(struct platform_device *pdev) wm8510_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8510_add_i2c_device(pdev, setup); } #endif #if defined(CONFIG_SPI_MASTER) if (setup->spi) { - codec->hw_write = (hw_write_t)wm8510_spi_write; ret = spi_register_driver(&wm8510_spi_driver); if (ret != 0) printk(KERN_ERR "can't add spi driver"); diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c new file mode 100644 index 000000000000..25870a4652fb --- /dev/null +++ b/sound/soc/codecs/wm8523.c @@ -0,0 +1,699 @@ +/* + * wm8523.c -- WM8523 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8523.h" + +static struct snd_soc_codec *wm8523_codec; +struct snd_soc_codec_device soc_codec_dev_wm8523; + +#define WM8523_NUM_SUPPLIES 2 +static const char *wm8523_supply_names[WM8523_NUM_SUPPLIES] = { + "AVDD", + "LINEVDD", +}; + +#define WM8523_NUM_RATES 7 + +/* codec private data */ +struct wm8523_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8523_REGISTER_COUNT]; + struct regulator_bulk_data supplies[WM8523_NUM_SUPPLIES]; + unsigned int sysclk; + unsigned int rate_constraint_list[WM8523_NUM_RATES]; + struct snd_pcm_hw_constraint_list rate_constraint; +}; + +static const u16 wm8523_reg[WM8523_REGISTER_COUNT] = { + 0x8523, /* R0 - DEVICE_ID */ + 0x0001, /* R1 - REVISION */ + 0x0000, /* R2 - PSCTRL1 */ + 0x1812, /* R3 - AIF_CTRL1 */ + 0x0000, /* R4 - AIF_CTRL2 */ + 0x0001, /* R5 - DAC_CTRL3 */ + 0x0190, /* R6 - DAC_GAINL */ + 0x0190, /* R7 - DAC_GAINR */ + 0x0000, /* R8 - ZERO_DETECT */ +}; + +static int wm8523_volatile_register(unsigned int reg) +{ + switch (reg) { + case WM8523_DEVICE_ID: + case WM8523_REVISION: + return 1; + default: + return 0; + } +} + +static int wm8523_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8523_DEVICE_ID, 0); +} + +static const DECLARE_TLV_DB_SCALE(dac_tlv, -10000, 25, 0); + +static const char *wm8523_zd_count_text[] = { + "1024", + "2048", +}; + +static const struct soc_enum wm8523_zc_count = + SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); + +static const struct snd_kcontrol_new wm8523_snd_controls[] = { +SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, + 0, 448, 0, dac_tlv), +SOC_SINGLE("ZC Switch", WM8523_DAC_CTRL3, 4, 1, 0), +SOC_SINGLE("Playback Deemphasis Switch", WM8523_AIF_CTRL1, 8, 1, 0), +SOC_DOUBLE("Playback Switch", WM8523_DAC_CTRL3, 2, 3, 1, 1), +SOC_SINGLE("Volume Ramp Up Switch", WM8523_DAC_CTRL3, 1, 1, 0), +SOC_SINGLE("Volume Ramp Down Switch", WM8523_DAC_CTRL3, 0, 1, 0), +SOC_ENUM("Zero Detect Count", wm8523_zc_count), +}; + +static const struct snd_soc_dapm_widget wm8523_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LINEVOUTL"), +SND_SOC_DAPM_OUTPUT("LINEVOUTR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + { "LINEVOUTL", NULL, "DAC" }, + { "LINEVOUTR", NULL, "DAC" }, +}; + +static int wm8523_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8523_dapm_widgets, + ARRAY_SIZE(wm8523_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static struct { + int value; + int ratio; +} lrclk_ratios[WM8523_NUM_RATES] = { + { 1, 128 }, + { 2, 192 }, + { 3, 256 }, + { 4, 384 }, + { 5, 512 }, + { 6, 768 }, + { 7, 1152 }, +}; + +static int wm8523_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8523_priv *wm8523 = codec->private_data; + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8523->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + return 0; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &wm8523->rate_constraint); + + return 0; +} + +static int wm8523_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8523_priv *wm8523 = codec->private_data; + int i; + u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); + u16 aifctrl2 = snd_soc_read(codec, WM8523_AIF_CTRL2); + + /* Find a supported LRCLK ratio */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (wm8523->sysclk / params_rate(params) == + lrclk_ratios[i].ratio) + break; + } + + /* Should never happen, should be handled by constraints */ + if (i == ARRAY_SIZE(lrclk_ratios)) { + dev_err(codec->dev, "MCLK/fs ratio %d unsupported\n", + wm8523->sysclk / params_rate(params)); + return -EINVAL; + } + + aifctrl2 &= ~WM8523_SR_MASK; + aifctrl2 |= lrclk_ratios[i].value; + + aifctrl1 &= ~WM8523_WL_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aifctrl1 |= 0x8; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aifctrl1 |= 0x10; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aifctrl1 |= 0x18; + break; + } + + snd_soc_write(codec, WM8523_AIF_CTRL1, aifctrl1); + snd_soc_write(codec, WM8523_AIF_CTRL2, aifctrl2); + + return 0; +} + +static int wm8523_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8523_priv *wm8523 = codec->private_data; + unsigned int val; + int i; + + wm8523->sysclk = freq; + + wm8523->rate_constraint.count = 0; + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + val = freq / lrclk_ratios[i].ratio; + /* Check that it's a standard rate since core can't + * cope with others and having the odd rates confuses + * constraint matching. + */ + switch (val) { + case 8000: + case 11025: + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 64000: + case 88200: + case 96000: + case 176400: + case 192000: + dev_dbg(codec->dev, "Supported sample rate: %dHz\n", + val); + wm8523->rate_constraint_list[i] = val; + wm8523->rate_constraint.count++; + break; + default: + dev_dbg(codec->dev, "Skipping sample rate: %dHz\n", + val); + } + } + + /* Need at least one supported rate... */ + if (wm8523->rate_constraint.count == 0) + return -EINVAL; + + return 0; +} + + +static int wm8523_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 aifctrl1 = snd_soc_read(codec, WM8523_AIF_CTRL1); + + aifctrl1 &= ~(WM8523_BCLK_INV_MASK | WM8523_LRCLK_INV_MASK | + WM8523_FMT_MASK | WM8523_AIF_MSTR_MASK); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aifctrl1 |= WM8523_AIF_MSTR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aifctrl1 |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aifctrl1 |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + aifctrl1 |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + aifctrl1 |= 0x0023; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aifctrl1 |= WM8523_BCLK_INV | WM8523_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aifctrl1 |= WM8523_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aifctrl1 |= WM8523_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8523_AIF_CTRL1, aifctrl1); + + return 0; +} + +static int wm8523_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8523_priv *wm8523 = codec->private_data; + int ret, i; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 3); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + + /* Initial power up */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 1); + + /* Sync back default/cached values */ + for (i = WM8523_AIF_CTRL1; + i < WM8523_MAX_REGISTER; i++) + snd_soc_write(codec, i, wm8523->reg_cache[i]); + + + msleep(100); + } + + /* Power up to mute */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 2); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us. */ + snd_soc_update_bits(codec, WM8523_PSCTRL1, + WM8523_SYS_ENA_MASK, 0); + msleep(100); + + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8523_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8523_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8523_dai_ops = { + .startup = wm8523_startup, + .hw_params = wm8523_hw_params, + .set_sysclk = wm8523_set_dai_sysclk, + .set_fmt = wm8523_set_dai_fmt, +}; + +struct snd_soc_dai wm8523_dai = { + .name = "WM8523", + .playback = { + .stream_name = "Playback", + .channels_min = 2, /* Mono modes not yet supported */ + .channels_max = 2, + .rates = WM8523_RATES, + .formats = WM8523_FORMATS, + }, + .ops = &wm8523_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8523_dai); + +#ifdef CONFIG_PM +static int wm8523_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8523_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8523_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8523_suspend NULL +#define wm8523_resume NULL +#endif + +static int wm8523_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8523_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8523_codec; + codec = wm8523_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8523_snd_controls, + ARRAY_SIZE(wm8523_snd_controls)); + wm8523_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8523_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8523 = { + .probe = wm8523_probe, + .remove = wm8523_remove, + .suspend = wm8523_suspend, + .resume = wm8523_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8523); + +static int wm8523_register(struct wm8523_priv *wm8523, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8523->codec; + int i; + + if (wm8523_codec) { + dev_err(codec->dev, "Another WM8523 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8523; + codec->name = "WM8523"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8523_set_bias_level; + codec->dai = &wm8523_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8523_REGISTER_COUNT; + codec->reg_cache = &wm8523->reg_cache; + codec->volatile_register = wm8523_volatile_register; + + wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; + wm8523->rate_constraint.count = + ARRAY_SIZE(wm8523->rate_constraint_list); + + memcpy(codec->reg_cache, wm8523_reg, sizeof(wm8523_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8523->supplies); i++) + wm8523->supplies[i].supply = wm8523_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8523->supplies), + wm8523->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = snd_soc_read(codec, WM8523_DEVICE_ID); + if (ret < 0) { + dev_err(codec->dev, "Failed to read ID register\n"); + goto err_enable; + } + if (ret != wm8523_reg[WM8523_DEVICE_ID]) { + dev_err(codec->dev, "Device is not a WM8523, ID is %x\n", ret); + ret = -EINVAL; + goto err_enable; + } + + ret = snd_soc_read(codec, WM8523_REVISION); + if (ret < 0) { + dev_err(codec->dev, "Failed to read revision register\n"); + goto err_enable; + } + dev_info(codec->dev, "revision %c\n", + (ret & WM8523_CHIP_REV_MASK) + 'A'); + + ret = wm8523_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err_enable; + } + + wm8523_dai.dev = codec->dev; + + /* Change some default settings - latch VU and enable ZC */ + wm8523->reg_cache[WM8523_DAC_GAINR] |= WM8523_DACR_VU; + wm8523->reg_cache[WM8523_DAC_CTRL3] |= WM8523_ZC; + + wm8523_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); + + wm8523_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8523_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); +err: + kfree(wm8523); + return ret; +} + +static void wm8523_unregister(struct wm8523_priv *wm8523) +{ + wm8523_set_bias_level(&wm8523->codec, SND_SOC_BIAS_OFF); + regulator_bulk_free(ARRAY_SIZE(wm8523->supplies), wm8523->supplies); + snd_soc_unregister_dai(&wm8523_dai); + snd_soc_unregister_codec(&wm8523->codec); + kfree(wm8523); + wm8523_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8523_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8523_priv *wm8523; + struct snd_soc_codec *codec; + + wm8523 = kzalloc(sizeof(struct wm8523_priv), GFP_KERNEL); + if (wm8523 == NULL) + return -ENOMEM; + + codec = &wm8523->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8523); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8523_register(wm8523, SND_SOC_I2C); +} + +static __devexit int wm8523_i2c_remove(struct i2c_client *client) +{ + struct wm8523_priv *wm8523 = i2c_get_clientdata(client); + wm8523_unregister(wm8523); + return 0; +} + +#ifdef CONFIG_PM +static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8523_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8523_i2c_suspend NULL +#define wm8523_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8523_i2c_id[] = { + { "wm8523", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8523_i2c_id); + +static struct i2c_driver wm8523_i2c_driver = { + .driver = { + .name = "WM8523", + .owner = THIS_MODULE, + }, + .probe = wm8523_i2c_probe, + .remove = __devexit_p(wm8523_i2c_remove), + .suspend = wm8523_i2c_suspend, + .resume = wm8523_i2c_resume, + .id_table = wm8523_i2c_id, +}; +#endif + +static int __init wm8523_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8523_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8523 I2C driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8523_modinit); + +static void __exit wm8523_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8523_i2c_driver); +#endif +} +module_exit(wm8523_exit); + +MODULE_DESCRIPTION("ASoC WM8523 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8523.h b/sound/soc/codecs/wm8523.h new file mode 100644 index 000000000000..1aa9ce3e1357 --- /dev/null +++ b/sound/soc/codecs/wm8523.h @@ -0,0 +1,160 @@ +/* + * wm8523.h -- WM8423 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics, plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * Based on wm8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8523_H +#define _WM8523_H + +/* + * Register values. + */ +#define WM8523_DEVICE_ID 0x00 +#define WM8523_REVISION 0x01 +#define WM8523_PSCTRL1 0x02 +#define WM8523_AIF_CTRL1 0x03 +#define WM8523_AIF_CTRL2 0x04 +#define WM8523_DAC_CTRL3 0x05 +#define WM8523_DAC_GAINL 0x06 +#define WM8523_DAC_GAINR 0x07 +#define WM8523_ZERO_DETECT 0x08 + +#define WM8523_REGISTER_COUNT 9 +#define WM8523_MAX_REGISTER 0x08 + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - DEVICE_ID + */ +#define WM8523_CHIP_ID_MASK 0xFFFF /* CHIP_ID - [15:0] */ +#define WM8523_CHIP_ID_SHIFT 0 /* CHIP_ID - [15:0] */ +#define WM8523_CHIP_ID_WIDTH 16 /* CHIP_ID - [15:0] */ + +/* + * R1 (0x01) - REVISION + */ +#define WM8523_CHIP_REV_MASK 0x0007 /* CHIP_REV - [2:0] */ +#define WM8523_CHIP_REV_SHIFT 0 /* CHIP_REV - [2:0] */ +#define WM8523_CHIP_REV_WIDTH 3 /* CHIP_REV - [2:0] */ + +/* + * R2 (0x02) - PSCTRL1 + */ +#define WM8523_SYS_ENA_MASK 0x0003 /* SYS_ENA - [1:0] */ +#define WM8523_SYS_ENA_SHIFT 0 /* SYS_ENA - [1:0] */ +#define WM8523_SYS_ENA_WIDTH 2 /* SYS_ENA - [1:0] */ + +/* + * R3 (0x03) - AIF_CTRL1 + */ +#define WM8523_TDM_MODE_MASK 0x1800 /* TDM_MODE - [12:11] */ +#define WM8523_TDM_MODE_SHIFT 11 /* TDM_MODE - [12:11] */ +#define WM8523_TDM_MODE_WIDTH 2 /* TDM_MODE - [12:11] */ +#define WM8523_TDM_SLOT_MASK 0x0600 /* TDM_SLOT - [10:9] */ +#define WM8523_TDM_SLOT_SHIFT 9 /* TDM_SLOT - [10:9] */ +#define WM8523_TDM_SLOT_WIDTH 2 /* TDM_SLOT - [10:9] */ +#define WM8523_DEEMPH 0x0100 /* DEEMPH */ +#define WM8523_DEEMPH_MASK 0x0100 /* DEEMPH */ +#define WM8523_DEEMPH_SHIFT 8 /* DEEMPH */ +#define WM8523_DEEMPH_WIDTH 1 /* DEEMPH */ +#define WM8523_AIF_MSTR 0x0080 /* AIF_MSTR */ +#define WM8523_AIF_MSTR_MASK 0x0080 /* AIF_MSTR */ +#define WM8523_AIF_MSTR_SHIFT 7 /* AIF_MSTR */ +#define WM8523_AIF_MSTR_WIDTH 1 /* AIF_MSTR */ +#define WM8523_LRCLK_INV 0x0040 /* LRCLK_INV */ +#define WM8523_LRCLK_INV_MASK 0x0040 /* LRCLK_INV */ +#define WM8523_LRCLK_INV_SHIFT 6 /* LRCLK_INV */ +#define WM8523_LRCLK_INV_WIDTH 1 /* LRCLK_INV */ +#define WM8523_BCLK_INV 0x0020 /* BCLK_INV */ +#define WM8523_BCLK_INV_MASK 0x0020 /* BCLK_INV */ +#define WM8523_BCLK_INV_SHIFT 5 /* BCLK_INV */ +#define WM8523_BCLK_INV_WIDTH 1 /* BCLK_INV */ +#define WM8523_WL_MASK 0x0018 /* WL - [4:3] */ +#define WM8523_WL_SHIFT 3 /* WL - [4:3] */ +#define WM8523_WL_WIDTH 2 /* WL - [4:3] */ +#define WM8523_FMT_MASK 0x0007 /* FMT - [2:0] */ +#define WM8523_FMT_SHIFT 0 /* FMT - [2:0] */ +#define WM8523_FMT_WIDTH 3 /* FMT - [2:0] */ + +/* + * R4 (0x04) - AIF_CTRL2 + */ +#define WM8523_DAC_OP_MUX_MASK 0x00C0 /* DAC_OP_MUX - [7:6] */ +#define WM8523_DAC_OP_MUX_SHIFT 6 /* DAC_OP_MUX - [7:6] */ +#define WM8523_DAC_OP_MUX_WIDTH 2 /* DAC_OP_MUX - [7:6] */ +#define WM8523_BCLKDIV_MASK 0x0038 /* BCLKDIV - [5:3] */ +#define WM8523_BCLKDIV_SHIFT 3 /* BCLKDIV - [5:3] */ +#define WM8523_BCLKDIV_WIDTH 3 /* BCLKDIV - [5:3] */ +#define WM8523_SR_MASK 0x0007 /* SR - [2:0] */ +#define WM8523_SR_SHIFT 0 /* SR - [2:0] */ +#define WM8523_SR_WIDTH 3 /* SR - [2:0] */ + +/* + * R5 (0x05) - DAC_CTRL3 + */ +#define WM8523_ZC 0x0010 /* ZC */ +#define WM8523_ZC_MASK 0x0010 /* ZC */ +#define WM8523_ZC_SHIFT 4 /* ZC */ +#define WM8523_ZC_WIDTH 1 /* ZC */ +#define WM8523_DACR 0x0008 /* DACR */ +#define WM8523_DACR_MASK 0x0008 /* DACR */ +#define WM8523_DACR_SHIFT 3 /* DACR */ +#define WM8523_DACR_WIDTH 1 /* DACR */ +#define WM8523_DACL 0x0004 /* DACL */ +#define WM8523_DACL_MASK 0x0004 /* DACL */ +#define WM8523_DACL_SHIFT 2 /* DACL */ +#define WM8523_DACL_WIDTH 1 /* DACL */ +#define WM8523_VOL_UP_RAMP 0x0002 /* VOL_UP_RAMP */ +#define WM8523_VOL_UP_RAMP_MASK 0x0002 /* VOL_UP_RAMP */ +#define WM8523_VOL_UP_RAMP_SHIFT 1 /* VOL_UP_RAMP */ +#define WM8523_VOL_UP_RAMP_WIDTH 1 /* VOL_UP_RAMP */ +#define WM8523_VOL_DOWN_RAMP 0x0001 /* VOL_DOWN_RAMP */ +#define WM8523_VOL_DOWN_RAMP_MASK 0x0001 /* VOL_DOWN_RAMP */ +#define WM8523_VOL_DOWN_RAMP_SHIFT 0 /* VOL_DOWN_RAMP */ +#define WM8523_VOL_DOWN_RAMP_WIDTH 1 /* VOL_DOWN_RAMP */ + +/* + * R6 (0x06) - DAC_GAINL + */ +#define WM8523_DACL_VU 0x0200 /* DACL_VU */ +#define WM8523_DACL_VU_MASK 0x0200 /* DACL_VU */ +#define WM8523_DACL_VU_SHIFT 9 /* DACL_VU */ +#define WM8523_DACL_VU_WIDTH 1 /* DACL_VU */ +#define WM8523_DACL_VOL_MASK 0x01FF /* DACL_VOL - [8:0] */ +#define WM8523_DACL_VOL_SHIFT 0 /* DACL_VOL - [8:0] */ +#define WM8523_DACL_VOL_WIDTH 9 /* DACL_VOL - [8:0] */ + +/* + * R7 (0x07) - DAC_GAINR + */ +#define WM8523_DACR_VU 0x0200 /* DACR_VU */ +#define WM8523_DACR_VU_MASK 0x0200 /* DACR_VU */ +#define WM8523_DACR_VU_SHIFT 9 /* DACR_VU */ +#define WM8523_DACR_VU_WIDTH 1 /* DACR_VU */ +#define WM8523_DACR_VOL_MASK 0x01FF /* DACR_VOL - [8:0] */ +#define WM8523_DACR_VOL_SHIFT 0 /* DACR_VOL - [8:0] */ +#define WM8523_DACR_VOL_WIDTH 9 /* DACR_VOL - [8:0] */ + +/* + * R8 (0x08) - ZERO_DETECT + */ +#define WM8523_ZD_COUNT_MASK 0x0003 /* ZD_COUNT - [1:0] */ +#define WM8523_ZD_COUNT_SHIFT 0 /* ZD_COUNT - [1:0] */ +#define WM8523_ZD_COUNT_WIDTH 2 /* ZD_COUNT - [1:0] */ + +extern struct snd_soc_dai wm8523_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8523; + +#endif diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 86c4b24db817..6bded8c78150 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -24,6 +24,8 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regulator/consumer.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -187,82 +189,22 @@ struct pll_state { unsigned int out; }; +#define WM8580_NUM_SUPPLIES 3 +static const char *wm8580_supply_names[WM8580_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "PVDD", +}; + /* codec private data */ struct wm8580_priv { struct snd_soc_codec codec; + struct regulator_bulk_data supplies[WM8580_NUM_SUPPLIES]; u16 reg_cache[WM8580_MAX_REGISTER + 1]; struct pll_state a; struct pll_state b; }; - -/* - * read wm8580 register cache - */ -static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); - return cache[reg]; -} - -/* - * write wm8580 register cache - */ -static inline void wm8580_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - - cache[reg] = value; -} - -/* - * write to the WM8580 register space - */ -static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); - - /* Registers are 9 bits wide */ - value &= 0x1ff; - - switch (reg) { - case WM8580_RESET: - /* Uncached */ - break; - default: - if (value == wm8580_read_reg_cache(codec, reg)) - return 0; - } - - /* data is - * D15..D9 WM8580 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8580_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -static inline unsigned int wm8580_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - switch (reg) { - default: - return wm8580_read_reg_cache(codec, reg); - } -} - static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static int wm8580_out_vu(struct snd_kcontrol *kcontrol, @@ -271,25 +213,22 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + u16 *reg_cache = codec->reg_cache; unsigned int reg = mc->reg; unsigned int reg2 = mc->rreg; int ret; - u16 val; /* Clear the register cache so we write without VU set */ - wm8580_write_reg_cache(codec, reg, 0); - wm8580_write_reg_cache(codec, reg2, 0); + reg_cache[reg] = 0; + reg_cache[reg2] = 0; ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); if (ret < 0) return ret; /* Now write again with the volume update bit set */ - val = wm8580_read_reg_cache(codec, reg); - wm8580_write(codec, reg, val | 0x0100); - - val = wm8580_read_reg_cache(codec, reg2); - wm8580_write(codec, reg2, val | 0x0100); + snd_soc_update_bits(codec, reg, 0x100, 0x100); + snd_soc_update_bits(codec, reg2, 0x100, 0x100); return 0; } @@ -512,27 +451,27 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, /* Always disable the PLL - it is not safe to leave it running * while reprogramming it. */ - reg = wm8580_read(codec, WM8580_PWRDN2); - wm8580_write(codec, WM8580_PWRDN2, reg | pwr_mask); + reg = snd_soc_read(codec, WM8580_PWRDN2); + snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask); if (!freq_in || !freq_out) return 0; - wm8580_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff); - wm8580_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0xff); - wm8580_write(codec, WM8580_PLLA3 + offset, + snd_soc_write(codec, WM8580_PLLA1 + offset, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8580_PLLA2 + offset, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8580_PLLA3 + offset, (pll_div.k >> 18 & 0xf) | (pll_div.n << 4)); - reg = wm8580_read(codec, WM8580_PLLA4 + offset); - reg &= ~0x3f; + reg = snd_soc_read(codec, WM8580_PLLA4 + offset); + reg &= ~0x1b; reg |= pll_div.prescale | pll_div.postscale << 1 | pll_div.freqmode << 3; - wm8580_write(codec, WM8580_PLLA4 + offset, reg); + snd_soc_write(codec, WM8580_PLLA4 + offset, reg); /* All done, turn it on */ - reg = wm8580_read(codec, WM8580_PWRDN2); - wm8580_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); + reg = snd_soc_read(codec, WM8580_PWRDN2); + snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask); return 0; } @@ -547,7 +486,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id); + u16 paifb = snd_soc_read(codec, WM8580_PAIF3 + dai->id); paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ @@ -567,7 +506,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb); + snd_soc_write(codec, WM8580_PAIF3 + dai->id, paifb); return 0; } @@ -579,8 +518,8 @@ static int wm8580_set_paif_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int aifb; int can_invert_lrclk; - aifa = wm8580_read(codec, WM8580_PAIF1 + codec_dai->id); - aifb = wm8580_read(codec, WM8580_PAIF3 + codec_dai->id); + aifa = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + aifb = snd_soc_read(codec, WM8580_PAIF3 + codec_dai->id); aifb &= ~(WM8580_AIF_FMT_MASK | WM8580_AIF_LRP | WM8580_AIF_BCP); @@ -646,8 +585,8 @@ static int wm8580_set_paif_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8580_write(codec, WM8580_PAIF1 + codec_dai->id, aifa); - wm8580_write(codec, WM8580_PAIF3 + codec_dai->id, aifb); + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, aifa); + snd_soc_write(codec, WM8580_PAIF3 + codec_dai->id, aifb); return 0; } @@ -660,7 +599,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8580_MCLK: - reg = wm8580_read(codec, WM8580_PLLB4); + reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_MCLKOUTSRC_MASK; switch (div) { @@ -682,11 +621,11 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, default: return -EINVAL; } - wm8580_write(codec, WM8580_PLLB4, reg); + snd_soc_write(codec, WM8580_PLLB4, reg); break; case WM8580_DAC_CLKSEL: - reg = wm8580_read(codec, WM8580_CLKSEL); + reg = snd_soc_read(codec, WM8580_CLKSEL); reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK; switch (div) { @@ -704,11 +643,11 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, default: return -EINVAL; } - wm8580_write(codec, WM8580_CLKSEL, reg); + snd_soc_write(codec, WM8580_CLKSEL, reg); break; case WM8580_CLKOUTSRC: - reg = wm8580_read(codec, WM8580_PLLB4); + reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; switch (div) { @@ -730,7 +669,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, default: return -EINVAL; } - wm8580_write(codec, WM8580_PLLB4, reg); + snd_soc_write(codec, WM8580_PLLB4, reg); break; default: @@ -745,14 +684,14 @@ static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_codec *codec = codec_dai->codec; unsigned int reg; - reg = wm8580_read(codec, WM8580_DAC_CONTROL5); + reg = snd_soc_read(codec, WM8580_DAC_CONTROL5); if (mute) reg |= WM8580_DAC_CONTROL5_MUTEALL; else reg &= ~WM8580_DAC_CONTROL5_MUTEALL; - wm8580_write(codec, WM8580_DAC_CONTROL5, reg); + snd_soc_write(codec, WM8580_DAC_CONTROL5, reg); return 0; } @@ -769,20 +708,20 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Power up and get individual control of the DACs */ - reg = wm8580_read(codec, WM8580_PWRDN1); + reg = snd_soc_read(codec, WM8580_PWRDN1); reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); - wm8580_write(codec, WM8580_PWRDN1, reg); + snd_soc_write(codec, WM8580_PWRDN1, reg); /* Make VMID high impedence */ - reg = wm8580_read(codec, WM8580_ADC_CONTROL1); + reg = snd_soc_read(codec, WM8580_ADC_CONTROL1); reg &= ~0x100; - wm8580_write(codec, WM8580_ADC_CONTROL1, reg); + snd_soc_write(codec, WM8580_ADC_CONTROL1, reg); } break; case SND_SOC_BIAS_OFF: - reg = wm8580_read(codec, WM8580_PWRDN1); - wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); + reg = snd_soc_read(codec, WM8580_PWRDN1); + snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN); break; } codec->bias_level = level; @@ -893,7 +832,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); -static int wm8580_register(struct wm8580_priv *wm8580) +static int wm8580_register(struct wm8580_priv *wm8580, + enum snd_soc_control_type control) { int ret, i; struct snd_soc_codec *codec = &wm8580->codec; @@ -911,8 +851,6 @@ static int wm8580_register(struct wm8580_priv *wm8580) codec->private_data = wm8580; codec->name = "WM8580"; codec->owner = THIS_MODULE; - codec->read = wm8580_read_reg_cache; - codec->write = wm8580_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8580_set_bias_level; codec->dai = wm8580_dai; @@ -922,11 +860,34 @@ static int wm8580_register(struct wm8580_priv *wm8580) memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8580->supplies); i++) + wm8580->supplies[i].supply = wm8580_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8580->supplies), + wm8580->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), + wm8580->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_regulator_get; + } + /* Get the codec into a known state */ - ret = wm8580_write(codec, WM8580_RESET, 0); + ret = snd_soc_write(codec, WM8580_RESET, 0); if (ret != 0) { dev_err(codec->dev, "Failed to reset codec: %d\n", ret); - goto err; + goto err_regulator_enable; } for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++) @@ -939,7 +900,7 @@ static int wm8580_register(struct wm8580_priv *wm8580) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto err; + goto err_regulator_enable; } ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); @@ -952,6 +913,10 @@ static int wm8580_register(struct wm8580_priv *wm8580) err_codec: snd_soc_unregister_codec(codec); +err_regulator_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); +err_regulator_get: + regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); err: kfree(wm8580); return ret; @@ -962,6 +927,8 @@ static void wm8580_unregister(struct wm8580_priv *wm8580) wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); snd_soc_unregister_codec(&wm8580->codec); + regulator_bulk_disable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); kfree(wm8580); wm8580_codec = NULL; } @@ -978,14 +945,13 @@ static int wm8580_i2c_probe(struct i2c_client *i2c, return -ENOMEM; codec = &wm8580->codec; - codec->hw_write = (hw_write_t)i2c_master_send; i2c_set_clientdata(i2c, wm8580); codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8580_register(wm8580); + return wm8580_register(wm8580, SND_SOC_I2C); } static int wm8580_i2c_remove(struct i2c_client *client) @@ -995,6 +961,21 @@ static int wm8580_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8580_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8580_i2c_suspend NULL +#define wm8580_i2c_resume NULL +#endif + static const struct i2c_device_id wm8580_i2c_id[] = { { "wm8580", 0 }, { } @@ -1008,6 +989,8 @@ static struct i2c_driver wm8580_i2c_driver = { }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, + .suspend = wm8580_i2c_suspend, + .resume = wm8580_i2c_resume, .id_table = wm8580_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index e7ff2121ede9..16e969a762c3 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -43,45 +43,6 @@ static const u16 wm8728_reg_defaults[] = { 0x100, }; -static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); - return cache[reg]; -} - -static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); - cache[reg] = value; -} - -/* - * write to the WM8728 register space - */ -static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8728 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8728_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); static const struct snd_kcontrol_new wm8728_snd_controls[] = { @@ -121,12 +82,12 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec) static int wm8728_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + u16 mute_reg = snd_soc_read(codec, WM8728_DACCTL); if (mute) - wm8728_write(codec, WM8728_DACCTL, mute_reg | 1); + snd_soc_write(codec, WM8728_DACCTL, mute_reg | 1); else - wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1); + snd_soc_write(codec, WM8728_DACCTL, mute_reg & ~1); return 0; } @@ -138,7 +99,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); + u16 dac = snd_soc_read(codec, WM8728_DACCTL); dac &= ~0x18; @@ -155,7 +116,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8728_write(codec, WM8728_DACCTL, dac); + snd_soc_write(codec, WM8728_DACCTL, dac); return 0; } @@ -164,7 +125,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL); + u16 iface = snd_soc_read(codec, WM8728_IFCTL); /* Currently only I2S is supported by the driver, though the * hardware is more flexible. @@ -204,7 +165,7 @@ static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8728_write(codec, WM8728_IFCTL, iface); + snd_soc_write(codec, WM8728_IFCTL, iface); return 0; } @@ -220,19 +181,19 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Power everything up... */ - reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); - wm8728_write(codec, WM8728_DACCTL, reg & ~0x4); + reg = snd_soc_read(codec, WM8728_DACCTL); + snd_soc_write(codec, WM8728_DACCTL, reg & ~0x4); /* ..then sync in the register cache. */ for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) - wm8728_write(codec, i, - wm8728_read_reg_cache(codec, i)); + snd_soc_write(codec, i, + snd_soc_read(codec, i)); } break; case SND_SOC_BIAS_OFF: - reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); - wm8728_write(codec, WM8728_DACCTL, reg | 0x4); + reg = snd_soc_read(codec, WM8728_DACCTL); + snd_soc_write(codec, WM8728_DACCTL, reg | 0x4); break; } codec->bias_level = level; @@ -287,15 +248,14 @@ static int wm8728_resume(struct platform_device *pdev) * initialise the WM8728 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8728_init(struct snd_soc_device *socdev) +static int wm8728_init(struct snd_soc_device *socdev, + enum snd_soc_control_type control) { struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8728"; codec->owner = THIS_MODULE; - codec->read = wm8728_read_reg_cache; - codec->write = wm8728_write; codec->set_bias_level = wm8728_set_bias_level; codec->dai = &wm8728_dai; codec->num_dai = 1; @@ -307,11 +267,18 @@ static int wm8728_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", + ret); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8728: failed to create pcms\n"); - goto pcm_err; + goto err; } /* power on device */ @@ -331,7 +298,7 @@ static int wm8728_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } @@ -357,7 +324,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = wm8728_init(socdev); + ret = wm8728_init(socdev, SND_SOC_I2C); if (ret < 0) pr_err("failed to initialise WM8728\n"); @@ -437,7 +404,7 @@ static int __devinit wm8728_spi_probe(struct spi_device *spi) codec->control_data = spi; - ret = wm8728_init(socdev); + ret = wm8728_init(socdev, SND_SOC_SPI); if (ret < 0) dev_err(&spi->dev, "failed to initialise WM8728\n"); @@ -458,30 +425,6 @@ static struct spi_driver wm8728_spi_driver = { .probe = wm8728_spi_probe, .remove = __devexit_p(wm8728_spi_remove), }; - -static int wm8728_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} #endif /* CONFIG_SPI_MASTER */ static int wm8728_probe(struct platform_device *pdev) @@ -506,13 +449,11 @@ static int wm8728_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8728_add_i2c_device(pdev, setup); } #endif #if defined(CONFIG_SPI_MASTER) if (setup->spi) { - codec->hw_write = (hw_write_t)wm8728_spi_write; ret = spi_register_driver(&wm8728_spi_driver); if (ret != 0) printk(KERN_ERR "can't add spi driver"); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7a205876ef4f..d3fd4f28d96e 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -26,6 +26,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "wm8731.h" @@ -39,9 +40,6 @@ struct wm8731_priv { unsigned int sysclk; }; -#ifdef CONFIG_SPI_MASTER -static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); -#endif /* * wm8731 register cache @@ -50,60 +48,12 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len); * There is no point in caching the reset register */ static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 + 0x0097, 0x0097, 0x0079, 0x0079, + 0x000a, 0x0008, 0x009f, 0x000a, + 0x0000, 0x0000 }; -/* - * read wm8731 register cache - */ -static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8731_RESET) - return 0; - if (reg >= WM8731_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8731 register cache - */ -static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8731_CACHEREGNUM) - return; - cache[reg] = value; -} - -/* - * write to the WM8731 register space - */ -static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8731 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8731_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0) +#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -113,20 +63,26 @@ static const struct soc_enum wm8731_enum[] = { SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph), }; +static const DECLARE_TLV_DB_SCALE(in_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -1500, 300, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + static const struct snd_kcontrol_new wm8731_snd_controls[] = { -SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V, - 0, 127, 0), +SOC_DOUBLE_R_TLV("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V, + 0, 127, 0, out_tlv), SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V, 7, 1, 0), -SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0), +SOC_DOUBLE_R_TLV("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0, + in_tlv), SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1), SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0), -SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1), +SOC_SINGLE("Mic Capture Switch", WM8731_APANA, 1, 1, 1), -SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1), +SOC_SINGLE_TLV("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1, + sidetone_tlv), SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1), SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), @@ -260,12 +216,12 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8731_priv *wm8731 = codec->private_data; - u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3; + u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; int i = get_coeff(wm8731->sysclk, params_rate(params)); u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | coeff_div[i].usb; - wm8731_write(codec, WM8731_SRATE, srate); + snd_soc_write(codec, WM8731_SRATE, srate); /* bit size */ switch (params_format(params)) { @@ -279,7 +235,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, break; } - wm8731_write(codec, WM8731_IFACE, iface); + snd_soc_write(codec, WM8731_IFACE, iface); return 0; } @@ -291,7 +247,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; /* set active */ - wm8731_write(codec, WM8731_ACTIVE, 0x0001); + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); return 0; } @@ -306,19 +262,19 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream, /* deactivate */ if (!codec->active) { udelay(50); - wm8731_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); } } static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8731_APDIGI) & 0xfff7; if (mute) - wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8); + snd_soc_write(codec, WM8731_APDIGI, mute_reg | 0x8); else - wm8731_write(codec, WM8731_APDIGI, mute_reg); + snd_soc_write(codec, WM8731_APDIGI, mute_reg); return 0; } @@ -396,7 +352,7 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8731_write(codec, WM8731_IFACE, iface); + snd_soc_write(codec, WM8731_IFACE, iface); return 0; } @@ -412,12 +368,12 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: /* Clear PWROFF, gate CLKOUT, everything else as-is */ - reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; - wm8731_write(codec, WM8731_PWR, reg | 0x0040); + reg = snd_soc_read(codec, WM8731_PWR) & 0xff7f; + snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - wm8731_write(codec, WM8731_ACTIVE, 0x0); - wm8731_write(codec, WM8731_PWR, 0xffff); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_PWR, 0xffff); break; } codec->bias_level = level; @@ -457,15 +413,17 @@ struct snd_soc_dai wm8731_dai = { .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, .ops = &wm8731_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(wm8731_dai); +#ifdef CONFIG_PM static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - wm8731_write(codec, WM8731_ACTIVE, 0x0); + snd_soc_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } @@ -488,6 +446,10 @@ static int wm8731_resume(struct platform_device *pdev) wm8731_set_bias_level(codec, codec->suspend_bias_level); return 0; } +#else +#define wm8731_suspend NULL +#define wm8731_resume NULL +#endif static int wm8731_probe(struct platform_device *pdev) { @@ -547,15 +509,16 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); -static int wm8731_register(struct wm8731_priv *wm8731) +static int wm8731_register(struct wm8731_priv *wm8731, + enum snd_soc_control_type control) { int ret; struct snd_soc_codec *codec = &wm8731->codec; - u16 reg; if (wm8731_codec) { dev_err(codec->dev, "Another WM8731 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } mutex_init(&codec->mutex); @@ -565,8 +528,6 @@ static int wm8731_register(struct wm8731_priv *wm8731) codec->private_data = wm8731; codec->name = "WM8731"; codec->owner = THIS_MODULE; - codec->read = wm8731_read_reg_cache; - codec->write = wm8731_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8731_set_bias_level; codec->dai = &wm8731_dai; @@ -576,10 +537,16 @@ static int wm8731_register(struct wm8731_priv *wm8731) memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8731_reset(codec); if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err; } wm8731_dai.dev = codec->dev; @@ -587,35 +554,36 @@ static int wm8731_register(struct wm8731_priv *wm8731) wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); - reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); + snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); + snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); + snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); + snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - reg = wm8731_read_reg_cache(codec, WM8731_APANA); - wm8731_write(codec, WM8731_APANA, reg & ~0x4); + snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0); wm8731_codec = codec; ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm8731_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8731); + return ret; } static void wm8731_unregister(struct wm8731_priv *wm8731) @@ -628,30 +596,6 @@ static void wm8731_unregister(struct wm8731_priv *wm8731) } #if defined(CONFIG_SPI_MASTER) -static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} - static int __devinit wm8731_spi_probe(struct spi_device *spi) { struct snd_soc_codec *codec; @@ -663,12 +607,11 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) codec = &wm8731->codec; codec->control_data = spi; - codec->hw_write = (hw_write_t)wm8731_spi_write; codec->dev = &spi->dev; dev_set_drvdata(&spi->dev, wm8731); - return wm8731_register(wm8731); + return wm8731_register(wm8731, SND_SOC_SPI); } static int __devexit wm8731_spi_remove(struct spi_device *spi) @@ -680,6 +623,21 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_PM +static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8731_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8731_spi_suspend NULL +#define wm8731_spi_resume NULL +#endif + static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", @@ -687,6 +645,8 @@ static struct spi_driver wm8731_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8731_spi_probe, + .suspend = wm8731_spi_suspend, + .resume = wm8731_spi_resume, .remove = __devexit_p(wm8731_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -703,14 +663,13 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, return -ENOMEM; codec = &wm8731->codec; - codec->hw_write = (hw_write_t)i2c_master_send; i2c_set_clientdata(i2c, wm8731); codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8731_register(wm8731); + return wm8731_register(wm8731, SND_SOC_I2C); } static __devexit int wm8731_i2c_remove(struct i2c_client *client) @@ -720,6 +679,21 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8731_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8731_i2c_suspend NULL +#define wm8731_i2c_resume NULL +#endif + static const struct i2c_device_id wm8731_i2c_id[] = { { "wm8731", 0 }, { } @@ -733,6 +707,8 @@ static struct i2c_driver wm8731_i2c_driver = { }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), + .suspend = wm8731_i2c_suspend, + .resume = wm8731_i2c_resume, .id_table = wm8731_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index b64509b01a49..4ba1e7e93fb4 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -55,50 +55,7 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; -/* - * read wm8750 register cache - */ -static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8750_CACHE_REGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8750 register cache - */ -static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8750_CACHE_REGNUM) - return; - cache[reg] = value; -} - -static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8750_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0) +#define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* * WM8750 Controls @@ -594,7 +551,7 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8750_write(codec, WM8750_IFACE, iface); + snd_soc_write(codec, WM8750_IFACE, iface); return 0; } @@ -606,8 +563,8 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8750_priv *wm8750 = codec->private_data; - u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3; - u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0; + u16 iface = snd_soc_read(codec, WM8750_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8750_SRATE) & 0x1c0; int coeff = get_coeff(wm8750->sysclk, params_rate(params)); /* bit size */ @@ -626,9 +583,9 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8750_write(codec, WM8750_IFACE, iface); + snd_soc_write(codec, WM8750_IFACE, iface); if (coeff >= 0) - wm8750_write(codec, WM8750_SRATE, srate | + snd_soc_write(codec, WM8750_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -637,35 +594,35 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8750_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8750_ADCDAC) & 0xfff7; if (mute) - wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8750_ADCDAC, mute_reg | 0x8); else - wm8750_write(codec, WM8750_ADCDAC, mute_reg); + snd_soc_write(codec, WM8750_ADCDAC, mute_reg); return 0; } static int wm8750_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; + u16 pwr_reg = snd_soc_read(codec, WM8750_PWR1) & 0xfe3e; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_PREPARE: /* set vmid to 5k for quick power up */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); break; case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ - wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); + snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x0141); break; case SND_SOC_BIAS_OFF: - wm8750_write(codec, WM8750_PWR1, 0x0001); + snd_soc_write(codec, WM8750_PWR1, 0x0001); break; } codec->bias_level = level; @@ -754,15 +711,14 @@ static int wm8750_resume(struct platform_device *pdev) * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev) +static int wm8750_init(struct snd_soc_device *socdev, + enum snd_soc_control_type control) { struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8750"; codec->owner = THIS_MODULE; - codec->read = wm8750_read_reg_cache; - codec->write = wm8750_write; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; @@ -771,13 +727,23 @@ static int wm8750_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; - wm8750_reset(codec); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); + goto err; + } + + ret = wm8750_reset(codec); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to reset: %d\n", ret); + goto err; + } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto pcm_err; + goto err; } /* charge output caps */ @@ -786,22 +752,22 @@ static int wm8750_init(struct snd_soc_device *socdev) schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ - reg = wm8750_read_reg_cache(codec, WM8750_LDAC); - wm8750_write(codec, WM8750_LDAC, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_RDAC); - wm8750_write(codec, WM8750_RDAC, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V); - wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V); - wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V); - wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V); - wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_LINVOL); - wm8750_write(codec, WM8750_LINVOL, reg | 0x0100); - reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); - wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LDAC); + snd_soc_write(codec, WM8750_LDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_RDAC); + snd_soc_write(codec, WM8750_RDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LOUT1V); + snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_ROUT1V); + snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LOUT2V); + snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_ROUT2V); + snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_LINVOL); + snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8750_RINVOL); + snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); @@ -816,7 +782,7 @@ static int wm8750_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } @@ -844,7 +810,7 @@ static int wm8750_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = wm8750_init(socdev); + ret = wm8750_init(socdev, SND_SOC_I2C); if (ret < 0) pr_err("failed to initialise WM8750\n"); @@ -924,7 +890,7 @@ static int __devinit wm8750_spi_probe(struct spi_device *spi) codec->control_data = spi; - ret = wm8750_init(socdev); + ret = wm8750_init(socdev, SND_SOC_SPI); if (ret < 0) dev_err(&spi->dev, "failed to initialise WM8750\n"); @@ -945,30 +911,6 @@ static struct spi_driver wm8750_spi_driver = { .probe = wm8750_spi_probe, .remove = __devexit_p(wm8750_spi_remove), }; - -static int wm8750_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} #endif static int wm8750_probe(struct platform_device *pdev) @@ -1002,13 +944,11 @@ static int wm8750_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8750_add_i2c_device(pdev, setup); } #endif #if defined(CONFIG_SPI_MASTER) if (setup->spi) { - codec->hw_write = (hw_write_t)wm8750_spi_write; ret = spi_register_driver(&wm8750_spi_driver); if (ret != 0) printk(KERN_ERR "can't add spi driver"); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 49c4b2898aff..d80d414cfbbd 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1766,6 +1766,21 @@ static int wm8753_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8753_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8753_i2c_suspend NULL +#define wm8753_i2c_resume NULL +#endif + static const struct i2c_device_id wm8753_i2c_id[] = { { "wm8753", 0 }, { } @@ -1779,6 +1794,8 @@ static struct i2c_driver wm8753_i2c_driver = { }, .probe = wm8753_i2c_probe, .remove = wm8753_i2c_remove, + .suspend = wm8753_i2c_suspend, + .resume = wm8753_i2c_resume, .id_table = wm8753_i2c_id, }; #endif @@ -1834,6 +1851,22 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_PM +static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8753_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} + +#else +#define wm8753_spi_suspend NULL +#define wm8753_spi_resume NULL +#endif + static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", @@ -1842,6 +1875,8 @@ static struct spi_driver wm8753_spi_driver = { }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), + .suspend = wm8753_spi_suspend, + .resume = wm8753_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c new file mode 100644 index 000000000000..a9829aa26e53 --- /dev/null +++ b/sound/soc/codecs/wm8776.c @@ -0,0 +1,744 @@ +/* + * wm8776.c -- WM8776 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * TODO: Input ALC/limiter support + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8776.h" + +static struct snd_soc_codec *wm8776_codec; +struct snd_soc_codec_device soc_codec_dev_wm8776; + +/* codec private data */ +struct wm8776_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8776_CACHEREGNUM]; + int sysclk[2]; +}; + +#ifdef CONFIG_SPI_MASTER +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len); +#endif + +static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { + 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ + 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ + 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ + 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ + 0xa6, 0x01, 0x01 +}; + +static int wm8776_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8776_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(hp_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new wm8776_snd_controls[] = { +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8776_HPLVOL, WM8776_HPRVOL, + 0, 127, 0, hp_tlv), +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8776_DACLVOL, WM8776_DACRVOL, + 0, 255, 0, dac_tlv), +SOC_SINGLE("Digital Playback ZC Switch", WM8776_DACCTRL1, 0, 1, 0), + +SOC_SINGLE("Deemphasis Switch", WM8776_DACCTRL2, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Volume", WM8776_ADCLVOL, WM8776_ADCRVOL, + 0, 255, 0, adc_tlv), +SOC_DOUBLE("Capture Switch", WM8776_ADCMUX, 7, 6, 1, 1), +SOC_DOUBLE_R("Capture ZC Switch", WM8776_ADCLVOL, WM8776_ADCRVOL, 8, 1, 0), +SOC_SINGLE("Capture HPF Switch", WM8776_ADCIFCTRL, 8, 1, 1), +}; + +static const struct snd_kcontrol_new inmix_controls[] = { +SOC_DAPM_SINGLE("AIN1 Switch", WM8776_ADCMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AIN2 Switch", WM8776_ADCMUX, 1, 1, 0), +SOC_DAPM_SINGLE("AIN3 Switch", WM8776_ADCMUX, 2, 1, 0), +SOC_DAPM_SINGLE("AIN4 Switch", WM8776_ADCMUX, 3, 1, 0), +SOC_DAPM_SINGLE("AIN5 Switch", WM8776_ADCMUX, 4, 1, 0), +}; + +static const struct snd_kcontrol_new outmix_controls[] = { +SOC_DAPM_SINGLE("DAC Switch", WM8776_OUTMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8776_OUTMUX, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_INPUT("AIN1"), +SND_SOC_DAPM_INPUT("AIN2"), +SND_SOC_DAPM_INPUT("AIN3"), +SND_SOC_DAPM_INPUT("AIN4"), +SND_SOC_DAPM_INPUT("AIN5"), + +SND_SOC_DAPM_MIXER("Input Mixer", WM8776_PWRDOWN, 6, 1, + inmix_controls, ARRAY_SIZE(inmix_controls)), + +SND_SOC_DAPM_ADC("ADC", "Capture", WM8776_PWRDOWN, 1, 1), +SND_SOC_DAPM_DAC("DAC", "Playback", WM8776_PWRDOWN, 2, 1), + +SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + outmix_controls, ARRAY_SIZE(outmix_controls)), + +SND_SOC_DAPM_PGA("Headphone PGA", WM8776_PWRDOWN, 3, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("VOUT"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +}; + +static const struct snd_soc_dapm_route routes[] = { + { "Input Mixer", "AIN1 Switch", "AIN1" }, + { "Input Mixer", "AIN2 Switch", "AIN2" }, + { "Input Mixer", "AIN3 Switch", "AIN3" }, + { "Input Mixer", "AIN4 Switch", "AIN4" }, + { "Input Mixer", "AIN5 Switch", "AIN5" }, + + { "ADC", NULL, "Input Mixer" }, + + { "Output Mixer", "DAC Switch", "DAC" }, + { "Output Mixer", "AUX Switch", "AUX" }, + { "Output Mixer", "Bypass Switch", "Input Mixer" }, + + { "VOUT", NULL, "Output Mixer" }, + + { "Headphone PGA", NULL, "Output Mixer" }, + + { "HPOUTL", NULL, "Headphone PGA" }, + { "HPOUTR", NULL, "Headphone PGA" }, +}; + +static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int reg, iface, master; + + switch (dai->id) { + case WM8776_DAI_DAC: + reg = WM8776_DACIFCTRL; + master = 0x80; + break; + case WM8776_DAI_ADC: + reg = WM8776_ADCIFCTRL; + master = 0x100; + break; + default: + return -EINVAL; + } + + iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + /* FIXME: CHECK A/B */ + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0007; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x00c; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x008; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x004; + break; + default: + return -EINVAL; + } + + /* Finally, write out the values */ + snd_soc_update_bits(codec, reg, 0xf, iface); + snd_soc_update_bits(codec, WM8776_MSTRCTRL, 0x180, master); + + return 0; +} + +static int mclk_ratios[] = { + 128, + 192, + 256, + 384, + 512, + 768, +}; + +static int wm8776_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + int iface_reg, iface; + int ratio_shift, master; + int i; + + iface = 0; + + switch (dai->id) { + case WM8776_DAI_DAC: + iface_reg = WM8776_DACIFCTRL; + master = 0x80; + ratio_shift = 4; + break; + case WM8776_DAI_ADC: + iface_reg = WM8776_ADCIFCTRL; + master = 0x100; + ratio_shift = 0; + break; + default: + return -EINVAL; + } + + + /* Set word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x30; + break; + } + + /* Only need to set MCLK/LRCLK ratio if we're master */ + if (snd_soc_read(codec, WM8776_MSTRCTRL) & master) { + for (i = 0; i < ARRAY_SIZE(mclk_ratios); i++) { + if (wm8776->sysclk[dai->id] / params_rate(params) + == mclk_ratios[i]) + break; + } + + if (i == ARRAY_SIZE(mclk_ratios)) { + dev_err(codec->dev, + "Unable to configure MCLK ratio %d/%d\n", + wm8776->sysclk[dai->id], params_rate(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]); + + snd_soc_update_bits(codec, WM8776_MSTRCTRL, + 0x7 << ratio_shift, i << ratio_shift); + } else { + dev_dbg(codec->dev, "DAI in slave mode\n"); + } + + snd_soc_update_bits(codec, iface_reg, 0x30, iface); + + return 0; +} + +static int wm8776_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_write(codec, WM8776_DACMUTE, !!mute); +} + +static int wm8776_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + + BUG_ON(dai->id >= ARRAY_SIZE(wm8776->sysclk)); + + wm8776->sysclk[dai->id] = freq; + + return 0; +} + +static int wm8776_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Disable the global powerdown; DAPM does the rest */ + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); + } + + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 1); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + + +#define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8776_dac_ops = { + .digital_mute = wm8776_mute, + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +static struct snd_soc_dai_ops wm8776_adc_ops = { + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +struct snd_soc_dai wm8776_dai[] = { + { + .name = "WM8776 Playback", + .id = WM8776_DAI_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_dac_ops, + }, + { + .name = "WM8776 Capture", + .id = WM8776_DAI_ADC, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_adc_ops, + }, +}; +EXPORT_SYMBOL_GPL(wm8776_dai); + +#ifdef CONFIG_PM +static int wm8776_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8776_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8776_suspend NULL +#define wm8776_resume NULL +#endif + +static int wm8776_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8776_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8776_codec; + codec = wm8776_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8776_snd_controls, + ARRAY_SIZE(wm8776_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets, + ARRAY_SIZE(wm8776_dapm_widgets)); + snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8776_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8776 = { + .probe = wm8776_probe, + .remove = wm8776_remove, + .suspend = wm8776_suspend, + .resume = wm8776_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8776); + +static int wm8776_register(struct wm8776_priv *wm8776, + enum snd_soc_control_type control) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8776->codec; + + if (wm8776_codec) { + dev_err(codec->dev, "Another WM8776 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8776; + codec->name = "WM8776"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8776_set_bias_level; + codec->dai = wm8776_dai; + codec->num_dai = ARRAY_SIZE(wm8776_dai); + codec->reg_cache_size = WM8776_CACHEREGNUM; + codec->reg_cache = &wm8776->reg_cache; + + memcpy(codec->reg_cache, wm8776_reg, sizeof(wm8776_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8776_dai); i++) + wm8776_dai[i].dev = codec->dev; + + ret = wm8776_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err; + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits; right channel only since we always + * update both. */ + snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100); + + wm8776_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8776); + return ret; +} + +static void wm8776_unregister(struct wm8776_priv *wm8776) +{ + wm8776_set_bias_level(&wm8776->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + snd_soc_unregister_codec(&wm8776->codec); + kfree(wm8776); + wm8776_codec = NULL; +} + +#if defined(CONFIG_SPI_MASTER) +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8776_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8776_priv *wm8776; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8776_spi_write; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8776); + + return wm8776_register(wm8776, SND_SOC_SPI); +} + +static int __devexit wm8776_spi_remove(struct spi_device *spi) +{ + struct wm8776_priv *wm8776 = dev_get_drvdata(&spi->dev); + + wm8776_unregister(wm8776); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8776_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8776_spi_suspend NULL +#define wm8776_spi_resume NULL +#endif + +static struct spi_driver wm8776_spi_driver = { + .driver = { + .name = "wm8776", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8776_spi_probe, + .suspend = wm8776_spi_suspend, + .resume = wm8776_spi_resume, + .remove = __devexit_p(wm8776_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8776_priv *wm8776; + struct snd_soc_codec *codec; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8776); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8776_register(wm8776, SND_SOC_I2C); +} + +static __devexit int wm8776_i2c_remove(struct i2c_client *client) +{ + struct wm8776_priv *wm8776 = i2c_get_clientdata(client); + wm8776_unregister(wm8776); + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8776_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8776_i2c_suspend NULL +#define wm8776_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8776_i2c_id[] = { + { "wm8776", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); + +static struct i2c_driver wm8776_i2c_driver = { + .driver = { + .name = "wm8776", + .owner = THIS_MODULE, + }, + .probe = wm8776_i2c_probe, + .remove = __devexit_p(wm8776_i2c_remove), + .suspend = wm8776_i2c_suspend, + .resume = wm8776_i2c_resume, + .id_table = wm8776_i2c_id, +}; +#endif + +static int __init wm8776_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8776_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8776_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 SPI driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8776_modinit); + +static void __exit wm8776_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8776_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8776_spi_driver); +#endif +} +module_exit(wm8776_exit); + +MODULE_DESCRIPTION("ASoC WM8776 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8776.h b/sound/soc/codecs/wm8776.h new file mode 100644 index 000000000000..6606d25d2d83 --- /dev/null +++ b/sound/soc/codecs/wm8776.h @@ -0,0 +1,51 @@ +/* + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8776_H +#define _WM8776_H + +/* Registers */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + +#define WM8776_CACHEREGNUM 0x17 + +#define WM8776_DAI_DAC 0 +#define WM8776_DAI_ADC 1 + +extern struct snd_soc_dai wm8776_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_wm8776; + +#endif diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 3c78945244b8..5e9c855c0036 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -116,6 +116,7 @@ #define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c #define WM8900_REG_DACCTRL_MUTE 0x004 +#define WM8900_REG_DACCTRL_DAC_SB_FILT 0x100 #define WM8900_REG_DACCTRL_AIF_LRCLKRATE 0x400 #define WM8900_REG_AUDIO3_ADCLRC_DIR 0x0800 @@ -182,111 +183,20 @@ static const u16 wm8900_reg_defaults[WM8900_MAXREG] = { /* Remaining registers all zero */ }; -/* - * read wm8900 register cache - */ -static inline unsigned int wm8900_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - - BUG_ON(reg >= WM8900_MAXREG); - - if (reg == WM8900_REG_ID) - return 0; - - return cache[reg]; -} - -/* - * write wm8900 register cache - */ -static inline void wm8900_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - - BUG_ON(reg >= WM8900_MAXREG); - - cache[reg] = value; -} - -/* - * write to the WM8900 register space - */ -static int wm8900_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - - if (value == wm8900_read_reg_cache(codec, reg)) - return 0; - - /* data is - * D15..D9 WM8900 register offset - * D8...D0 register data - */ - data[0] = reg; - data[1] = value >> 8; - data[2] = value & 0x00ff; - - wm8900_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 3) == 3) - return 0; - else - return -EIO; -} - -/* - * Read from the wm8900. - */ -static unsigned int wm8900_chip_read(struct snd_soc_codec *codec, u8 reg) -{ - struct i2c_msg xfer[2]; - u16 data; - int ret; - struct i2c_client *client = codec->control_data; - - BUG_ON(reg != WM8900_REG_ID && reg != WM8900_REG_POWER1); - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - printk(KERN_CRIT "i2c_transfer returned %d\n", ret); - return 0; - } - - return (data >> 8) | ((data & 0xff) << 8); -} - -/* - * Read from the WM8900 register space. Most registers can't be read - * and are therefore supplied from cache. - */ -static unsigned int wm8900_read(struct snd_soc_codec *codec, unsigned int reg) +static int wm8900_volatile_register(unsigned int reg) { switch (reg) { case WM8900_REG_ID: - return wm8900_chip_read(codec, reg); + case WM8900_REG_POWER1: + return 1; default: - return wm8900_read_reg_cache(codec, reg); + return 0; } } static void wm8900_reset(struct snd_soc_codec *codec) { - wm8900_write(codec, WM8900_REG_RESET, 0); + snd_soc_write(codec, WM8900_REG_RESET, 0); memcpy(codec->reg_cache, wm8900_reg_defaults, sizeof(codec->reg_cache)); @@ -296,14 +206,14 @@ static int wm8900_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 hpctl1 = wm8900_read(codec, WM8900_REG_HPCTL1); + u16 hpctl1 = snd_soc_read(codec, WM8900_REG_HPCTL1); switch (event) { case SND_SOC_DAPM_PRE_PMU: /* Clamp headphone outputs */ hpctl1 = WM8900_REG_HPCTL1_HP_CLAMP_IP | WM8900_REG_HPCTL1_HP_CLAMP_OP; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); break; case SND_SOC_DAPM_POST_PMU: @@ -312,41 +222,41 @@ static int wm8900_hp_event(struct snd_soc_dapm_widget *w, hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT | WM8900_REG_HPCTL1_HP_SHORT2 | WM8900_REG_HPCTL1_HP_IPSTAGE_ENA; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); msleep(400); /* Enable the output stage */ hpctl1 &= ~WM8900_REG_HPCTL1_HP_CLAMP_OP; hpctl1 |= WM8900_REG_HPCTL1_HP_OPSTAGE_ENA; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); /* Remove the shorts */ hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT2; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); hpctl1 &= ~WM8900_REG_HPCTL1_HP_SHORT; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); break; case SND_SOC_DAPM_PRE_PMD: /* Short the output */ hpctl1 |= WM8900_REG_HPCTL1_HP_SHORT; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); /* Disable the output stage */ hpctl1 &= ~WM8900_REG_HPCTL1_HP_OPSTAGE_ENA; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); /* Clamp the outputs and power down input */ hpctl1 |= WM8900_REG_HPCTL1_HP_CLAMP_IP | WM8900_REG_HPCTL1_HP_CLAMP_OP; hpctl1 &= ~WM8900_REG_HPCTL1_HP_IPSTAGE_ENA; - wm8900_write(codec, WM8900_REG_HPCTL1, hpctl1); + snd_soc_write(codec, WM8900_REG_HPCTL1, hpctl1); break; case SND_SOC_DAPM_POST_PMD: /* Disable everything */ - wm8900_write(codec, WM8900_REG_HPCTL1, 0); + snd_soc_write(codec, WM8900_REG_HPCTL1, 0); break; default: @@ -439,7 +349,6 @@ SOC_SINGLE("DAC Soft Mute Switch", WM8900_REG_DACCTRL, 6, 1, 1), SOC_ENUM("DAC Mute Rate", dac_mute_rate), SOC_SINGLE("DAC Mono Switch", WM8900_REG_DACCTRL, 9, 1, 0), SOC_ENUM("DAC Deemphasis", dac_deemphasis), -SOC_SINGLE("DAC Sloping Stopband Filter Switch", WM8900_REG_DACCTRL, 8, 1, 0), SOC_SINGLE("DAC Sigma-Delta Modulator Clock Switch", WM8900_REG_DACCTRL, 12, 1, 0), @@ -723,7 +632,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u16 reg; - reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60; + reg = snd_soc_read(codec, WM8900_REG_AUDIO1) & ~0x60; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -741,7 +650,18 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8900_write(codec, WM8900_REG_AUDIO1, reg); + snd_soc_write(codec, WM8900_REG_AUDIO1, reg); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg = snd_soc_read(codec, WM8900_REG_DACCTRL); + + if (params_rate(params) <= 24000) + reg |= WM8900_REG_DACCTRL_DAC_SB_FILT; + else + reg &= ~WM8900_REG_DACCTRL_DAC_SB_FILT; + + snd_soc_write(codec, WM8900_REG_DACCTRL, reg); + } return 0; } @@ -834,18 +754,18 @@ static int wm8900_set_fll(struct snd_soc_codec *codec, return 0; /* The digital side should be disabled during any change. */ - reg = wm8900_read(codec, WM8900_REG_POWER1); - wm8900_write(codec, WM8900_REG_POWER1, + reg = snd_soc_read(codec, WM8900_REG_POWER1); + snd_soc_write(codec, WM8900_REG_POWER1, reg & (~WM8900_REG_POWER1_FLL_ENA)); /* Disable the FLL? */ if (!freq_in || !freq_out) { - reg = wm8900_read(codec, WM8900_REG_CLOCKING1); - wm8900_write(codec, WM8900_REG_CLOCKING1, + reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); + snd_soc_write(codec, WM8900_REG_CLOCKING1, reg & (~WM8900_REG_CLOCKING1_MCLK_SRC)); - reg = wm8900_read(codec, WM8900_REG_FLLCTL1); - wm8900_write(codec, WM8900_REG_FLLCTL1, + reg = snd_soc_read(codec, WM8900_REG_FLLCTL1); + snd_soc_write(codec, WM8900_REG_FLLCTL1, reg & (~WM8900_REG_FLLCTL1_OSC_ENA)); wm8900->fll_in = freq_in; @@ -862,33 +782,33 @@ static int wm8900_set_fll(struct snd_soc_codec *codec, /* The osclilator *MUST* be enabled before we enable the * digital circuit. */ - wm8900_write(codec, WM8900_REG_FLLCTL1, + snd_soc_write(codec, WM8900_REG_FLLCTL1, fll_div.fll_ratio | WM8900_REG_FLLCTL1_OSC_ENA); - wm8900_write(codec, WM8900_REG_FLLCTL4, fll_div.n >> 5); - wm8900_write(codec, WM8900_REG_FLLCTL5, + snd_soc_write(codec, WM8900_REG_FLLCTL4, fll_div.n >> 5); + snd_soc_write(codec, WM8900_REG_FLLCTL5, (fll_div.fllclk_div << 6) | (fll_div.n & 0x1f)); if (fll_div.k) { - wm8900_write(codec, WM8900_REG_FLLCTL2, + snd_soc_write(codec, WM8900_REG_FLLCTL2, (fll_div.k >> 8) | 0x100); - wm8900_write(codec, WM8900_REG_FLLCTL3, fll_div.k & 0xff); + snd_soc_write(codec, WM8900_REG_FLLCTL3, fll_div.k & 0xff); } else - wm8900_write(codec, WM8900_REG_FLLCTL2, 0); + snd_soc_write(codec, WM8900_REG_FLLCTL2, 0); if (fll_div.fll_slow_lock_ref) - wm8900_write(codec, WM8900_REG_FLLCTL6, + snd_soc_write(codec, WM8900_REG_FLLCTL6, WM8900_REG_FLLCTL6_FLL_SLOW_LOCK_REF); else - wm8900_write(codec, WM8900_REG_FLLCTL6, 0); + snd_soc_write(codec, WM8900_REG_FLLCTL6, 0); - reg = wm8900_read(codec, WM8900_REG_POWER1); - wm8900_write(codec, WM8900_REG_POWER1, + reg = snd_soc_read(codec, WM8900_REG_POWER1); + snd_soc_write(codec, WM8900_REG_POWER1, reg | WM8900_REG_POWER1_FLL_ENA); reenable: - reg = wm8900_read(codec, WM8900_REG_CLOCKING1); - wm8900_write(codec, WM8900_REG_CLOCKING1, + reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); + snd_soc_write(codec, WM8900_REG_CLOCKING1, reg | WM8900_REG_CLOCKING1_MCLK_SRC); return 0; @@ -908,38 +828,38 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8900_BCLK_DIV: - reg = wm8900_read(codec, WM8900_REG_CLOCKING1); - wm8900_write(codec, WM8900_REG_CLOCKING1, + reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); + snd_soc_write(codec, WM8900_REG_CLOCKING1, div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK)); break; case WM8900_OPCLK_DIV: - reg = wm8900_read(codec, WM8900_REG_CLOCKING1); - wm8900_write(codec, WM8900_REG_CLOCKING1, + reg = snd_soc_read(codec, WM8900_REG_CLOCKING1); + snd_soc_write(codec, WM8900_REG_CLOCKING1, div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK)); break; case WM8900_DAC_LRCLK: - reg = wm8900_read(codec, WM8900_REG_AUDIO4); - wm8900_write(codec, WM8900_REG_AUDIO4, + reg = snd_soc_read(codec, WM8900_REG_AUDIO4); + snd_soc_write(codec, WM8900_REG_AUDIO4, div | (reg & WM8900_LRC_MASK)); break; case WM8900_ADC_LRCLK: - reg = wm8900_read(codec, WM8900_REG_AUDIO3); - wm8900_write(codec, WM8900_REG_AUDIO3, + reg = snd_soc_read(codec, WM8900_REG_AUDIO3); + snd_soc_write(codec, WM8900_REG_AUDIO3, div | (reg & WM8900_LRC_MASK)); break; case WM8900_DAC_CLKDIV: - reg = wm8900_read(codec, WM8900_REG_CLOCKING2); - wm8900_write(codec, WM8900_REG_CLOCKING2, + reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); + snd_soc_write(codec, WM8900_REG_CLOCKING2, div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV)); break; case WM8900_ADC_CLKDIV: - reg = wm8900_read(codec, WM8900_REG_CLOCKING2); - wm8900_write(codec, WM8900_REG_CLOCKING2, + reg = snd_soc_read(codec, WM8900_REG_CLOCKING2); + snd_soc_write(codec, WM8900_REG_CLOCKING2, div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV)); break; case WM8900_LRCLK_MODE: - reg = wm8900_read(codec, WM8900_REG_DACCTRL); - wm8900_write(codec, WM8900_REG_DACCTRL, + reg = snd_soc_read(codec, WM8900_REG_DACCTRL); + snd_soc_write(codec, WM8900_REG_DACCTRL, div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE)); break; default: @@ -956,10 +876,10 @@ static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; unsigned int clocking1, aif1, aif3, aif4; - clocking1 = wm8900_read(codec, WM8900_REG_CLOCKING1); - aif1 = wm8900_read(codec, WM8900_REG_AUDIO1); - aif3 = wm8900_read(codec, WM8900_REG_AUDIO3); - aif4 = wm8900_read(codec, WM8900_REG_AUDIO4); + clocking1 = snd_soc_read(codec, WM8900_REG_CLOCKING1); + aif1 = snd_soc_read(codec, WM8900_REG_AUDIO1); + aif3 = snd_soc_read(codec, WM8900_REG_AUDIO3); + aif4 = snd_soc_read(codec, WM8900_REG_AUDIO4); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1055,10 +975,10 @@ static int wm8900_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8900_write(codec, WM8900_REG_CLOCKING1, clocking1); - wm8900_write(codec, WM8900_REG_AUDIO1, aif1); - wm8900_write(codec, WM8900_REG_AUDIO3, aif3); - wm8900_write(codec, WM8900_REG_AUDIO4, aif4); + snd_soc_write(codec, WM8900_REG_CLOCKING1, clocking1); + snd_soc_write(codec, WM8900_REG_AUDIO1, aif1); + snd_soc_write(codec, WM8900_REG_AUDIO3, aif3); + snd_soc_write(codec, WM8900_REG_AUDIO4, aif4); return 0; } @@ -1068,14 +988,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_codec *codec = codec_dai->codec; u16 reg; - reg = wm8900_read(codec, WM8900_REG_DACCTRL); + reg = snd_soc_read(codec, WM8900_REG_DACCTRL); if (mute) reg |= WM8900_REG_DACCTRL_MUTE; else reg &= ~WM8900_REG_DACCTRL_MUTE; - wm8900_write(codec, WM8900_REG_DACCTRL, reg); + snd_soc_write(codec, WM8900_REG_DACCTRL, reg); return 0; } @@ -1124,11 +1044,11 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: /* Enable thermal shutdown */ - reg = wm8900_read(codec, WM8900_REG_GPIO); - wm8900_write(codec, WM8900_REG_GPIO, + reg = snd_soc_read(codec, WM8900_REG_GPIO); + snd_soc_write(codec, WM8900_REG_GPIO, reg | WM8900_REG_GPIO_TEMP_ENA); - reg = wm8900_read(codec, WM8900_REG_ADDCTL); - wm8900_write(codec, WM8900_REG_ADDCTL, + reg = snd_soc_read(codec, WM8900_REG_ADDCTL); + snd_soc_write(codec, WM8900_REG_ADDCTL, reg | WM8900_REG_ADDCTL_TEMP_SD); break; @@ -1139,69 +1059,69 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, /* Charge capacitors if initial power up */ if (codec->bias_level == SND_SOC_BIAS_OFF) { /* STARTUP_BIAS_ENA on */ - wm8900_write(codec, WM8900_REG_POWER1, + snd_soc_write(codec, WM8900_REG_POWER1, WM8900_REG_POWER1_STARTUP_BIAS_ENA); /* Startup bias mode */ - wm8900_write(codec, WM8900_REG_ADDCTL, + snd_soc_write(codec, WM8900_REG_ADDCTL, WM8900_REG_ADDCTL_BIAS_SRC | WM8900_REG_ADDCTL_VMID_SOFTST); /* VMID 2x50k */ - wm8900_write(codec, WM8900_REG_POWER1, + snd_soc_write(codec, WM8900_REG_POWER1, WM8900_REG_POWER1_STARTUP_BIAS_ENA | 0x1); /* Allow capacitors to charge */ schedule_timeout_interruptible(msecs_to_jiffies(400)); /* Enable bias */ - wm8900_write(codec, WM8900_REG_POWER1, + snd_soc_write(codec, WM8900_REG_POWER1, WM8900_REG_POWER1_STARTUP_BIAS_ENA | WM8900_REG_POWER1_BIAS_ENA | 0x1); - wm8900_write(codec, WM8900_REG_ADDCTL, 0); + snd_soc_write(codec, WM8900_REG_ADDCTL, 0); - wm8900_write(codec, WM8900_REG_POWER1, + snd_soc_write(codec, WM8900_REG_POWER1, WM8900_REG_POWER1_BIAS_ENA | 0x1); } - reg = wm8900_read(codec, WM8900_REG_POWER1); - wm8900_write(codec, WM8900_REG_POWER1, + reg = snd_soc_read(codec, WM8900_REG_POWER1); + snd_soc_write(codec, WM8900_REG_POWER1, (reg & WM8900_REG_POWER1_FLL_ENA) | WM8900_REG_POWER1_BIAS_ENA | 0x1); - wm8900_write(codec, WM8900_REG_POWER2, + snd_soc_write(codec, WM8900_REG_POWER2, WM8900_REG_POWER2_SYSCLK_ENA); - wm8900_write(codec, WM8900_REG_POWER3, 0); + snd_soc_write(codec, WM8900_REG_POWER3, 0); break; case SND_SOC_BIAS_OFF: /* Startup bias enable */ - reg = wm8900_read(codec, WM8900_REG_POWER1); - wm8900_write(codec, WM8900_REG_POWER1, + reg = snd_soc_read(codec, WM8900_REG_POWER1); + snd_soc_write(codec, WM8900_REG_POWER1, reg & WM8900_REG_POWER1_STARTUP_BIAS_ENA); - wm8900_write(codec, WM8900_REG_ADDCTL, + snd_soc_write(codec, WM8900_REG_ADDCTL, WM8900_REG_ADDCTL_BIAS_SRC | WM8900_REG_ADDCTL_VMID_SOFTST); /* Discharge caps */ - wm8900_write(codec, WM8900_REG_POWER1, + snd_soc_write(codec, WM8900_REG_POWER1, WM8900_REG_POWER1_STARTUP_BIAS_ENA); schedule_timeout_interruptible(msecs_to_jiffies(500)); /* Remove clamp */ - wm8900_write(codec, WM8900_REG_HPCTL1, 0); + snd_soc_write(codec, WM8900_REG_HPCTL1, 0); /* Power down */ - wm8900_write(codec, WM8900_REG_ADDCTL, 0); - wm8900_write(codec, WM8900_REG_POWER1, 0); - wm8900_write(codec, WM8900_REG_POWER2, 0); - wm8900_write(codec, WM8900_REG_POWER3, 0); + snd_soc_write(codec, WM8900_REG_ADDCTL, 0); + snd_soc_write(codec, WM8900_REG_POWER1, 0); + snd_soc_write(codec, WM8900_REG_POWER2, 0); + snd_soc_write(codec, WM8900_REG_POWER3, 0); /* Need to let things settle before stopping the clock * to ensure that restart works, see "Stopping the * master clock" in the datasheet. */ schedule_timeout_interruptible(msecs_to_jiffies(1)); - wm8900_write(codec, WM8900_REG_POWER2, + snd_soc_write(codec, WM8900_REG_POWER2, WM8900_REG_POWER2_SYSCLK_ENA); break; } @@ -1264,7 +1184,7 @@ static int wm8900_resume(struct platform_device *pdev) if (cache) { for (i = 0; i < WM8900_MAXREG; i++) - wm8900_write(codec, i, cache[i]); + snd_soc_write(codec, i, cache[i]); kfree(cache); } else dev_err(&pdev->dev, "Unable to allocate register cache\n"); @@ -1297,16 +1217,20 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, codec->name = "WM8900"; codec->owner = THIS_MODULE; - codec->read = wm8900_read; - codec->write = wm8900_write; codec->dai = &wm8900_dai; codec->num_dai = 1; - codec->hw_write = (hw_write_t)i2c_master_send; codec->control_data = i2c; codec->set_bias_level = wm8900_set_bias_level; + codec->volatile_register = wm8900_volatile_register; codec->dev = &i2c->dev; - reg = wm8900_read(codec, WM8900_REG_ID); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + reg = snd_soc_read(codec, WM8900_REG_ID); if (reg != 0x8900) { dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg); ret = -ENODEV; @@ -1314,7 +1238,7 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, } /* Read back from the chip */ - reg = wm8900_chip_read(codec, WM8900_REG_POWER1); + reg = snd_soc_read(codec, WM8900_REG_POWER1); reg = (reg >> 12) & 0xf; dev_info(&i2c->dev, "WM8900 revision %d\n", reg); @@ -1324,29 +1248,29 @@ static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the volume update bits */ - wm8900_write(codec, WM8900_REG_LINVOL, - wm8900_read(codec, WM8900_REG_LINVOL) | 0x100); - wm8900_write(codec, WM8900_REG_RINVOL, - wm8900_read(codec, WM8900_REG_RINVOL) | 0x100); - wm8900_write(codec, WM8900_REG_LOUT1CTL, - wm8900_read(codec, WM8900_REG_LOUT1CTL) | 0x100); - wm8900_write(codec, WM8900_REG_ROUT1CTL, - wm8900_read(codec, WM8900_REG_ROUT1CTL) | 0x100); - wm8900_write(codec, WM8900_REG_LOUT2CTL, - wm8900_read(codec, WM8900_REG_LOUT2CTL) | 0x100); - wm8900_write(codec, WM8900_REG_ROUT2CTL, - wm8900_read(codec, WM8900_REG_ROUT2CTL) | 0x100); - wm8900_write(codec, WM8900_REG_LDAC_DV, - wm8900_read(codec, WM8900_REG_LDAC_DV) | 0x100); - wm8900_write(codec, WM8900_REG_RDAC_DV, - wm8900_read(codec, WM8900_REG_RDAC_DV) | 0x100); - wm8900_write(codec, WM8900_REG_LADC_DV, - wm8900_read(codec, WM8900_REG_LADC_DV) | 0x100); - wm8900_write(codec, WM8900_REG_RADC_DV, - wm8900_read(codec, WM8900_REG_RADC_DV) | 0x100); + snd_soc_write(codec, WM8900_REG_LINVOL, + snd_soc_read(codec, WM8900_REG_LINVOL) | 0x100); + snd_soc_write(codec, WM8900_REG_RINVOL, + snd_soc_read(codec, WM8900_REG_RINVOL) | 0x100); + snd_soc_write(codec, WM8900_REG_LOUT1CTL, + snd_soc_read(codec, WM8900_REG_LOUT1CTL) | 0x100); + snd_soc_write(codec, WM8900_REG_ROUT1CTL, + snd_soc_read(codec, WM8900_REG_ROUT1CTL) | 0x100); + snd_soc_write(codec, WM8900_REG_LOUT2CTL, + snd_soc_read(codec, WM8900_REG_LOUT2CTL) | 0x100); + snd_soc_write(codec, WM8900_REG_ROUT2CTL, + snd_soc_read(codec, WM8900_REG_ROUT2CTL) | 0x100); + snd_soc_write(codec, WM8900_REG_LDAC_DV, + snd_soc_read(codec, WM8900_REG_LDAC_DV) | 0x100); + snd_soc_write(codec, WM8900_REG_RDAC_DV, + snd_soc_read(codec, WM8900_REG_RDAC_DV) | 0x100); + snd_soc_write(codec, WM8900_REG_LADC_DV, + snd_soc_read(codec, WM8900_REG_LADC_DV) | 0x100); + snd_soc_write(codec, WM8900_REG_RADC_DV, + snd_soc_read(codec, WM8900_REG_RADC_DV) | 0x100); /* Set the DAC and mixer output bias */ - wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81); + snd_soc_write(codec, WM8900_REG_OUTBIASCTL, 0x81); wm8900_dai.dev = &i2c->dev; @@ -1388,6 +1312,21 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8900_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8900_i2c_suspend NULL +#define wm8900_i2c_resume NULL +#endif + static const struct i2c_device_id wm8900_i2c_id[] = { { "wm8900", 0 }, { } @@ -1401,6 +1340,8 @@ static struct i2c_driver wm8900_i2c_driver = { }, .probe = wm8900_i2c_probe, .remove = __devexit_p(wm8900_i2c_remove), + .suspend = wm8900_i2c_suspend, + .resume = wm8900_i2c_resume, .id_table = wm8900_i2c_id, }; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index e8d2e3e14c45..fe1307b500cf 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -225,94 +225,18 @@ struct wm8903_priv { struct snd_pcm_substream *slave_substream; }; - -static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - - BUG_ON(reg >= ARRAY_SIZE(wm8903_reg_defaults)); - - return cache[reg]; -} - -static unsigned int wm8903_hw_read(struct snd_soc_codec *codec, u8 reg) -{ - struct i2c_msg xfer[2]; - u16 data; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - pr_err("i2c_transfer returned %d\n", ret); - return 0; - } - - return (data >> 8) | ((data & 0xff) << 8); -} - -static unsigned int wm8903_read(struct snd_soc_codec *codec, - unsigned int reg) +static int wm8903_volatile_register(unsigned int reg) { switch (reg) { case WM8903_SW_RESET_AND_ID: case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: - return wm8903_hw_read(codec, reg); + return 1; default: - return wm8903_read_reg_cache(codec, reg); - } -} - -static void wm8903_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - - BUG_ON(reg >= ARRAY_SIZE(wm8903_reg_defaults)); - - switch (reg) { - case WM8903_SW_RESET_AND_ID: - case WM8903_REVISION_NUMBER: - break; - - default: - cache[reg] = value; - break; - } -} - -static int wm8903_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - - wm8903_write_reg_cache(codec, reg, value); - - /* Data format is 1 byte of address followed by 2 bytes of data */ - data[0] = reg; - data[1] = (value >> 8) & 0xff; - data[2] = value & 0xff; - - if (codec->hw_write(codec->control_data, data, 3) == 2) return 0; - else - return -EIO; + } } static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) @@ -323,13 +247,13 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) BUG_ON(start > 48); /* Enable the sequencer */ - reg[0] = wm8903_read(codec, WM8903_WRITE_SEQUENCER_0); + reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0); reg[0] |= WM8903_WSEQ_ENA; - wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]); + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]); dev_dbg(&i2c->dev, "Starting sequence at %d\n", start); - wm8903_write(codec, WM8903_WRITE_SEQUENCER_3, + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3, start | WM8903_WSEQ_START); /* Wait for it to complete. If we have the interrupt wired up then @@ -339,13 +263,13 @@ static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start) do { msleep(10); - reg[4] = wm8903_read(codec, WM8903_WRITE_SEQUENCER_4); + reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4); } while (reg[4] & WM8903_WSEQ_BUSY); dev_dbg(&i2c->dev, "Sequence complete\n"); /* Disable the sequencer again */ - wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0] & ~WM8903_WSEQ_ENA); return 0; @@ -357,12 +281,12 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache) /* There really ought to be something better we can do here :/ */ for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++) - cache[i] = wm8903_hw_read(codec, i); + cache[i] = codec->hw_read(codec, i); } static void wm8903_reset(struct snd_soc_codec *codec) { - wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0); + snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0); memcpy(codec->reg_cache, wm8903_reg_defaults, sizeof(wm8903_reg_defaults)); } @@ -423,52 +347,52 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, } if (event & SND_SOC_DAPM_PRE_PMU) { - val = wm8903_read(codec, reg); + val = snd_soc_read(codec, reg); /* Short the output */ val &= ~(WM8903_OUTPUT_SHORT << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); } if (event & SND_SOC_DAPM_POST_PMU) { - val = wm8903_read(codec, reg); + val = snd_soc_read(codec, reg); val |= (WM8903_OUTPUT_IN << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); val |= (WM8903_OUTPUT_INT << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); /* Turn on the output ENA_OUTP */ val |= (WM8903_OUTPUT_OUT << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); /* Enable the DC servo */ - dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0); + dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); dcs_reg |= dcs_bit; - wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg); + snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); /* Remove the short */ val |= (WM8903_OUTPUT_SHORT << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); } if (event & SND_SOC_DAPM_PRE_PMD) { - val = wm8903_read(codec, reg); + val = snd_soc_read(codec, reg); /* Short the output */ val &= ~(WM8903_OUTPUT_SHORT << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); /* Disable the DC servo */ - dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0); + dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); dcs_reg &= ~dcs_bit; - wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg); + snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); /* Then disable the intermediate and output stages */ val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | WM8903_OUTPUT_IN) << shift); - wm8903_write(codec, reg, val); + snd_soc_write(codec, reg, val); } return 0; @@ -492,13 +416,13 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, u16 reg; int ret; - reg = wm8903_read(codec, WM8903_CLASS_W_0); + reg = snd_soc_read(codec, WM8903_CLASS_W_0); /* Turn it off if we're about to enable bypass */ if (ucontrol->value.integer.value[0]) { if (wm8903->class_w_users == 0) { dev_dbg(&i2c->dev, "Disabling Class W\n"); - wm8903_write(codec, WM8903_CLASS_W_0, reg & + snd_soc_write(codec, WM8903_CLASS_W_0, reg & ~(WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V)); } wm8903->class_w_users++; @@ -511,7 +435,7 @@ static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, if (!ucontrol->value.integer.value[0]) { if (wm8903->class_w_users == 1) { dev_dbg(&i2c->dev, "Enabling Class W\n"); - wm8903_write(codec, WM8903_CLASS_W_0, reg | + snd_soc_write(codec, WM8903_CLASS_W_0, reg | WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V); } wm8903->class_w_users--; @@ -715,8 +639,6 @@ SOC_ENUM("DAC Soft Mute Rate", soft_mute), SOC_ENUM("DAC Mute Mode", mute_mode), SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0), SOC_ENUM("DAC De-emphasis", dac_deemphasis), -SOC_SINGLE("DAC Sloping Stopband Filter Switch", - WM8903_DAC_DIGITAL_1, 11, 1, 0), SOC_ENUM("DAC Companding Mode", dac_companding), SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0), @@ -1011,55 +933,55 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: - reg = wm8903_read(codec, WM8903_VMID_CONTROL_0); + reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); reg &= ~(WM8903_VMID_RES_MASK); reg |= WM8903_VMID_RES_50K; - wm8903_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { - wm8903_write(codec, WM8903_CLOCK_RATES_2, + snd_soc_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); /* Change DC servo dither level in startup sequence */ - wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11); - wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257); - wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2); + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11); + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257); + snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2); wm8903_run_sequence(codec, 0); wm8903_sync_reg_cache(codec, codec->reg_cache); /* Enable low impedence charge pump output */ - reg = wm8903_read(codec, + reg = snd_soc_read(codec, WM8903_CONTROL_INTERFACE_TEST_1); - wm8903_write(codec, WM8903_CONTROL_INTERFACE_TEST_1, + snd_soc_write(codec, WM8903_CONTROL_INTERFACE_TEST_1, reg | WM8903_TEST_KEY); - reg2 = wm8903_read(codec, WM8903_CHARGE_PUMP_TEST_1); - wm8903_write(codec, WM8903_CHARGE_PUMP_TEST_1, + reg2 = snd_soc_read(codec, WM8903_CHARGE_PUMP_TEST_1); + snd_soc_write(codec, WM8903_CHARGE_PUMP_TEST_1, reg2 | WM8903_CP_SW_KELVIN_MODE_MASK); - wm8903_write(codec, WM8903_CONTROL_INTERFACE_TEST_1, + snd_soc_write(codec, WM8903_CONTROL_INTERFACE_TEST_1, reg); /* By default no bypass paths are enabled so * enable Class W support. */ dev_dbg(&i2c->dev, "Enabling Class W\n"); - wm8903_write(codec, WM8903_CLASS_W_0, reg | + snd_soc_write(codec, WM8903_CLASS_W_0, reg | WM8903_CP_DYN_FREQ | WM8903_CP_DYN_V); } - reg = wm8903_read(codec, WM8903_VMID_CONTROL_0); + reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0); reg &= ~(WM8903_VMID_RES_MASK); reg |= WM8903_VMID_RES_250K; - wm8903_write(codec, WM8903_VMID_CONTROL_0, reg); + snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg); break; case SND_SOC_BIAS_OFF: wm8903_run_sequence(codec, 32); - reg = wm8903_read(codec, WM8903_CLOCK_RATES_2); + reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2); reg &= ~WM8903_CLK_SYS_ENA; - wm8903_write(codec, WM8903_CLOCK_RATES_2, reg); + snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg); break; } @@ -1083,7 +1005,7 @@ static int wm8903_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 aif1 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_1); + u16 aif1 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_1); aif1 &= ~(WM8903_LRCLK_DIR | WM8903_BCLK_DIR | WM8903_AIF_FMT_MASK | WM8903_AIF_LRCLK_INV | WM8903_AIF_BCLK_INV); @@ -1161,7 +1083,7 @@ static int wm8903_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM8903_AUDIO_INTERFACE_1, aif1); return 0; } @@ -1171,14 +1093,14 @@ static int wm8903_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_codec *codec = codec_dai->codec; u16 reg; - reg = wm8903_read(codec, WM8903_DAC_DIGITAL_1); + reg = snd_soc_read(codec, WM8903_DAC_DIGITAL_1); if (mute) reg |= WM8903_DAC_MUTE; else reg &= ~WM8903_DAC_MUTE; - wm8903_write(codec, WM8903_DAC_DIGITAL_1, reg); + snd_soc_write(codec, WM8903_DAC_DIGITAL_1, reg); return 0; } @@ -1368,17 +1290,24 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, int cur_val; int clk_sys; - u16 aif1 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_1); - u16 aif2 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_2); - u16 aif3 = wm8903_read(codec, WM8903_AUDIO_INTERFACE_3); - u16 clock0 = wm8903_read(codec, WM8903_CLOCK_RATES_0); - u16 clock1 = wm8903_read(codec, WM8903_CLOCK_RATES_1); + u16 aif1 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_1); + u16 aif2 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_2); + u16 aif3 = snd_soc_read(codec, WM8903_AUDIO_INTERFACE_3); + u16 clock0 = snd_soc_read(codec, WM8903_CLOCK_RATES_0); + u16 clock1 = snd_soc_read(codec, WM8903_CLOCK_RATES_1); + u16 dac_digital1 = snd_soc_read(codec, WM8903_DAC_DIGITAL_1); if (substream == wm8903->slave_substream) { dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); return 0; } + /* Enable sloping stopband filter for low sample rates */ + if (fs <= 24000) + dac_digital1 |= WM8903_DAC_SB_FILT; + else + dac_digital1 &= ~WM8903_DAC_SB_FILT; + /* Configure sample rate logic for DSP - choose nearest rate */ dsp_config = 0; best_val = abs(sample_rates[dsp_config].rate - fs); @@ -1498,11 +1427,12 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, aif2 |= bclk_divs[bclk_div].div; aif3 |= bclk / fs; - wm8903_write(codec, WM8903_CLOCK_RATES_0, clock0); - wm8903_write(codec, WM8903_CLOCK_RATES_1, clock1); - wm8903_write(codec, WM8903_AUDIO_INTERFACE_1, aif1); - wm8903_write(codec, WM8903_AUDIO_INTERFACE_2, aif2); - wm8903_write(codec, WM8903_AUDIO_INTERFACE_3, aif3); + snd_soc_write(codec, WM8903_CLOCK_RATES_0, clock0); + snd_soc_write(codec, WM8903_CLOCK_RATES_1, clock1); + snd_soc_write(codec, WM8903_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM8903_AUDIO_INTERFACE_2, aif2); + snd_soc_write(codec, WM8903_AUDIO_INTERFACE_3, aif3); + snd_soc_write(codec, WM8903_DAC_DIGITAL_1, dac_digital1); return 0; } @@ -1587,7 +1517,7 @@ static int wm8903_resume(struct platform_device *pdev) if (tmp_cache) { for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) if (tmp_cache[i] != reg_cache[i]) - wm8903_write(codec, i, tmp_cache[i]); + snd_soc_write(codec, i, tmp_cache[i]); } else { dev_err(&i2c->dev, "Failed to allocate temporary cache\n"); } @@ -1618,9 +1548,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; codec->name = "WM8903"; codec->owner = THIS_MODULE; - codec->read = wm8903_read; - codec->write = wm8903_write; - codec->hw_write = (hw_write_t)i2c_master_send; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8903_set_bias_level; codec->dai = &wm8903_dai; @@ -1628,18 +1555,25 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache); codec->reg_cache = &wm8903->reg_cache[0]; codec->private_data = wm8903; + codec->volatile_register = wm8903_volatile_register; i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + val = snd_soc_read(codec, WM8903_SW_RESET_AND_ID); if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) { dev_err(&i2c->dev, "Device with ID register %x is not a WM8903\n", val); return -ENODEV; } - val = wm8903_read(codec, WM8903_REVISION_NUMBER); + val = snd_soc_read(codec, WM8903_REVISION_NUMBER); dev_info(&i2c->dev, "WM8903 revision %d\n", val & WM8903_CHIP_REV_MASK); @@ -1649,35 +1583,35 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch volume update bits */ - val = wm8903_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT); + val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT); val |= WM8903_ADCVU; - wm8903_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val); - wm8903_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val); + snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val); + snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val); - val = wm8903_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT); + val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT); val |= WM8903_DACVU; - wm8903_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val); - wm8903_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val); + snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val); + snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val); - val = wm8903_read(codec, WM8903_ANALOGUE_OUT1_LEFT); + val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT); val |= WM8903_HPOUTVU; - wm8903_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val); - wm8903_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val); + snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val); + snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val); - val = wm8903_read(codec, WM8903_ANALOGUE_OUT2_LEFT); + val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT); val |= WM8903_LINEOUTVU; - wm8903_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val); - wm8903_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val); + snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val); + snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val); - val = wm8903_read(codec, WM8903_ANALOGUE_OUT3_LEFT); + val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT); val |= WM8903_SPKVU; - wm8903_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val); - wm8903_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); + snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val); + snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); /* Enable DAC soft mute by default */ - val = wm8903_read(codec, WM8903_DAC_DIGITAL_1); + val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1); val |= WM8903_DAC_MUTEMODE; - wm8903_write(codec, WM8903_DAC_DIGITAL_1, val); + snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val); wm8903_dai.dev = &i2c->dev; wm8903_codec = codec; @@ -1721,6 +1655,21 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8903_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8903_i2c_suspend NULL +#define wm8903_i2c_resume NULL +#endif + /* i2c codec control layer */ static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, @@ -1735,6 +1684,8 @@ static struct i2c_driver wm8903_i2c_driver = { }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), + .suspend = wm8903_i2c_suspend, + .resume = wm8903_i2c_resume, .id_table = wm8903_i2c_id, }; diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b8e17d6bc1f7..da97aae475a2 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -106,50 +106,6 @@ static u16 wm8940_reg_defaults[] = { 0x0000, /* Mono Mixer Control */ }; -static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - - if (reg >= ARRAY_SIZE(wm8940_reg_defaults)) - return -1; - - return cache[reg]; -} - -static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - - if (reg >= ARRAY_SIZE(wm8940_reg_defaults)) - return -1; - - cache[reg] = value; - - return 0; -} - -static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; - u8 data[3] = { reg, - (value & 0xff00) >> 8, - (value & 0x00ff) - }; - - wm8940_write_reg_cache(codec, reg, value); - - ret = codec->hw_write(codec->control_data, data, 3); - - if (ret < 0) - return ret; - else if (ret != 3) - return -EIO; - return 0; -} - static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; static const struct soc_enum wm8940_adc_companding_enum = SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding); @@ -348,14 +304,14 @@ error_ret: return ret; } -#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0); +#define wm8940_reset(c) snd_soc_write(c, WM8940_SOFTRESET, 0); static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67; - u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe; + u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFE67; + u16 clk = snd_soc_read(codec, WM8940_CLOCK) & 0x1fe; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -366,7 +322,7 @@ static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, default: return -EINVAL; } - wm8940_write(codec, WM8940_CLOCK, clk); + snd_soc_write(codec, WM8940_CLOCK, clk); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -399,7 +355,7 @@ static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - wm8940_write(codec, WM8940_IFACE, iface); + snd_soc_write(codec, WM8940_IFACE, iface); return 0; } @@ -411,9 +367,9 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F; - u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1; - u16 companding = wm8940_read_reg_cache(codec, + u16 iface = snd_soc_read(codec, WM8940_IFACE) & 0xFD9F; + u16 addcntrl = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFF1; + u16 companding = snd_soc_read(codec, WM8940_COMPANDINGCTL) & 0xFFDF; int ret; @@ -442,7 +398,7 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_RATE_48000: break; } - ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl); + ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl); if (ret) goto error_ret; @@ -462,10 +418,10 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, iface |= (3 << 5); break; } - ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding); + ret = snd_soc_write(codec, WM8940_COMPANDINGCTL, companding); if (ret) goto error_ret; - ret = wm8940_write(codec, WM8940_IFACE, iface); + ret = snd_soc_write(codec, WM8940_IFACE, iface); error_ret: return ret; @@ -474,19 +430,19 @@ error_ret: static int wm8940_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf; + u16 mute_reg = snd_soc_read(codec, WM8940_DAC) & 0xffbf; if (mute) mute_reg |= 0x40; - return wm8940_write(codec, WM8940_DAC, mute_reg); + return snd_soc_write(codec, WM8940_DAC, mute_reg); } static int wm8940_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { u16 val; - u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0; + u16 pwr_reg = snd_soc_read(codec, WM8940_POWER1) & 0x1F0; int ret = 0; switch (level) { @@ -494,26 +450,26 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec, /* ensure bufioen and biasen */ pwr_reg |= (1 << 2) | (1 << 3); /* Enable thermal shutdown */ - val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); - ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2); + val = snd_soc_read(codec, WM8940_OUTPUTCTL); + ret = snd_soc_write(codec, WM8940_OUTPUTCTL, val | 0x2); if (ret) break; /* set vmid to 75k */ - ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1); break; case SND_SOC_BIAS_PREPARE: /* ensure bufioen and biasen */ pwr_reg |= (1 << 2) | (1 << 3); - ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1); + ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1); break; case SND_SOC_BIAS_STANDBY: /* ensure bufioen and biasen */ pwr_reg |= (1 << 2) | (1 << 3); /* set vmid to 300k for standby */ - ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2); + ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x2); break; case SND_SOC_BIAS_OFF: - ret = wm8940_write(codec, WM8940_POWER1, pwr_reg); + ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg); break; } @@ -587,36 +543,36 @@ static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, u16 reg; /* Turn off PLL */ - reg = wm8940_read_reg_cache(codec, WM8940_POWER1); - wm8940_write(codec, WM8940_POWER1, reg & 0x1df); + reg = snd_soc_read(codec, WM8940_POWER1); + snd_soc_write(codec, WM8940_POWER1, reg & 0x1df); if (freq_in == 0 || freq_out == 0) { /* Clock CODEC directly from MCLK */ - reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); - wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff); + reg = snd_soc_read(codec, WM8940_CLOCK); + snd_soc_write(codec, WM8940_CLOCK, reg & 0x0ff); /* Pll power down */ - wm8940_write(codec, WM8940_PLLN, (1 << 7)); + snd_soc_write(codec, WM8940_PLLN, (1 << 7)); return 0; } /* Pll is followed by a frequency divide by 4 */ pll_factors(freq_out*4, freq_in); if (pll_div.k) - wm8940_write(codec, WM8940_PLLN, + snd_soc_write(codec, WM8940_PLLN, (pll_div.pre_scale << 4) | pll_div.n | (1 << 6)); else /* No factional component */ - wm8940_write(codec, WM8940_PLLN, + snd_soc_write(codec, WM8940_PLLN, (pll_div.pre_scale << 4) | pll_div.n); - wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18); - wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff); - wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8940_PLLK1, pll_div.k >> 18); + snd_soc_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff); /* Enable the PLL */ - reg = wm8940_read_reg_cache(codec, WM8940_POWER1); - wm8940_write(codec, WM8940_POWER1, reg | 0x020); + reg = snd_soc_read(codec, WM8940_POWER1); + snd_soc_write(codec, WM8940_POWER1, reg | 0x020); /* Run CODEC from PLL instead of MCLK */ - reg = wm8940_read_reg_cache(codec, WM8940_CLOCK); - wm8940_write(codec, WM8940_CLOCK, reg | 0x100); + reg = snd_soc_read(codec, WM8940_CLOCK); + snd_soc_write(codec, WM8940_CLOCK, reg | 0x100); return 0; } @@ -648,16 +604,16 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8940_BCLKDIV: - reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3; - ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2)); + reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3; + ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2)); break; case WM8940_MCLKDIV: - reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F; - ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5)); + reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFF1F; + ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 5)); break; case WM8940_OPCLKDIV: - reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF; - ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + reg = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFCF; + ret = snd_soc_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); break; } return ret; @@ -808,7 +764,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8940 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940); -static int wm8940_register(struct wm8940_priv *wm8940) +static int wm8940_register(struct wm8940_priv *wm8940, + enum snd_soc_control_type control) { struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data; struct snd_soc_codec *codec = &wm8940->codec; @@ -825,8 +782,6 @@ static int wm8940_register(struct wm8940_priv *wm8940) codec->private_data = wm8940; codec->name = "WM8940"; codec->owner = THIS_MODULE; - codec->read = wm8940_read_reg_cache; - codec->write = wm8940_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8940_set_bias_level; codec->dai = &wm8940_dai; @@ -834,6 +789,12 @@ static int wm8940_register(struct wm8940_priv *wm8940) codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults); codec->reg_cache = &wm8940->reg_cache; + ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); + if (ret == 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + memcpy(codec->reg_cache, wm8940_reg_defaults, sizeof(wm8940_reg_defaults)); @@ -847,15 +808,15 @@ static int wm8940_register(struct wm8940_priv *wm8940) wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = wm8940_write(codec, WM8940_POWER1, 0x180); + ret = snd_soc_write(codec, WM8940_POWER1, 0x180); if (ret < 0) return ret; if (!pdata) dev_warn(codec->dev, "No platform data supplied\n"); else { - reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL); - ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi); + reg = snd_soc_read(codec, WM8940_OUTPUTCTL); + ret = snd_soc_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi); if (ret < 0) return ret; } @@ -904,7 +865,7 @@ static int wm8940_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8940_register(wm8940); + return wm8940_register(wm8940, SND_SOC_I2C); } static int __devexit wm8940_i2c_remove(struct i2c_client *client) @@ -916,6 +877,21 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8940_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8940_i2c_suspend NULL +#define wm8940_i2c_resume NULL +#endif + static const struct i2c_device_id wm8940_i2c_id[] = { { "wm8940", 0 }, { } @@ -929,6 +905,8 @@ static struct i2c_driver wm8940_i2c_driver = { }, .probe = wm8940_i2c_probe, .remove = __devexit_p(wm8940_i2c_remove), + .suspend = wm8940_i2c_suspend, + .resume = wm8940_i2c_resume, .id_table = wm8940_i2c_id, }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e224d8add170..f59703be61c8 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -69,61 +69,7 @@ struct wm8960_priv { struct snd_soc_codec codec; }; -/* - * read wm8960 register cache - */ -static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg == WM8960_RESET) - return 0; - if (reg >= WM8960_CACHEREGNUM) - return -1; - return cache[reg]; -} - -/* - * write wm8960 register cache - */ -static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec, - u16 reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg >= WM8960_CACHEREGNUM) - return; - cache[reg] = value; -} - -static inline unsigned int wm8960_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - return wm8960_read_reg_cache(codec, reg); -} - -/* - * write to the WM8960 register space - */ -static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8960 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8960_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0) +#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) /* enumerated controls */ static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -420,7 +366,7 @@ static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, } /* set iface */ - wm8960_write(codec, WM8960_IFACE1, iface); + snd_soc_write(codec, WM8960_IFACE1, iface); return 0; } @@ -431,7 +377,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3; + u16 iface = snd_soc_read(codec, WM8960_IFACE1) & 0xfff3; /* bit size */ switch (params_format(params)) { @@ -446,19 +392,19 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, } /* set iface */ - wm8960_write(codec, WM8960_IFACE1, iface); + snd_soc_write(codec, WM8960_IFACE1, iface); return 0; } static int wm8960_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7; if (mute) - wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8); else - wm8960_write(codec, WM8960_DACCTL1, mute_reg); + snd_soc_write(codec, WM8960_DACCTL1, mute_reg); return 0; } @@ -474,16 +420,16 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg &= ~0x180; reg |= 0x80; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, + snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); @@ -491,43 +437,43 @@ static int wm8960_set_bias_level(struct snd_soc_codec *codec, reg = WM8960_DISOP; if (pdata) reg |= pdata->dres << 4; - wm8960_write(codec, WM8960_APOP2, reg); + snd_soc_write(codec, WM8960_APOP2, reg); msleep(400); - wm8960_write(codec, WM8960_APOP2, 0); + snd_soc_write(codec, WM8960_APOP2, 0); /* Enable & ramp VMID at 2x50k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg |= 0x80; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); msleep(100); /* Enable VREF */ - wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF); + snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF); /* Disable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN); + snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN); } /* Set VMID to 2x250k */ - reg = wm8960_read(codec, WM8960_POWER1); + reg = snd_soc_read(codec, WM8960_POWER1); reg &= ~0x180; reg |= 0x100; - wm8960_write(codec, WM8960_POWER1, reg); + snd_soc_write(codec, WM8960_POWER1, reg); break; case SND_SOC_BIAS_OFF: /* Enable anti-pop features */ - wm8960_write(codec, WM8960_APOP1, + snd_soc_write(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Disable VMID and VREF, let them discharge */ - wm8960_write(codec, WM8960_POWER1, 0); + snd_soc_write(codec, WM8960_POWER1, 0); msleep(600); - wm8960_write(codec, WM8960_APOP1, 0); + snd_soc_write(codec, WM8960_APOP1, 0); break; } @@ -610,33 +556,33 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, /* Disable the PLL: even if we are changing the frequency the * PLL needs to be disabled while we do so. */ - wm8960_write(codec, WM8960_CLOCK1, - wm8960_read(codec, WM8960_CLOCK1) & ~1); - wm8960_write(codec, WM8960_POWER2, - wm8960_read(codec, WM8960_POWER2) & ~1); + snd_soc_write(codec, WM8960_CLOCK1, + snd_soc_read(codec, WM8960_CLOCK1) & ~1); + snd_soc_write(codec, WM8960_POWER2, + snd_soc_read(codec, WM8960_POWER2) & ~1); if (!freq_in || !freq_out) return 0; - reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f; + reg = snd_soc_read(codec, WM8960_PLL1) & ~0x3f; reg |= pll_div.pre_div << 4; reg |= pll_div.n; if (pll_div.k) { reg |= 0x20; - wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); } - wm8960_write(codec, WM8960_PLL1, reg); + snd_soc_write(codec, WM8960_PLL1, reg); /* Turn it on */ - wm8960_write(codec, WM8960_POWER2, - wm8960_read(codec, WM8960_POWER2) | 1); + snd_soc_write(codec, WM8960_POWER2, + snd_soc_read(codec, WM8960_POWER2) | 1); msleep(250); - wm8960_write(codec, WM8960_CLOCK1, - wm8960_read(codec, WM8960_CLOCK1) | 1); + snd_soc_write(codec, WM8960_CLOCK1, + snd_soc_read(codec, WM8960_CLOCK1) | 1); return 0; } @@ -649,28 +595,28 @@ static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8960_SYSCLKSEL: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1fe; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_SYSCLKDIV: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1f9; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_DACDIV: - reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7; - wm8960_write(codec, WM8960_CLOCK1, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK1) & 0x1c7; + snd_soc_write(codec, WM8960_CLOCK1, reg | div); break; case WM8960_OPCLKDIV: - reg = wm8960_read(codec, WM8960_PLL1) & 0x03f; - wm8960_write(codec, WM8960_PLL1, reg | div); + reg = snd_soc_read(codec, WM8960_PLL1) & 0x03f; + snd_soc_write(codec, WM8960_PLL1, reg | div); break; case WM8960_DCLKDIV: - reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f; - wm8960_write(codec, WM8960_CLOCK2, reg | div); + reg = snd_soc_read(codec, WM8960_CLOCK2) & 0x03f; + snd_soc_write(codec, WM8960_CLOCK2, reg | div); break; case WM8960_TOCLKSEL: - reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd; - wm8960_write(codec, WM8960_ADDCTL1, reg | div); + reg = snd_soc_read(codec, WM8960_ADDCTL1) & 0x1fd; + snd_soc_write(codec, WM8960_ADDCTL1, reg | div); break; default: return -EINVAL; @@ -801,7 +747,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8960 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960); -static int wm8960_register(struct wm8960_priv *wm8960) +static int wm8960_register(struct wm8960_priv *wm8960, + enum snd_soc_control_type control) { struct wm8960_data *pdata = wm8960->codec.dev->platform_data; struct snd_soc_codec *codec = &wm8960->codec; @@ -810,7 +757,8 @@ static int wm8960_register(struct wm8960_priv *wm8960) if (wm8960_codec) { dev_err(codec->dev, "Another WM8960 is registered\n"); - return -EINVAL; + ret = -EINVAL; + goto err; } if (!pdata) { @@ -829,8 +777,6 @@ static int wm8960_register(struct wm8960_priv *wm8960) codec->private_data = wm8960; codec->name = "WM8960"; codec->owner = THIS_MODULE; - codec->read = wm8960_read_reg_cache; - codec->write = wm8960_write; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8960_set_bias_level; codec->dai = &wm8960_dai; @@ -840,10 +786,16 @@ static int wm8960_register(struct wm8960_priv *wm8960) memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8960_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } wm8960_dai.dev = codec->dev; @@ -851,43 +803,48 @@ static int wm8960_register(struct wm8960_priv *wm8960) wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = wm8960_read(codec, WM8960_LINVOL); - wm8960_write(codec, WM8960_LINVOL, reg | 0x100); - reg = wm8960_read(codec, WM8960_RINVOL); - wm8960_write(codec, WM8960_RINVOL, reg | 0x100); - reg = wm8960_read(codec, WM8960_LADC); - wm8960_write(codec, WM8960_LADC, reg | 0x100); - reg = wm8960_read(codec, WM8960_RADC); - wm8960_write(codec, WM8960_RADC, reg | 0x100); - reg = wm8960_read(codec, WM8960_LDAC); - wm8960_write(codec, WM8960_LDAC, reg | 0x100); - reg = wm8960_read(codec, WM8960_RDAC); - wm8960_write(codec, WM8960_RDAC, reg | 0x100); - reg = wm8960_read(codec, WM8960_LOUT1); - wm8960_write(codec, WM8960_LOUT1, reg | 0x100); - reg = wm8960_read(codec, WM8960_ROUT1); - wm8960_write(codec, WM8960_ROUT1, reg | 0x100); - reg = wm8960_read(codec, WM8960_LOUT2); - wm8960_write(codec, WM8960_LOUT2, reg | 0x100); - reg = wm8960_read(codec, WM8960_ROUT2); - wm8960_write(codec, WM8960_ROUT2, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LINVOL); + snd_soc_write(codec, WM8960_LINVOL, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RINVOL); + snd_soc_write(codec, WM8960_RINVOL, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LADC); + snd_soc_write(codec, WM8960_LADC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RADC); + snd_soc_write(codec, WM8960_RADC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LDAC); + snd_soc_write(codec, WM8960_LDAC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_RDAC); + snd_soc_write(codec, WM8960_RDAC, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LOUT1); + snd_soc_write(codec, WM8960_LOUT1, reg | 0x100); + reg = snd_soc_read(codec, WM8960_ROUT1); + snd_soc_write(codec, WM8960_ROUT1, reg | 0x100); + reg = snd_soc_read(codec, WM8960_LOUT2); + snd_soc_write(codec, WM8960_LOUT2, reg | 0x100); + reg = snd_soc_read(codec, WM8960_ROUT2); + snd_soc_write(codec, WM8960_ROUT2, reg | 0x100); wm8960_codec = codec; ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm8960_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); - snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8960); + return ret; } static void wm8960_unregister(struct wm8960_priv *wm8960) @@ -910,14 +867,13 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, return -ENOMEM; codec = &wm8960->codec; - codec->hw_write = (hw_write_t)i2c_master_send; i2c_set_clientdata(i2c, wm8960); codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8960_register(wm8960); + return wm8960_register(wm8960, SND_SOC_I2C); } static __devexit int wm8960_i2c_remove(struct i2c_client *client) @@ -927,6 +883,21 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8960_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8960_i2c_suspend NULL +#define wm8960_i2c_resume NULL +#endif + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -940,6 +911,8 @@ static struct i2c_driver wm8960_i2c_driver = { }, .probe = wm8960_i2c_probe, .remove = __devexit_p(wm8960_i2c_remove), + .suspend = wm8960_i2c_suspend, + .resume = wm8960_i2c_resume, .id_table = wm8960_i2c_id, }; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c new file mode 100644 index 000000000000..503032085899 --- /dev/null +++ b/sound/soc/codecs/wm8961.c @@ -0,0 +1,1265 @@ +/* + * wm8961.c -- WM8961 ALSA SoC Audio driver + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Currently unimplemented features: + * - ALC + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8961.h" + +#define WM8961_MAX_REGISTER 0xFC + +static u16 wm8961_reg_defaults[] = { + 0x009F, /* R0 - Left Input volume */ + 0x009F, /* R1 - Right Input volume */ + 0x0000, /* R2 - LOUT1 volume */ + 0x0000, /* R3 - ROUT1 volume */ + 0x0020, /* R4 - Clocking1 */ + 0x0008, /* R5 - ADC & DAC Control 1 */ + 0x0000, /* R6 - ADC & DAC Control 2 */ + 0x000A, /* R7 - Audio Interface 0 */ + 0x01F4, /* R8 - Clocking2 */ + 0x0000, /* R9 - Audio Interface 1 */ + 0x00FF, /* R10 - Left DAC volume */ + 0x00FF, /* R11 - Right DAC volume */ + 0x0000, /* R12 */ + 0x0000, /* R13 */ + 0x0040, /* R14 - Audio Interface 2 */ + 0x0000, /* R15 - Software Reset */ + 0x0000, /* R16 */ + 0x007B, /* R17 - ALC1 */ + 0x0000, /* R18 - ALC2 */ + 0x0032, /* R19 - ALC3 */ + 0x0000, /* R20 - Noise Gate */ + 0x00C0, /* R21 - Left ADC volume */ + 0x00C0, /* R22 - Right ADC volume */ + 0x0120, /* R23 - Additional control(1) */ + 0x0000, /* R24 - Additional control(2) */ + 0x0000, /* R25 - Pwr Mgmt (1) */ + 0x0000, /* R26 - Pwr Mgmt (2) */ + 0x0000, /* R27 - Additional Control (3) */ + 0x0000, /* R28 - Anti-pop */ + 0x0000, /* R29 */ + 0x005F, /* R30 - Clocking 3 */ + 0x0000, /* R31 */ + 0x0000, /* R32 - ADCL signal path */ + 0x0000, /* R33 - ADCR signal path */ + 0x0000, /* R34 */ + 0x0000, /* R35 */ + 0x0000, /* R36 */ + 0x0000, /* R37 */ + 0x0000, /* R38 */ + 0x0000, /* R39 */ + 0x0000, /* R40 - LOUT2 volume */ + 0x0000, /* R41 - ROUT2 volume */ + 0x0000, /* R42 */ + 0x0000, /* R43 */ + 0x0000, /* R44 */ + 0x0000, /* R45 */ + 0x0000, /* R46 */ + 0x0000, /* R47 - Pwr Mgmt (3) */ + 0x0023, /* R48 - Additional Control (4) */ + 0x0000, /* R49 - Class D Control 1 */ + 0x0000, /* R50 */ + 0x0003, /* R51 - Class D Control 2 */ + 0x0000, /* R52 */ + 0x0000, /* R53 */ + 0x0000, /* R54 */ + 0x0000, /* R55 */ + 0x0106, /* R56 - Clocking 4 */ + 0x0000, /* R57 - DSP Sidetone 0 */ + 0x0000, /* R58 - DSP Sidetone 1 */ + 0x0000, /* R59 */ + 0x0000, /* R60 - DC Servo 0 */ + 0x0000, /* R61 - DC Servo 1 */ + 0x0000, /* R62 */ + 0x015E, /* R63 - DC Servo 3 */ + 0x0010, /* R64 */ + 0x0010, /* R65 - DC Servo 5 */ + 0x0000, /* R66 */ + 0x0001, /* R67 */ + 0x0003, /* R68 - Analogue PGA Bias */ + 0x0000, /* R69 - Analogue HP 0 */ + 0x0060, /* R70 */ + 0x01FB, /* R71 - Analogue HP 2 */ + 0x0000, /* R72 - Charge Pump 1 */ + 0x0065, /* R73 */ + 0x005F, /* R74 */ + 0x0059, /* R75 */ + 0x006B, /* R76 */ + 0x0038, /* R77 */ + 0x000C, /* R78 */ + 0x000A, /* R79 */ + 0x006B, /* R80 */ + 0x0000, /* R81 */ + 0x0000, /* R82 - Charge Pump B */ + 0x0087, /* R83 */ + 0x0000, /* R84 */ + 0x005C, /* R85 */ + 0x0000, /* R86 */ + 0x0000, /* R87 - Write Sequencer 1 */ + 0x0000, /* R88 - Write Sequencer 2 */ + 0x0000, /* R89 - Write Sequencer 3 */ + 0x0000, /* R90 - Write Sequencer 4 */ + 0x0000, /* R91 - Write Sequencer 5 */ + 0x0000, /* R92 - Write Sequencer 6 */ + 0x0000, /* R93 - Write Sequencer 7 */ + 0x0000, /* R94 */ + 0x0000, /* R95 */ + 0x0000, /* R96 */ + 0x0000, /* R97 */ + 0x0000, /* R98 */ + 0x0000, /* R99 */ + 0x0000, /* R100 */ + 0x0000, /* R101 */ + 0x0000, /* R102 */ + 0x0000, /* R103 */ + 0x0000, /* R104 */ + 0x0000, /* R105 */ + 0x0000, /* R106 */ + 0x0000, /* R107 */ + 0x0000, /* R108 */ + 0x0000, /* R109 */ + 0x0000, /* R110 */ + 0x0000, /* R111 */ + 0x0000, /* R112 */ + 0x0000, /* R113 */ + 0x0000, /* R114 */ + 0x0000, /* R115 */ + 0x0000, /* R116 */ + 0x0000, /* R117 */ + 0x0000, /* R118 */ + 0x0000, /* R119 */ + 0x0000, /* R120 */ + 0x0000, /* R121 */ + 0x0000, /* R122 */ + 0x0000, /* R123 */ + 0x0000, /* R124 */ + 0x0000, /* R125 */ + 0x0000, /* R126 */ + 0x0000, /* R127 */ + 0x0000, /* R128 */ + 0x0000, /* R129 */ + 0x0000, /* R130 */ + 0x0000, /* R131 */ + 0x0000, /* R132 */ + 0x0000, /* R133 */ + 0x0000, /* R134 */ + 0x0000, /* R135 */ + 0x0000, /* R136 */ + 0x0000, /* R137 */ + 0x0000, /* R138 */ + 0x0000, /* R139 */ + 0x0000, /* R140 */ + 0x0000, /* R141 */ + 0x0000, /* R142 */ + 0x0000, /* R143 */ + 0x0000, /* R144 */ + 0x0000, /* R145 */ + 0x0000, /* R146 */ + 0x0000, /* R147 */ + 0x0000, /* R148 */ + 0x0000, /* R149 */ + 0x0000, /* R150 */ + 0x0000, /* R151 */ + 0x0000, /* R152 */ + 0x0000, /* R153 */ + 0x0000, /* R154 */ + 0x0000, /* R155 */ + 0x0000, /* R156 */ + 0x0000, /* R157 */ + 0x0000, /* R158 */ + 0x0000, /* R159 */ + 0x0000, /* R160 */ + 0x0000, /* R161 */ + 0x0000, /* R162 */ + 0x0000, /* R163 */ + 0x0000, /* R164 */ + 0x0000, /* R165 */ + 0x0000, /* R166 */ + 0x0000, /* R167 */ + 0x0000, /* R168 */ + 0x0000, /* R169 */ + 0x0000, /* R170 */ + 0x0000, /* R171 */ + 0x0000, /* R172 */ + 0x0000, /* R173 */ + 0x0000, /* R174 */ + 0x0000, /* R175 */ + 0x0000, /* R176 */ + 0x0000, /* R177 */ + 0x0000, /* R178 */ + 0x0000, /* R179 */ + 0x0000, /* R180 */ + 0x0000, /* R181 */ + 0x0000, /* R182 */ + 0x0000, /* R183 */ + 0x0000, /* R184 */ + 0x0000, /* R185 */ + 0x0000, /* R186 */ + 0x0000, /* R187 */ + 0x0000, /* R188 */ + 0x0000, /* R189 */ + 0x0000, /* R190 */ + 0x0000, /* R191 */ + 0x0000, /* R192 */ + 0x0000, /* R193 */ + 0x0000, /* R194 */ + 0x0000, /* R195 */ + 0x0030, /* R196 */ + 0x0006, /* R197 */ + 0x0000, /* R198 */ + 0x0060, /* R199 */ + 0x0000, /* R200 */ + 0x003F, /* R201 */ + 0x0000, /* R202 */ + 0x0000, /* R203 */ + 0x0000, /* R204 */ + 0x0001, /* R205 */ + 0x0000, /* R206 */ + 0x0181, /* R207 */ + 0x0005, /* R208 */ + 0x0008, /* R209 */ + 0x0008, /* R210 */ + 0x0000, /* R211 */ + 0x013B, /* R212 */ + 0x0000, /* R213 */ + 0x0000, /* R214 */ + 0x0000, /* R215 */ + 0x0000, /* R216 */ + 0x0070, /* R217 */ + 0x0000, /* R218 */ + 0x0000, /* R219 */ + 0x0000, /* R220 */ + 0x0000, /* R221 */ + 0x0000, /* R222 */ + 0x0003, /* R223 */ + 0x0000, /* R224 */ + 0x0000, /* R225 */ + 0x0001, /* R226 */ + 0x0008, /* R227 */ + 0x0000, /* R228 */ + 0x0000, /* R229 */ + 0x0000, /* R230 */ + 0x0000, /* R231 */ + 0x0004, /* R232 */ + 0x0000, /* R233 */ + 0x0000, /* R234 */ + 0x0000, /* R235 */ + 0x0000, /* R236 */ + 0x0000, /* R237 */ + 0x0080, /* R238 */ + 0x0000, /* R239 */ + 0x0000, /* R240 */ + 0x0000, /* R241 */ + 0x0000, /* R242 */ + 0x0000, /* R243 */ + 0x0000, /* R244 */ + 0x0052, /* R245 */ + 0x0110, /* R246 */ + 0x0040, /* R247 */ + 0x0000, /* R248 */ + 0x0030, /* R249 */ + 0x0000, /* R250 */ + 0x0000, /* R251 */ + 0x0001, /* R252 - General test 1 */ +}; + +struct wm8961_priv { + struct snd_soc_codec codec; + int sysclk; + u16 reg_cache[WM8961_MAX_REGISTER]; +}; + +static int wm8961_volatile_register(unsigned int reg) +{ + switch (reg) { + case WM8961_SOFTWARE_RESET: + case WM8961_WRITE_SEQUENCER_7: + case WM8961_DC_SERVO_1: + return 1; + + default: + return 0; + } +} + +static int wm8961_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, WM8961_SOFTWARE_RESET, 0); +} + +/* + * The headphone output supports special anti-pop sequences giving + * silent power up and power down. + */ +static int wm8961_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 hp_reg = snd_soc_read(codec, WM8961_ANALOGUE_HP_0); + u16 cp_reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_1); + u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2); + u16 dcs_reg = snd_soc_read(codec, WM8961_DC_SERVO_1); + int timeout = 500; + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Make sure the output is shorted */ + hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT); + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Enable the charge pump */ + cp_reg |= WM8961_CP_ENA; + snd_soc_write(codec, WM8961_CHARGE_PUMP_1, cp_reg); + mdelay(5); + + /* Enable the PGA */ + pwr_reg |= WM8961_LOUT1_PGA | WM8961_ROUT1_PGA; + snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + + /* Enable the amplifier */ + hp_reg |= WM8961_HPR_ENA | WM8961_HPL_ENA; + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Second stage enable */ + hp_reg |= WM8961_HPR_ENA_DLY | WM8961_HPL_ENA_DLY; + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Enable the DC servo & trigger startup */ + dcs_reg |= + WM8961_DCS_ENA_CHAN_HPR | WM8961_DCS_TRIG_STARTUP_HPR | + WM8961_DCS_ENA_CHAN_HPL | WM8961_DCS_TRIG_STARTUP_HPL; + dev_dbg(codec->dev, "Enabling DC servo\n"); + + snd_soc_write(codec, WM8961_DC_SERVO_1, dcs_reg); + do { + msleep(1); + dcs_reg = snd_soc_read(codec, WM8961_DC_SERVO_1); + } while (--timeout && + dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR | + WM8961_DCS_TRIG_STARTUP_HPL)); + if (dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR | + WM8961_DCS_TRIG_STARTUP_HPL)) + dev_err(codec->dev, "DC servo timed out\n"); + else + dev_dbg(codec->dev, "DC servo startup complete\n"); + + /* Enable the output stage */ + hp_reg |= WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP; + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Remove the short on the output stage */ + hp_reg |= WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT; + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + } + + if (event & SND_SOC_DAPM_PRE_PMD) { + /* Short the output */ + hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT); + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Disable the output stage */ + hp_reg &= ~(WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP); + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Disable DC offset cancellation */ + dcs_reg &= ~(WM8961_DCS_ENA_CHAN_HPR | + WM8961_DCS_ENA_CHAN_HPL); + snd_soc_write(codec, WM8961_DC_SERVO_1, dcs_reg); + + /* Finish up */ + hp_reg &= ~(WM8961_HPR_ENA_DLY | WM8961_HPR_ENA | + WM8961_HPL_ENA_DLY | WM8961_HPL_ENA); + snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg); + + /* Disable the PGA */ + pwr_reg &= ~(WM8961_LOUT1_PGA | WM8961_ROUT1_PGA); + snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + + /* Disable the charge pump */ + dev_dbg(codec->dev, "Disabling charge pump\n"); + snd_soc_write(codec, WM8961_CHARGE_PUMP_1, + cp_reg & ~WM8961_CP_ENA); + } + + return 0; +} + +static int wm8961_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2); + u16 spk_reg = snd_soc_read(codec, WM8961_CLASS_D_CONTROL_1); + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Enable the PGA */ + pwr_reg |= WM8961_SPKL_PGA | WM8961_SPKR_PGA; + snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + + /* Enable the amplifier */ + spk_reg |= WM8961_SPKL_ENA | WM8961_SPKR_ENA; + snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); + } + + if (event & SND_SOC_DAPM_PRE_PMD) { + /* Enable the amplifier */ + spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA); + snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg); + + /* Enable the PGA */ + pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA); + snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg); + } + + return 0; +} + +static const char *adc_hpf_text[] = { + "Hi-fi", "Voice 1", "Voice 2", "Voice 3", +}; + +static const struct soc_enum adc_hpf = + SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text); + +static const char *dac_deemph_text[] = { + "None", "32kHz", "44.1kHz", "48kHz", +}; + +static const struct soc_enum dac_deemph = + SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text); + +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0); +static unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(13, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(20, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(29, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(pga_tlv, -2325, 75, 0); + +static const struct snd_kcontrol_new wm8961_snd_controls[] = { +SOC_DOUBLE_R_TLV("Headphone Volume", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME, + 0, 127, 0, out_tlv), +SOC_DOUBLE_TLV("Headphone Secondary Volume", WM8961_ANALOGUE_HP_2, + 6, 3, 7, 0, hp_sec_tlv), +SOC_DOUBLE_R("Headphone ZC Switch", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Volume", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker ZC Switch", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME, + 7, 1, 0), +SOC_SINGLE("Speaker AC Gain", WM8961_CLASS_D_CONTROL_2, 0, 7, 0), + +SOC_SINGLE("DAC x128 OSR Switch", WM8961_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_ENUM("DAC Deemphasis", dac_deemph), +SOC_SINGLE("DAC Soft Mute Switch", WM8961_ADC_DAC_CONTROL_2, 3, 1, 0), + +SOC_DOUBLE_R_TLV("Sidetone Volume", WM8961_DSP_SIDETONE_0, + WM8961_DSP_SIDETONE_1, 4, 12, 0, sidetone_tlv), + +SOC_SINGLE("ADC High Pass Filter Switch", WM8961_ADC_DAC_CONTROL_1, 0, 1, 0), +SOC_ENUM("ADC High Pass Filter Mode", adc_hpf), + +SOC_DOUBLE_R_TLV("Capture Volume", + WM8961_LEFT_ADC_VOLUME, WM8961_RIGHT_ADC_VOLUME, + 1, 119, 0, adc_tlv), +SOC_DOUBLE_R_TLV("Capture Boost Volume", + WM8961_ADCL_SIGNAL_PATH, WM8961_ADCR_SIGNAL_PATH, + 4, 3, 0, boost_tlv), +SOC_DOUBLE_R_TLV("Capture PGA Volume", + WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME, + 0, 62, 0, pga_tlv), +SOC_DOUBLE_R("Capture PGA ZC Switch", + WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME, + 6, 1, 1), +SOC_DOUBLE_R("Capture PGA Switch", + WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME, + 7, 1, 1), +}; + +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum dacl_sidetone = + SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text); + +static const struct soc_enum dacr_sidetone = + SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text); + +static const struct snd_kcontrol_new dacl_mux = + SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone); + +static const struct snd_kcontrol_new dacr_mux = + SOC_DAPM_ENUM("DACR Sidetone", dacr_sidetone); + +static const struct snd_soc_dapm_widget wm8961_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("LINPUT"), +SND_SOC_DAPM_INPUT("RINPUT"), + +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8961_CLOCKING2, 4, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Left Input", WM8961_PWR_MGMT_1, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Input", WM8961_PWR_MGMT_1, 4, 0, NULL, 0), + +SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", WM8961_PWR_MGMT_1, 3, 0), +SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", WM8961_PWR_MGMT_1, 2, 0), + +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8961_PWR_MGMT_1, 1, 0), + +SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &dacl_mux), +SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &dacr_mux), + +SND_SOC_DAPM_DAC("DACL", "HiFi Playback", WM8961_PWR_MGMT_2, 8, 0), +SND_SOC_DAPM_DAC("DACR", "HiFi Playback", WM8961_PWR_MGMT_2, 7, 0), + +/* Handle as a mono path for DCS */ +SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, + 4, 0, NULL, 0, wm8961_hp_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_E("Speaker Output", SND_SOC_NOPM, + 4, 0, NULL, 0, wm8961_spk_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_OUTPUT("HP_L"), +SND_SOC_DAPM_OUTPUT("HP_R"), +SND_SOC_DAPM_OUTPUT("SPK_LN"), +SND_SOC_DAPM_OUTPUT("SPK_LP"), +SND_SOC_DAPM_OUTPUT("SPK_RN"), +SND_SOC_DAPM_OUTPUT("SPK_RP"), +}; + + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DACL", NULL, "CLK_DSP" }, + { "DACL", NULL, "DACL Sidetone" }, + { "DACR", NULL, "CLK_DSP" }, + { "DACR", NULL, "DACR Sidetone" }, + + { "DACL Sidetone", "Left", "ADCL" }, + { "DACL Sidetone", "Right", "ADCR" }, + + { "DACR Sidetone", "Left", "ADCL" }, + { "DACR Sidetone", "Right", "ADCR" }, + + { "HP_L", NULL, "Headphone Output" }, + { "HP_R", NULL, "Headphone Output" }, + { "Headphone Output", NULL, "DACL" }, + { "Headphone Output", NULL, "DACR" }, + + { "SPK_LN", NULL, "Speaker Output" }, + { "SPK_LP", NULL, "Speaker Output" }, + { "SPK_RN", NULL, "Speaker Output" }, + { "SPK_RP", NULL, "Speaker Output" }, + + { "Speaker Output", NULL, "DACL" }, + { "Speaker Output", NULL, "DACR" }, + + { "ADCL", NULL, "Left Input" }, + { "ADCL", NULL, "CLK_DSP" }, + { "ADCR", NULL, "Right Input" }, + { "ADCR", NULL, "CLK_DSP" }, + + { "Left Input", NULL, "LINPUT" }, + { "Right Input", NULL, "RINPUT" }, + +}; + +/* Values for CLK_SYS_RATE */ +static struct { + int ratio; + u16 val; +} wm8961_clk_sys_ratio[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 768, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +/* Values for SAMPLE_RATE */ +static struct { + int rate; + u16 val; +} wm8961_srate[] = { + { 48000, 0 }, + { 44100, 0 }, + { 32000, 1 }, + { 22050, 2 }, + { 24000, 2 }, + { 16000, 3 }, + { 11250, 4 }, + { 12000, 4 }, + { 8000, 5 }, +}; + +static int wm8961_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8961_priv *wm8961 = codec->private_data; + int i, best, target, fs; + u16 reg; + + fs = params_rate(params); + + if (!wm8961->sysclk) { + dev_err(codec->dev, "MCLK has not been specified\n"); + return -EINVAL; + } + + /* Find the closest sample rate for the filters */ + best = 0; + for (i = 0; i < ARRAY_SIZE(wm8961_srate); i++) { + if (abs(wm8961_srate[i].rate - fs) < + abs(wm8961_srate[best].rate - fs)) + best = i; + } + reg = snd_soc_read(codec, WM8961_ADDITIONAL_CONTROL_3); + reg &= ~WM8961_SAMPLE_RATE_MASK; + reg |= wm8961_srate[best].val; + snd_soc_write(codec, WM8961_ADDITIONAL_CONTROL_3, reg); + dev_dbg(codec->dev, "Selected SRATE %dHz for %dHz\n", + wm8961_srate[best].rate, fs); + + /* Select a CLK_SYS/fs ratio equal to or higher than required */ + target = wm8961->sysclk / fs; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && target < 64) { + dev_err(codec->dev, + "SYSCLK must be at least 64*fs for DAC\n"); + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && target < 256) { + dev_err(codec->dev, + "SYSCLK must be at least 256*fs for ADC\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(wm8961_clk_sys_ratio); i++) { + if (wm8961_clk_sys_ratio[i].ratio >= target) + break; + } + if (i == ARRAY_SIZE(wm8961_clk_sys_ratio)) { + dev_err(codec->dev, "Unable to generate CLK_SYS_RATE\n"); + return -EINVAL; + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATE of %d for %d/%d=%d\n", + wm8961_clk_sys_ratio[i].ratio, wm8961->sysclk, fs, + wm8961->sysclk / fs); + + reg = snd_soc_read(codec, WM8961_CLOCKING_4); + reg &= ~WM8961_CLK_SYS_RATE_MASK; + reg |= wm8961_clk_sys_ratio[i].val << WM8961_CLK_SYS_RATE_SHIFT; + snd_soc_write(codec, WM8961_CLOCKING_4, reg); + + reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0); + reg &= ~WM8961_WL_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 1 << WM8961_WL_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 2 << WM8961_WL_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 3 << WM8961_WL_SHIFT; + break; + default: + return -EINVAL; + } + snd_soc_write(codec, WM8961_AUDIO_INTERFACE_0, reg); + + /* Sloping stop-band filter is recommended for <= 24kHz */ + reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_2); + if (fs <= 24000) + reg |= WM8961_DACSLOPE; + else + reg &= WM8961_DACSLOPE; + snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg); + + return 0; +} + +static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, + int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8961_priv *wm8961 = codec->private_data; + u16 reg = snd_soc_read(codec, WM8961_CLOCKING1); + + if (freq > 33000000) { + dev_err(codec->dev, "MCLK must be <33MHz\n"); + return -EINVAL; + } + + if (freq > 16500000) { + dev_dbg(codec->dev, "Using MCLK/2 for %dHz MCLK\n", freq); + reg |= WM8961_MCLKDIV; + freq /= 2; + } else { + dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq); + reg &= WM8961_MCLKDIV; + } + + snd_soc_write(codec, WM8961_CLOCKING1, reg); + + wm8961->sysclk = freq; + + return 0; +} + +static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u16 aif = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0); + + aif &= ~(WM8961_BCLKINV | WM8961_LRP | + WM8961_MS | WM8961_FORMAT_MASK); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aif |= WM8961_MS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + break; + + case SND_SOC_DAIFMT_LEFT_J: + aif |= 1; + break; + + case SND_SOC_DAIFMT_I2S: + aif |= 2; + break; + + case SND_SOC_DAIFMT_DSP_B: + aif |= WM8961_LRP; + case SND_SOC_DAIFMT_DSP_A: + aif |= 3; + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + aif |= WM8961_LRP; + break; + case SND_SOC_DAIFMT_IB_NF: + aif |= WM8961_BCLKINV; + break; + case SND_SOC_DAIFMT_IB_IF: + aif |= WM8961_BCLKINV | WM8961_LRP; + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, WM8961_AUDIO_INTERFACE_0, aif); +} + +static int wm8961_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg = snd_soc_read(codec, WM8961_ADDITIONAL_CONTROL_2); + + if (tristate) + reg |= WM8961_TRIS; + else + reg &= ~WM8961_TRIS; + + return snd_soc_write(codec, WM8961_ADDITIONAL_CONTROL_2, reg); +} + +static int wm8961_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_1); + + if (mute) + reg |= WM8961_DACMU; + else + reg &= ~WM8961_DACMU; + + msleep(17); + + return snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_1, reg); +} + +static int wm8961_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct snd_soc_codec *codec = dai->codec; + u16 reg; + + switch (div_id) { + case WM8961_BCLK: + reg = snd_soc_read(codec, WM8961_CLOCKING2); + reg &= ~WM8961_BCLKDIV_MASK; + reg |= div; + snd_soc_write(codec, WM8961_CLOCKING2, reg); + break; + + case WM8961_LRCLK: + reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_2); + reg &= ~WM8961_LRCLK_RATE_MASK; + reg |= div; + snd_soc_write(codec, WM8961_AUDIO_INTERFACE_2, reg); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int wm8961_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + + /* This is all slightly unusual since we have no bypass paths + * and the output amplifier structure means we can just slam + * the biases straight up rather than having to ramp them + * slowly. + */ + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + /* Enable bias generation */ + reg = snd_soc_read(codec, WM8961_ANTI_POP); + reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN; + snd_soc_write(codec, WM8961_ANTI_POP, reg); + + /* VMID=2*50k, VREF */ + reg = snd_soc_read(codec, WM8961_PWR_MGMT_1); + reg &= ~WM8961_VMIDSEL_MASK; + reg |= (1 << WM8961_VMIDSEL_SHIFT) | WM8961_VREF; + snd_soc_write(codec, WM8961_PWR_MGMT_1, reg); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_PREPARE) { + /* VREF off */ + reg = snd_soc_read(codec, WM8961_PWR_MGMT_1); + reg &= ~WM8961_VREF; + snd_soc_write(codec, WM8961_PWR_MGMT_1, reg); + + /* Bias generation off */ + reg = snd_soc_read(codec, WM8961_ANTI_POP); + reg &= ~(WM8961_BUFIOEN | WM8961_BUFDCOPEN); + snd_soc_write(codec, WM8961_ANTI_POP, reg); + + /* VMID off */ + reg = snd_soc_read(codec, WM8961_PWR_MGMT_1); + reg &= ~WM8961_VMIDSEL_MASK; + snd_soc_write(codec, WM8961_PWR_MGMT_1, reg); + } + break; + + case SND_SOC_BIAS_OFF: + break; + } + + codec->bias_level = level; + + return 0; +} + + +#define WM8961_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8961_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8961_dai_ops = { + .hw_params = wm8961_hw_params, + .set_sysclk = wm8961_set_sysclk, + .set_fmt = wm8961_set_fmt, + .digital_mute = wm8961_digital_mute, + .set_tristate = wm8961_set_tristate, + .set_clkdiv = wm8961_set_clkdiv, +}; + +struct snd_soc_dai wm8961_dai = { + .name = "WM8961", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8961_RATES, + .formats = WM8961_FORMATS,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8961_RATES, + .formats = WM8961_FORMATS,}, + .ops = &wm8961_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8961_dai); + + +static struct snd_soc_codec *wm8961_codec; + +static int wm8961_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8961_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8961_codec; + codec = wm8961_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8961_snd_controls, + ARRAY_SIZE(wm8961_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets, + ARRAY_SIZE(wm8961_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8961_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8961_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8961_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + u16 *reg_cache = codec->reg_cache; + int i; + + for (i = 0; i < codec->reg_cache_size; i++) { + if (i == WM8961_SOFTWARE_RESET) + continue; + + snd_soc_write(codec, i, reg_cache[i]); + } + + wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8961_suspend NULL +#define wm8961_resume NULL +#endif + +struct snd_soc_codec_device soc_codec_dev_wm8961 = { + .probe = wm8961_probe, + .remove = wm8961_remove, + .suspend = wm8961_suspend, + .resume = wm8961_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8961); + +static int wm8961_register(struct wm8961_priv *wm8961) +{ + struct snd_soc_codec *codec = &wm8961->codec; + int ret; + u16 reg; + + if (wm8961_codec) { + dev_err(codec->dev, "Another WM8961 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8961; + codec->name = "WM8961"; + codec->owner = THIS_MODULE; + codec->dai = &wm8961_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8961->reg_cache); + codec->reg_cache = &wm8961->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8961_set_bias_level; + codec->volatile_register = wm8961_volatile_register; + + memcpy(codec->reg_cache, wm8961_reg_defaults, + sizeof(wm8961_reg_defaults)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + reg = snd_soc_read(codec, WM8961_SOFTWARE_RESET); + if (reg != 0x1801) { + dev_err(codec->dev, "Device is not a WM8961: ID=0x%x\n", reg); + ret = -EINVAL; + goto err; + } + + /* This isn't volatile - readback doesn't correspond to write */ + reg = codec->hw_read(codec, WM8961_RIGHT_INPUT_VOLUME); + dev_info(codec->dev, "WM8961 family %d revision %c\n", + (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT, + ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT) + + 'A'); + + ret = wm8961_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + /* Enable class W */ + reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B); + reg |= WM8961_CP_DYN_PWR_MASK; + snd_soc_write(codec, WM8961_CHARGE_PUMP_B, reg); + + /* Latch volume update bits (right channel only, we always + * write both out) and default ZC on. */ + reg = snd_soc_read(codec, WM8961_ROUT1_VOLUME); + snd_soc_write(codec, WM8961_ROUT1_VOLUME, + reg | WM8961_LO1ZC | WM8961_OUT1VU); + snd_soc_write(codec, WM8961_LOUT1_VOLUME, reg | WM8961_LO1ZC); + reg = snd_soc_read(codec, WM8961_ROUT2_VOLUME); + snd_soc_write(codec, WM8961_ROUT2_VOLUME, + reg | WM8961_SPKRZC | WM8961_SPKVU); + snd_soc_write(codec, WM8961_LOUT2_VOLUME, reg | WM8961_SPKLZC); + + reg = snd_soc_read(codec, WM8961_RIGHT_ADC_VOLUME); + snd_soc_write(codec, WM8961_RIGHT_ADC_VOLUME, reg | WM8961_ADCVU); + reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME); + snd_soc_write(codec, WM8961_RIGHT_INPUT_VOLUME, reg | WM8961_IPVU); + + /* Use soft mute by default */ + reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_2); + reg |= WM8961_DACSMM; + snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg); + + /* Use automatic clocking mode by default; for now this is all + * we support. + */ + reg = snd_soc_read(codec, WM8961_CLOCKING_3); + reg &= ~WM8961_MANUAL_MODE; + snd_soc_write(codec, WM8961_CLOCKING_3, reg); + + wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8961_dai.dev = codec->dev; + + wm8961_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8961_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm8961); + return ret; +} + +static void wm8961_unregister(struct wm8961_priv *wm8961) +{ + wm8961_set_bias_level(&wm8961->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8961_dai); + snd_soc_unregister_codec(&wm8961->codec); + kfree(wm8961); + wm8961_codec = NULL; +} + +static __devinit int wm8961_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8961_priv *wm8961; + struct snd_soc_codec *codec; + + wm8961 = kzalloc(sizeof(struct wm8961_priv), GFP_KERNEL); + if (wm8961 == NULL) + return -ENOMEM; + + codec = &wm8961->codec; + + i2c_set_clientdata(i2c, wm8961); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8961_register(wm8961); +} + +static __devexit int wm8961_i2c_remove(struct i2c_client *client) +{ + struct wm8961_priv *wm8961 = i2c_get_clientdata(client); + wm8961_unregister(wm8961); + return 0; +} + +#ifdef CONFIG_PM +static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8961_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8961_i2c_suspend NULL +#define wm8961_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8961_i2c_id[] = { + { "wm8961", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id); + +static struct i2c_driver wm8961_i2c_driver = { + .driver = { + .name = "wm8961", + .owner = THIS_MODULE, + }, + .probe = wm8961_i2c_probe, + .remove = __devexit_p(wm8961_i2c_remove), + .suspend = wm8961_i2c_suspend, + .resume = wm8961_i2c_resume, + .id_table = wm8961_i2c_id, +}; + +static int __init wm8961_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8961_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8961 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm8961_modinit); + +static void __exit wm8961_exit(void) +{ + i2c_del_driver(&wm8961_i2c_driver); +} +module_exit(wm8961_exit); + + +MODULE_DESCRIPTION("ASoC WM8961 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8961.h b/sound/soc/codecs/wm8961.h new file mode 100644 index 000000000000..5513bfd720d6 --- /dev/null +++ b/sound/soc/codecs/wm8961.h @@ -0,0 +1,866 @@ +/* + * wm8961.h -- WM8961 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8961_H +#define _WM8961_H + +#include <sound/soc.h> + +extern struct snd_soc_codec_device soc_codec_dev_wm8961; +extern struct snd_soc_dai wm8961_dai; + +#define WM8961_BCLK 1 +#define WM8961_LRCLK 2 + +#define WM8961_BCLK_DIV_1 0 +#define WM8961_BCLK_DIV_1_5 1 +#define WM8961_BCLK_DIV_2 2 +#define WM8961_BCLK_DIV_3 3 +#define WM8961_BCLK_DIV_4 4 +#define WM8961_BCLK_DIV_5_5 5 +#define WM8961_BCLK_DIV_6 6 +#define WM8961_BCLK_DIV_8 7 +#define WM8961_BCLK_DIV_11 8 +#define WM8961_BCLK_DIV_12 9 +#define WM8961_BCLK_DIV_16 10 +#define WM8961_BCLK_DIV_24 11 +#define WM8961_BCLK_DIV_32 13 + + +/* + * Register values. + */ +#define WM8961_LEFT_INPUT_VOLUME 0x00 +#define WM8961_RIGHT_INPUT_VOLUME 0x01 +#define WM8961_LOUT1_VOLUME 0x02 +#define WM8961_ROUT1_VOLUME 0x03 +#define WM8961_CLOCKING1 0x04 +#define WM8961_ADC_DAC_CONTROL_1 0x05 +#define WM8961_ADC_DAC_CONTROL_2 0x06 +#define WM8961_AUDIO_INTERFACE_0 0x07 +#define WM8961_CLOCKING2 0x08 +#define WM8961_AUDIO_INTERFACE_1 0x09 +#define WM8961_LEFT_DAC_VOLUME 0x0A +#define WM8961_RIGHT_DAC_VOLUME 0x0B +#define WM8961_AUDIO_INTERFACE_2 0x0E +#define WM8961_SOFTWARE_RESET 0x0F +#define WM8961_ALC1 0x11 +#define WM8961_ALC2 0x12 +#define WM8961_ALC3 0x13 +#define WM8961_NOISE_GATE 0x14 +#define WM8961_LEFT_ADC_VOLUME 0x15 +#define WM8961_RIGHT_ADC_VOLUME 0x16 +#define WM8961_ADDITIONAL_CONTROL_1 0x17 +#define WM8961_ADDITIONAL_CONTROL_2 0x18 +#define WM8961_PWR_MGMT_1 0x19 +#define WM8961_PWR_MGMT_2 0x1A +#define WM8961_ADDITIONAL_CONTROL_3 0x1B +#define WM8961_ANTI_POP 0x1C +#define WM8961_CLOCKING_3 0x1E +#define WM8961_ADCL_SIGNAL_PATH 0x20 +#define WM8961_ADCR_SIGNAL_PATH 0x21 +#define WM8961_LOUT2_VOLUME 0x28 +#define WM8961_ROUT2_VOLUME 0x29 +#define WM8961_PWR_MGMT_3 0x2F +#define WM8961_ADDITIONAL_CONTROL_4 0x30 +#define WM8961_CLASS_D_CONTROL_1 0x31 +#define WM8961_CLASS_D_CONTROL_2 0x33 +#define WM8961_CLOCKING_4 0x38 +#define WM8961_DSP_SIDETONE_0 0x39 +#define WM8961_DSP_SIDETONE_1 0x3A +#define WM8961_DC_SERVO_0 0x3C +#define WM8961_DC_SERVO_1 0x3D +#define WM8961_DC_SERVO_3 0x3F +#define WM8961_DC_SERVO_5 0x41 +#define WM8961_ANALOGUE_PGA_BIAS 0x44 +#define WM8961_ANALOGUE_HP_0 0x45 +#define WM8961_ANALOGUE_HP_2 0x47 +#define WM8961_CHARGE_PUMP_1 0x48 +#define WM8961_CHARGE_PUMP_B 0x52 +#define WM8961_WRITE_SEQUENCER_1 0x57 +#define WM8961_WRITE_SEQUENCER_2 0x58 +#define WM8961_WRITE_SEQUENCER_3 0x59 +#define WM8961_WRITE_SEQUENCER_4 0x5A +#define WM8961_WRITE_SEQUENCER_5 0x5B +#define WM8961_WRITE_SEQUENCER_6 0x5C +#define WM8961_WRITE_SEQUENCER_7 0x5D +#define WM8961_GENERAL_TEST_1 0xFC + + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Left Input volume + */ +#define WM8961_IPVU 0x0100 /* IPVU */ +#define WM8961_IPVU_MASK 0x0100 /* IPVU */ +#define WM8961_IPVU_SHIFT 8 /* IPVU */ +#define WM8961_IPVU_WIDTH 1 /* IPVU */ +#define WM8961_LINMUTE 0x0080 /* LINMUTE */ +#define WM8961_LINMUTE_MASK 0x0080 /* LINMUTE */ +#define WM8961_LINMUTE_SHIFT 7 /* LINMUTE */ +#define WM8961_LINMUTE_WIDTH 1 /* LINMUTE */ +#define WM8961_LIZC 0x0040 /* LIZC */ +#define WM8961_LIZC_MASK 0x0040 /* LIZC */ +#define WM8961_LIZC_SHIFT 6 /* LIZC */ +#define WM8961_LIZC_WIDTH 1 /* LIZC */ +#define WM8961_LINVOL_MASK 0x003F /* LINVOL - [5:0] */ +#define WM8961_LINVOL_SHIFT 0 /* LINVOL - [5:0] */ +#define WM8961_LINVOL_WIDTH 6 /* LINVOL - [5:0] */ + +/* + * R1 (0x01) - Right Input volume + */ +#define WM8961_DEVICE_ID_MASK 0xF000 /* DEVICE_ID - [15:12] */ +#define WM8961_DEVICE_ID_SHIFT 12 /* DEVICE_ID - [15:12] */ +#define WM8961_DEVICE_ID_WIDTH 4 /* DEVICE_ID - [15:12] */ +#define WM8961_CHIP_REV_MASK 0x0E00 /* CHIP_REV - [11:9] */ +#define WM8961_CHIP_REV_SHIFT 9 /* CHIP_REV - [11:9] */ +#define WM8961_CHIP_REV_WIDTH 3 /* CHIP_REV - [11:9] */ +#define WM8961_IPVU 0x0100 /* IPVU */ +#define WM8961_IPVU_MASK 0x0100 /* IPVU */ +#define WM8961_IPVU_SHIFT 8 /* IPVU */ +#define WM8961_IPVU_WIDTH 1 /* IPVU */ +#define WM8961_RINMUTE 0x0080 /* RINMUTE */ +#define WM8961_RINMUTE_MASK 0x0080 /* RINMUTE */ +#define WM8961_RINMUTE_SHIFT 7 /* RINMUTE */ +#define WM8961_RINMUTE_WIDTH 1 /* RINMUTE */ +#define WM8961_RIZC 0x0040 /* RIZC */ +#define WM8961_RIZC_MASK 0x0040 /* RIZC */ +#define WM8961_RIZC_SHIFT 6 /* RIZC */ +#define WM8961_RIZC_WIDTH 1 /* RIZC */ +#define WM8961_RINVOL_MASK 0x003F /* RINVOL - [5:0] */ +#define WM8961_RINVOL_SHIFT 0 /* RINVOL - [5:0] */ +#define WM8961_RINVOL_WIDTH 6 /* RINVOL - [5:0] */ + +/* + * R2 (0x02) - LOUT1 volume + */ +#define WM8961_OUT1VU 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_MASK 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_SHIFT 8 /* OUT1VU */ +#define WM8961_OUT1VU_WIDTH 1 /* OUT1VU */ +#define WM8961_LO1ZC 0x0080 /* LO1ZC */ +#define WM8961_LO1ZC_MASK 0x0080 /* LO1ZC */ +#define WM8961_LO1ZC_SHIFT 7 /* LO1ZC */ +#define WM8961_LO1ZC_WIDTH 1 /* LO1ZC */ +#define WM8961_LOUT1VOL_MASK 0x007F /* LOUT1VOL - [6:0] */ +#define WM8961_LOUT1VOL_SHIFT 0 /* LOUT1VOL - [6:0] */ +#define WM8961_LOUT1VOL_WIDTH 7 /* LOUT1VOL - [6:0] */ + +/* + * R3 (0x03) - ROUT1 volume + */ +#define WM8961_OUT1VU 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_MASK 0x0100 /* OUT1VU */ +#define WM8961_OUT1VU_SHIFT 8 /* OUT1VU */ +#define WM8961_OUT1VU_WIDTH 1 /* OUT1VU */ +#define WM8961_RO1ZC 0x0080 /* RO1ZC */ +#define WM8961_RO1ZC_MASK 0x0080 /* RO1ZC */ +#define WM8961_RO1ZC_SHIFT 7 /* RO1ZC */ +#define WM8961_RO1ZC_WIDTH 1 /* RO1ZC */ +#define WM8961_ROUT1VOL_MASK 0x007F /* ROUT1VOL - [6:0] */ +#define WM8961_ROUT1VOL_SHIFT 0 /* ROUT1VOL - [6:0] */ +#define WM8961_ROUT1VOL_WIDTH 7 /* ROUT1VOL - [6:0] */ + +/* + * R4 (0x04) - Clocking1 + */ +#define WM8961_ADCDIV_MASK 0x01C0 /* ADCDIV - [8:6] */ +#define WM8961_ADCDIV_SHIFT 6 /* ADCDIV - [8:6] */ +#define WM8961_ADCDIV_WIDTH 3 /* ADCDIV - [8:6] */ +#define WM8961_DACDIV_MASK 0x0038 /* DACDIV - [5:3] */ +#define WM8961_DACDIV_SHIFT 3 /* DACDIV - [5:3] */ +#define WM8961_DACDIV_WIDTH 3 /* DACDIV - [5:3] */ +#define WM8961_MCLKDIV 0x0004 /* MCLKDIV */ +#define WM8961_MCLKDIV_MASK 0x0004 /* MCLKDIV */ +#define WM8961_MCLKDIV_SHIFT 2 /* MCLKDIV */ +#define WM8961_MCLKDIV_WIDTH 1 /* MCLKDIV */ + +/* + * R5 (0x05) - ADC & DAC Control 1 + */ +#define WM8961_ADCPOL_MASK 0x0060 /* ADCPOL - [6:5] */ +#define WM8961_ADCPOL_SHIFT 5 /* ADCPOL - [6:5] */ +#define WM8961_ADCPOL_WIDTH 2 /* ADCPOL - [6:5] */ +#define WM8961_DACMU 0x0008 /* DACMU */ +#define WM8961_DACMU_MASK 0x0008 /* DACMU */ +#define WM8961_DACMU_SHIFT 3 /* DACMU */ +#define WM8961_DACMU_WIDTH 1 /* DACMU */ +#define WM8961_DEEMPH_MASK 0x0006 /* DEEMPH - [2:1] */ +#define WM8961_DEEMPH_SHIFT 1 /* DEEMPH - [2:1] */ +#define WM8961_DEEMPH_WIDTH 2 /* DEEMPH - [2:1] */ +#define WM8961_ADCHPD 0x0001 /* ADCHPD */ +#define WM8961_ADCHPD_MASK 0x0001 /* ADCHPD */ +#define WM8961_ADCHPD_SHIFT 0 /* ADCHPD */ +#define WM8961_ADCHPD_WIDTH 1 /* ADCHPD */ + +/* + * R6 (0x06) - ADC & DAC Control 2 + */ +#define WM8961_ADC_HPF_CUT_MASK 0x0180 /* ADC_HPF_CUT - [8:7] */ +#define WM8961_ADC_HPF_CUT_SHIFT 7 /* ADC_HPF_CUT - [8:7] */ +#define WM8961_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [8:7] */ +#define WM8961_DACPOL_MASK 0x0060 /* DACPOL - [6:5] */ +#define WM8961_DACPOL_SHIFT 5 /* DACPOL - [6:5] */ +#define WM8961_DACPOL_WIDTH 2 /* DACPOL - [6:5] */ +#define WM8961_DACSMM 0x0008 /* DACSMM */ +#define WM8961_DACSMM_MASK 0x0008 /* DACSMM */ +#define WM8961_DACSMM_SHIFT 3 /* DACSMM */ +#define WM8961_DACSMM_WIDTH 1 /* DACSMM */ +#define WM8961_DACMR 0x0004 /* DACMR */ +#define WM8961_DACMR_MASK 0x0004 /* DACMR */ +#define WM8961_DACMR_SHIFT 2 /* DACMR */ +#define WM8961_DACMR_WIDTH 1 /* DACMR */ +#define WM8961_DACSLOPE 0x0002 /* DACSLOPE */ +#define WM8961_DACSLOPE_MASK 0x0002 /* DACSLOPE */ +#define WM8961_DACSLOPE_SHIFT 1 /* DACSLOPE */ +#define WM8961_DACSLOPE_WIDTH 1 /* DACSLOPE */ +#define WM8961_DAC_OSR128 0x0001 /* DAC_OSR128 */ +#define WM8961_DAC_OSR128_MASK 0x0001 /* DAC_OSR128 */ +#define WM8961_DAC_OSR128_SHIFT 0 /* DAC_OSR128 */ +#define WM8961_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */ + +/* + * R7 (0x07) - Audio Interface 0 + */ +#define WM8961_ALRSWAP 0x0100 /* ALRSWAP */ +#define WM8961_ALRSWAP_MASK 0x0100 /* ALRSWAP */ +#define WM8961_ALRSWAP_SHIFT 8 /* ALRSWAP */ +#define WM8961_ALRSWAP_WIDTH 1 /* ALRSWAP */ +#define WM8961_BCLKINV 0x0080 /* BCLKINV */ +#define WM8961_BCLKINV_MASK 0x0080 /* BCLKINV */ +#define WM8961_BCLKINV_SHIFT 7 /* BCLKINV */ +#define WM8961_BCLKINV_WIDTH 1 /* BCLKINV */ +#define WM8961_MS 0x0040 /* MS */ +#define WM8961_MS_MASK 0x0040 /* MS */ +#define WM8961_MS_SHIFT 6 /* MS */ +#define WM8961_MS_WIDTH 1 /* MS */ +#define WM8961_DLRSWAP 0x0020 /* DLRSWAP */ +#define WM8961_DLRSWAP_MASK 0x0020 /* DLRSWAP */ +#define WM8961_DLRSWAP_SHIFT 5 /* DLRSWAP */ +#define WM8961_DLRSWAP_WIDTH 1 /* DLRSWAP */ +#define WM8961_LRP 0x0010 /* LRP */ +#define WM8961_LRP_MASK 0x0010 /* LRP */ +#define WM8961_LRP_SHIFT 4 /* LRP */ +#define WM8961_LRP_WIDTH 1 /* LRP */ +#define WM8961_WL_MASK 0x000C /* WL - [3:2] */ +#define WM8961_WL_SHIFT 2 /* WL - [3:2] */ +#define WM8961_WL_WIDTH 2 /* WL - [3:2] */ +#define WM8961_FORMAT_MASK 0x0003 /* FORMAT - [1:0] */ +#define WM8961_FORMAT_SHIFT 0 /* FORMAT - [1:0] */ +#define WM8961_FORMAT_WIDTH 2 /* FORMAT - [1:0] */ + +/* + * R8 (0x08) - Clocking2 + */ +#define WM8961_DCLKDIV_MASK 0x01C0 /* DCLKDIV - [8:6] */ +#define WM8961_DCLKDIV_SHIFT 6 /* DCLKDIV - [8:6] */ +#define WM8961_DCLKDIV_WIDTH 3 /* DCLKDIV - [8:6] */ +#define WM8961_CLK_SYS_ENA 0x0020 /* CLK_SYS_ENA */ +#define WM8961_CLK_SYS_ENA_MASK 0x0020 /* CLK_SYS_ENA */ +#define WM8961_CLK_SYS_ENA_SHIFT 5 /* CLK_SYS_ENA */ +#define WM8961_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ +#define WM8961_CLK_DSP_ENA 0x0010 /* CLK_DSP_ENA */ +#define WM8961_CLK_DSP_ENA_MASK 0x0010 /* CLK_DSP_ENA */ +#define WM8961_CLK_DSP_ENA_SHIFT 4 /* CLK_DSP_ENA */ +#define WM8961_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ +#define WM8961_BCLKDIV_MASK 0x000F /* BCLKDIV - [3:0] */ +#define WM8961_BCLKDIV_SHIFT 0 /* BCLKDIV - [3:0] */ +#define WM8961_BCLKDIV_WIDTH 4 /* BCLKDIV - [3:0] */ + +/* + * R9 (0x09) - Audio Interface 1 + */ +#define WM8961_DACCOMP_MASK 0x0018 /* DACCOMP - [4:3] */ +#define WM8961_DACCOMP_SHIFT 3 /* DACCOMP - [4:3] */ +#define WM8961_DACCOMP_WIDTH 2 /* DACCOMP - [4:3] */ +#define WM8961_ADCCOMP_MASK 0x0006 /* ADCCOMP - [2:1] */ +#define WM8961_ADCCOMP_SHIFT 1 /* ADCCOMP - [2:1] */ +#define WM8961_ADCCOMP_WIDTH 2 /* ADCCOMP - [2:1] */ +#define WM8961_LOOPBACK 0x0001 /* LOOPBACK */ +#define WM8961_LOOPBACK_MASK 0x0001 /* LOOPBACK */ +#define WM8961_LOOPBACK_SHIFT 0 /* LOOPBACK */ +#define WM8961_LOOPBACK_WIDTH 1 /* LOOPBACK */ + +/* + * R10 (0x0A) - Left DAC volume + */ +#define WM8961_DACVU 0x0100 /* DACVU */ +#define WM8961_DACVU_MASK 0x0100 /* DACVU */ +#define WM8961_DACVU_SHIFT 8 /* DACVU */ +#define WM8961_DACVU_WIDTH 1 /* DACVU */ +#define WM8961_LDACVOL_MASK 0x00FF /* LDACVOL - [7:0] */ +#define WM8961_LDACVOL_SHIFT 0 /* LDACVOL - [7:0] */ +#define WM8961_LDACVOL_WIDTH 8 /* LDACVOL - [7:0] */ + +/* + * R11 (0x0B) - Right DAC volume + */ +#define WM8961_DACVU 0x0100 /* DACVU */ +#define WM8961_DACVU_MASK 0x0100 /* DACVU */ +#define WM8961_DACVU_SHIFT 8 /* DACVU */ +#define WM8961_DACVU_WIDTH 1 /* DACVU */ +#define WM8961_RDACVOL_MASK 0x00FF /* RDACVOL - [7:0] */ +#define WM8961_RDACVOL_SHIFT 0 /* RDACVOL - [7:0] */ +#define WM8961_RDACVOL_WIDTH 8 /* RDACVOL - [7:0] */ + +/* + * R14 (0x0E) - Audio Interface 2 + */ +#define WM8961_LRCLK_RATE_MASK 0x01FF /* LRCLK_RATE - [8:0] */ +#define WM8961_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [8:0] */ +#define WM8961_LRCLK_RATE_WIDTH 9 /* LRCLK_RATE - [8:0] */ + +/* + * R15 (0x0F) - Software Reset + */ +#define WM8961_SW_RST_DEV_ID1_MASK 0xFFFF /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8961_SW_RST_DEV_ID1_SHIFT 0 /* SW_RST_DEV_ID1 - [15:0] */ +#define WM8961_SW_RST_DEV_ID1_WIDTH 16 /* SW_RST_DEV_ID1 - [15:0] */ + +/* + * R17 (0x11) - ALC1 + */ +#define WM8961_ALCSEL_MASK 0x0180 /* ALCSEL - [8:7] */ +#define WM8961_ALCSEL_SHIFT 7 /* ALCSEL - [8:7] */ +#define WM8961_ALCSEL_WIDTH 2 /* ALCSEL - [8:7] */ +#define WM8961_MAXGAIN_MASK 0x0070 /* MAXGAIN - [6:4] */ +#define WM8961_MAXGAIN_SHIFT 4 /* MAXGAIN - [6:4] */ +#define WM8961_MAXGAIN_WIDTH 3 /* MAXGAIN - [6:4] */ +#define WM8961_ALCL_MASK 0x000F /* ALCL - [3:0] */ +#define WM8961_ALCL_SHIFT 0 /* ALCL - [3:0] */ +#define WM8961_ALCL_WIDTH 4 /* ALCL - [3:0] */ + +/* + * R18 (0x12) - ALC2 + */ +#define WM8961_ALCZC 0x0080 /* ALCZC */ +#define WM8961_ALCZC_MASK 0x0080 /* ALCZC */ +#define WM8961_ALCZC_SHIFT 7 /* ALCZC */ +#define WM8961_ALCZC_WIDTH 1 /* ALCZC */ +#define WM8961_MINGAIN_MASK 0x0070 /* MINGAIN - [6:4] */ +#define WM8961_MINGAIN_SHIFT 4 /* MINGAIN - [6:4] */ +#define WM8961_MINGAIN_WIDTH 3 /* MINGAIN - [6:4] */ +#define WM8961_HLD_MASK 0x000F /* HLD - [3:0] */ +#define WM8961_HLD_SHIFT 0 /* HLD - [3:0] */ +#define WM8961_HLD_WIDTH 4 /* HLD - [3:0] */ + +/* + * R19 (0x13) - ALC3 + */ +#define WM8961_ALCMODE 0x0100 /* ALCMODE */ +#define WM8961_ALCMODE_MASK 0x0100 /* ALCMODE */ +#define WM8961_ALCMODE_SHIFT 8 /* ALCMODE */ +#define WM8961_ALCMODE_WIDTH 1 /* ALCMODE */ +#define WM8961_DCY_MASK 0x00F0 /* DCY - [7:4] */ +#define WM8961_DCY_SHIFT 4 /* DCY - [7:4] */ +#define WM8961_DCY_WIDTH 4 /* DCY - [7:4] */ +#define WM8961_ATK_MASK 0x000F /* ATK - [3:0] */ +#define WM8961_ATK_SHIFT 0 /* ATK - [3:0] */ +#define WM8961_ATK_WIDTH 4 /* ATK - [3:0] */ + +/* + * R20 (0x14) - Noise Gate + */ +#define WM8961_NGTH_MASK 0x00F8 /* NGTH - [7:3] */ +#define WM8961_NGTH_SHIFT 3 /* NGTH - [7:3] */ +#define WM8961_NGTH_WIDTH 5 /* NGTH - [7:3] */ +#define WM8961_NGG 0x0002 /* NGG */ +#define WM8961_NGG_MASK 0x0002 /* NGG */ +#define WM8961_NGG_SHIFT 1 /* NGG */ +#define WM8961_NGG_WIDTH 1 /* NGG */ +#define WM8961_NGAT 0x0001 /* NGAT */ +#define WM8961_NGAT_MASK 0x0001 /* NGAT */ +#define WM8961_NGAT_SHIFT 0 /* NGAT */ +#define WM8961_NGAT_WIDTH 1 /* NGAT */ + +/* + * R21 (0x15) - Left ADC volume + */ +#define WM8961_ADCVU 0x0100 /* ADCVU */ +#define WM8961_ADCVU_MASK 0x0100 /* ADCVU */ +#define WM8961_ADCVU_SHIFT 8 /* ADCVU */ +#define WM8961_ADCVU_WIDTH 1 /* ADCVU */ +#define WM8961_LADCVOL_MASK 0x00FF /* LADCVOL - [7:0] */ +#define WM8961_LADCVOL_SHIFT 0 /* LADCVOL - [7:0] */ +#define WM8961_LADCVOL_WIDTH 8 /* LADCVOL - [7:0] */ + +/* + * R22 (0x16) - Right ADC volume + */ +#define WM8961_ADCVU 0x0100 /* ADCVU */ +#define WM8961_ADCVU_MASK 0x0100 /* ADCVU */ +#define WM8961_ADCVU_SHIFT 8 /* ADCVU */ +#define WM8961_ADCVU_WIDTH 1 /* ADCVU */ +#define WM8961_RADCVOL_MASK 0x00FF /* RADCVOL - [7:0] */ +#define WM8961_RADCVOL_SHIFT 0 /* RADCVOL - [7:0] */ +#define WM8961_RADCVOL_WIDTH 8 /* RADCVOL - [7:0] */ + +/* + * R23 (0x17) - Additional control(1) + */ +#define WM8961_TSDEN 0x0100 /* TSDEN */ +#define WM8961_TSDEN_MASK 0x0100 /* TSDEN */ +#define WM8961_TSDEN_SHIFT 8 /* TSDEN */ +#define WM8961_TSDEN_WIDTH 1 /* TSDEN */ +#define WM8961_DMONOMIX 0x0010 /* DMONOMIX */ +#define WM8961_DMONOMIX_MASK 0x0010 /* DMONOMIX */ +#define WM8961_DMONOMIX_SHIFT 4 /* DMONOMIX */ +#define WM8961_DMONOMIX_WIDTH 1 /* DMONOMIX */ +#define WM8961_TOEN 0x0001 /* TOEN */ +#define WM8961_TOEN_MASK 0x0001 /* TOEN */ +#define WM8961_TOEN_SHIFT 0 /* TOEN */ +#define WM8961_TOEN_WIDTH 1 /* TOEN */ + +/* + * R24 (0x18) - Additional control(2) + */ +#define WM8961_TRIS 0x0008 /* TRIS */ +#define WM8961_TRIS_MASK 0x0008 /* TRIS */ +#define WM8961_TRIS_SHIFT 3 /* TRIS */ +#define WM8961_TRIS_WIDTH 1 /* TRIS */ + +/* + * R25 (0x19) - Pwr Mgmt (1) + */ +#define WM8961_VMIDSEL_MASK 0x0180 /* VMIDSEL - [8:7] */ +#define WM8961_VMIDSEL_SHIFT 7 /* VMIDSEL - [8:7] */ +#define WM8961_VMIDSEL_WIDTH 2 /* VMIDSEL - [8:7] */ +#define WM8961_VREF 0x0040 /* VREF */ +#define WM8961_VREF_MASK 0x0040 /* VREF */ +#define WM8961_VREF_SHIFT 6 /* VREF */ +#define WM8961_VREF_WIDTH 1 /* VREF */ +#define WM8961_AINL 0x0020 /* AINL */ +#define WM8961_AINL_MASK 0x0020 /* AINL */ +#define WM8961_AINL_SHIFT 5 /* AINL */ +#define WM8961_AINL_WIDTH 1 /* AINL */ +#define WM8961_AINR 0x0010 /* AINR */ +#define WM8961_AINR_MASK 0x0010 /* AINR */ +#define WM8961_AINR_SHIFT 4 /* AINR */ +#define WM8961_AINR_WIDTH 1 /* AINR */ +#define WM8961_ADCL 0x0008 /* ADCL */ +#define WM8961_ADCL_MASK 0x0008 /* ADCL */ +#define WM8961_ADCL_SHIFT 3 /* ADCL */ +#define WM8961_ADCL_WIDTH 1 /* ADCL */ +#define WM8961_ADCR 0x0004 /* ADCR */ +#define WM8961_ADCR_MASK 0x0004 /* ADCR */ +#define WM8961_ADCR_SHIFT 2 /* ADCR */ +#define WM8961_ADCR_WIDTH 1 /* ADCR */ +#define WM8961_MICB 0x0002 /* MICB */ +#define WM8961_MICB_MASK 0x0002 /* MICB */ +#define WM8961_MICB_SHIFT 1 /* MICB */ +#define WM8961_MICB_WIDTH 1 /* MICB */ + +/* + * R26 (0x1A) - Pwr Mgmt (2) + */ +#define WM8961_DACL 0x0100 /* DACL */ +#define WM8961_DACL_MASK 0x0100 /* DACL */ +#define WM8961_DACL_SHIFT 8 /* DACL */ +#define WM8961_DACL_WIDTH 1 /* DACL */ +#define WM8961_DACR 0x0080 /* DACR */ +#define WM8961_DACR_MASK 0x0080 /* DACR */ +#define WM8961_DACR_SHIFT 7 /* DACR */ +#define WM8961_DACR_WIDTH 1 /* DACR */ +#define WM8961_LOUT1_PGA 0x0040 /* LOUT1_PGA */ +#define WM8961_LOUT1_PGA_MASK 0x0040 /* LOUT1_PGA */ +#define WM8961_LOUT1_PGA_SHIFT 6 /* LOUT1_PGA */ +#define WM8961_LOUT1_PGA_WIDTH 1 /* LOUT1_PGA */ +#define WM8961_ROUT1_PGA 0x0020 /* ROUT1_PGA */ +#define WM8961_ROUT1_PGA_MASK 0x0020 /* ROUT1_PGA */ +#define WM8961_ROUT1_PGA_SHIFT 5 /* ROUT1_PGA */ +#define WM8961_ROUT1_PGA_WIDTH 1 /* ROUT1_PGA */ +#define WM8961_SPKL_PGA 0x0010 /* SPKL_PGA */ +#define WM8961_SPKL_PGA_MASK 0x0010 /* SPKL_PGA */ +#define WM8961_SPKL_PGA_SHIFT 4 /* SPKL_PGA */ +#define WM8961_SPKL_PGA_WIDTH 1 /* SPKL_PGA */ +#define WM8961_SPKR_PGA 0x0008 /* SPKR_PGA */ +#define WM8961_SPKR_PGA_MASK 0x0008 /* SPKR_PGA */ +#define WM8961_SPKR_PGA_SHIFT 3 /* SPKR_PGA */ +#define WM8961_SPKR_PGA_WIDTH 1 /* SPKR_PGA */ + +/* + * R27 (0x1B) - Additional Control (3) + */ +#define WM8961_SAMPLE_RATE_MASK 0x0007 /* SAMPLE_RATE - [2:0] */ +#define WM8961_SAMPLE_RATE_SHIFT 0 /* SAMPLE_RATE - [2:0] */ +#define WM8961_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [2:0] */ + +/* + * R28 (0x1C) - Anti-pop + */ +#define WM8961_BUFDCOPEN 0x0010 /* BUFDCOPEN */ +#define WM8961_BUFDCOPEN_MASK 0x0010 /* BUFDCOPEN */ +#define WM8961_BUFDCOPEN_SHIFT 4 /* BUFDCOPEN */ +#define WM8961_BUFDCOPEN_WIDTH 1 /* BUFDCOPEN */ +#define WM8961_BUFIOEN 0x0008 /* BUFIOEN */ +#define WM8961_BUFIOEN_MASK 0x0008 /* BUFIOEN */ +#define WM8961_BUFIOEN_SHIFT 3 /* BUFIOEN */ +#define WM8961_BUFIOEN_WIDTH 1 /* BUFIOEN */ +#define WM8961_SOFT_ST 0x0004 /* SOFT_ST */ +#define WM8961_SOFT_ST_MASK 0x0004 /* SOFT_ST */ +#define WM8961_SOFT_ST_SHIFT 2 /* SOFT_ST */ +#define WM8961_SOFT_ST_WIDTH 1 /* SOFT_ST */ + +/* + * R30 (0x1E) - Clocking 3 + */ +#define WM8961_CLK_TO_DIV_MASK 0x0180 /* CLK_TO_DIV - [8:7] */ +#define WM8961_CLK_TO_DIV_SHIFT 7 /* CLK_TO_DIV - [8:7] */ +#define WM8961_CLK_TO_DIV_WIDTH 2 /* CLK_TO_DIV - [8:7] */ +#define WM8961_CLK_256K_DIV_MASK 0x007E /* CLK_256K_DIV - [6:1] */ +#define WM8961_CLK_256K_DIV_SHIFT 1 /* CLK_256K_DIV - [6:1] */ +#define WM8961_CLK_256K_DIV_WIDTH 6 /* CLK_256K_DIV - [6:1] */ +#define WM8961_MANUAL_MODE 0x0001 /* MANUAL_MODE */ +#define WM8961_MANUAL_MODE_MASK 0x0001 /* MANUAL_MODE */ +#define WM8961_MANUAL_MODE_SHIFT 0 /* MANUAL_MODE */ +#define WM8961_MANUAL_MODE_WIDTH 1 /* MANUAL_MODE */ + +/* + * R32 (0x20) - ADCL signal path + */ +#define WM8961_LMICBOOST_MASK 0x0030 /* LMICBOOST - [5:4] */ +#define WM8961_LMICBOOST_SHIFT 4 /* LMICBOOST - [5:4] */ +#define WM8961_LMICBOOST_WIDTH 2 /* LMICBOOST - [5:4] */ + +/* + * R33 (0x21) - ADCR signal path + */ +#define WM8961_RMICBOOST_MASK 0x0030 /* RMICBOOST - [5:4] */ +#define WM8961_RMICBOOST_SHIFT 4 /* RMICBOOST - [5:4] */ +#define WM8961_RMICBOOST_WIDTH 2 /* RMICBOOST - [5:4] */ + +/* + * R40 (0x28) - LOUT2 volume + */ +#define WM8961_SPKVU 0x0100 /* SPKVU */ +#define WM8961_SPKVU_MASK 0x0100 /* SPKVU */ +#define WM8961_SPKVU_SHIFT 8 /* SPKVU */ +#define WM8961_SPKVU_WIDTH 1 /* SPKVU */ +#define WM8961_SPKLZC 0x0080 /* SPKLZC */ +#define WM8961_SPKLZC_MASK 0x0080 /* SPKLZC */ +#define WM8961_SPKLZC_SHIFT 7 /* SPKLZC */ +#define WM8961_SPKLZC_WIDTH 1 /* SPKLZC */ +#define WM8961_SPKLVOL_MASK 0x007F /* SPKLVOL - [6:0] */ +#define WM8961_SPKLVOL_SHIFT 0 /* SPKLVOL - [6:0] */ +#define WM8961_SPKLVOL_WIDTH 7 /* SPKLVOL - [6:0] */ + +/* + * R41 (0x29) - ROUT2 volume + */ +#define WM8961_SPKVU 0x0100 /* SPKVU */ +#define WM8961_SPKVU_MASK 0x0100 /* SPKVU */ +#define WM8961_SPKVU_SHIFT 8 /* SPKVU */ +#define WM8961_SPKVU_WIDTH 1 /* SPKVU */ +#define WM8961_SPKRZC 0x0080 /* SPKRZC */ +#define WM8961_SPKRZC_MASK 0x0080 /* SPKRZC */ +#define WM8961_SPKRZC_SHIFT 7 /* SPKRZC */ +#define WM8961_SPKRZC_WIDTH 1 /* SPKRZC */ +#define WM8961_SPKRVOL_MASK 0x007F /* SPKRVOL - [6:0] */ +#define WM8961_SPKRVOL_SHIFT 0 /* SPKRVOL - [6:0] */ +#define WM8961_SPKRVOL_WIDTH 7 /* SPKRVOL - [6:0] */ + +/* + * R47 (0x2F) - Pwr Mgmt (3) + */ +#define WM8961_TEMP_SHUT 0x0002 /* TEMP_SHUT */ +#define WM8961_TEMP_SHUT_MASK 0x0002 /* TEMP_SHUT */ +#define WM8961_TEMP_SHUT_SHIFT 1 /* TEMP_SHUT */ +#define WM8961_TEMP_SHUT_WIDTH 1 /* TEMP_SHUT */ +#define WM8961_TEMP_WARN 0x0001 /* TEMP_WARN */ +#define WM8961_TEMP_WARN_MASK 0x0001 /* TEMP_WARN */ +#define WM8961_TEMP_WARN_SHIFT 0 /* TEMP_WARN */ +#define WM8961_TEMP_WARN_WIDTH 1 /* TEMP_WARN */ + +/* + * R48 (0x30) - Additional Control (4) + */ +#define WM8961_TSENSEN 0x0002 /* TSENSEN */ +#define WM8961_TSENSEN_MASK 0x0002 /* TSENSEN */ +#define WM8961_TSENSEN_SHIFT 1 /* TSENSEN */ +#define WM8961_TSENSEN_WIDTH 1 /* TSENSEN */ +#define WM8961_MBSEL 0x0001 /* MBSEL */ +#define WM8961_MBSEL_MASK 0x0001 /* MBSEL */ +#define WM8961_MBSEL_SHIFT 0 /* MBSEL */ +#define WM8961_MBSEL_WIDTH 1 /* MBSEL */ + +/* + * R49 (0x31) - Class D Control 1 + */ +#define WM8961_SPKR_ENA 0x0080 /* SPKR_ENA */ +#define WM8961_SPKR_ENA_MASK 0x0080 /* SPKR_ENA */ +#define WM8961_SPKR_ENA_SHIFT 7 /* SPKR_ENA */ +#define WM8961_SPKR_ENA_WIDTH 1 /* SPKR_ENA */ +#define WM8961_SPKL_ENA 0x0040 /* SPKL_ENA */ +#define WM8961_SPKL_ENA_MASK 0x0040 /* SPKL_ENA */ +#define WM8961_SPKL_ENA_SHIFT 6 /* SPKL_ENA */ +#define WM8961_SPKL_ENA_WIDTH 1 /* SPKL_ENA */ + +/* + * R51 (0x33) - Class D Control 2 + */ +#define WM8961_CLASSD_ACGAIN_MASK 0x0007 /* CLASSD_ACGAIN - [2:0] */ +#define WM8961_CLASSD_ACGAIN_SHIFT 0 /* CLASSD_ACGAIN - [2:0] */ +#define WM8961_CLASSD_ACGAIN_WIDTH 3 /* CLASSD_ACGAIN - [2:0] */ + +/* + * R56 (0x38) - Clocking 4 + */ +#define WM8961_CLK_DCS_DIV_MASK 0x01E0 /* CLK_DCS_DIV - [8:5] */ +#define WM8961_CLK_DCS_DIV_SHIFT 5 /* CLK_DCS_DIV - [8:5] */ +#define WM8961_CLK_DCS_DIV_WIDTH 4 /* CLK_DCS_DIV - [8:5] */ +#define WM8961_CLK_SYS_RATE_MASK 0x001E /* CLK_SYS_RATE - [4:1] */ +#define WM8961_CLK_SYS_RATE_SHIFT 1 /* CLK_SYS_RATE - [4:1] */ +#define WM8961_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [4:1] */ + +/* + * R57 (0x39) - DSP Sidetone 0 + */ +#define WM8961_ADCR_DAC_SVOL_MASK 0x00F0 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8961_ADCR_DAC_SVOL_SHIFT 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8961_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [7:4] */ +#define WM8961_ADC_TO_DACR_MASK 0x000C /* ADC_TO_DACR - [3:2] */ +#define WM8961_ADC_TO_DACR_SHIFT 2 /* ADC_TO_DACR - [3:2] */ +#define WM8961_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [3:2] */ + +/* + * R58 (0x3A) - DSP Sidetone 1 + */ +#define WM8961_ADCL_DAC_SVOL_MASK 0x00F0 /* ADCL_DAC_SVOL - [7:4] */ +#define WM8961_ADCL_DAC_SVOL_SHIFT 4 /* ADCL_DAC_SVOL - [7:4] */ +#define WM8961_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [7:4] */ +#define WM8961_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */ +#define WM8961_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */ +#define WM8961_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */ + +/* + * R60 (0x3C) - DC Servo 0 + */ +#define WM8961_DCS_ENA_CHAN_INL 0x0080 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_ENA_CHAN_INL_MASK 0x0080 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_ENA_CHAN_INL_SHIFT 7 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_ENA_CHAN_INL_WIDTH 1 /* DCS_ENA_CHAN_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL 0x0040 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL_MASK 0x0040 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL_SHIFT 6 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_STARTUP_INL_WIDTH 1 /* DCS_TRIG_STARTUP_INL */ +#define WM8961_DCS_TRIG_SERIES_INL 0x0010 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_TRIG_SERIES_INL_MASK 0x0010 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_TRIG_SERIES_INL_SHIFT 4 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_TRIG_SERIES_INL_WIDTH 1 /* DCS_TRIG_SERIES_INL */ +#define WM8961_DCS_ENA_CHAN_INR 0x0008 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_ENA_CHAN_INR_MASK 0x0008 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_ENA_CHAN_INR_SHIFT 3 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_ENA_CHAN_INR_WIDTH 1 /* DCS_ENA_CHAN_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR 0x0004 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR_MASK 0x0004 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR_SHIFT 2 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_STARTUP_INR_WIDTH 1 /* DCS_TRIG_STARTUP_INR */ +#define WM8961_DCS_TRIG_SERIES_INR 0x0001 /* DCS_TRIG_SERIES_INR */ +#define WM8961_DCS_TRIG_SERIES_INR_MASK 0x0001 /* DCS_TRIG_SERIES_INR */ +#define WM8961_DCS_TRIG_SERIES_INR_SHIFT 0 /* DCS_TRIG_SERIES_INR */ +#define WM8961_DCS_TRIG_SERIES_INR_WIDTH 1 /* DCS_TRIG_SERIES_INR */ + +/* + * R61 (0x3D) - DC Servo 1 + */ +#define WM8961_DCS_ENA_CHAN_HPL 0x0080 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_ENA_CHAN_HPL_MASK 0x0080 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_ENA_CHAN_HPL_SHIFT 7 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_ENA_CHAN_HPL_WIDTH 1 /* DCS_ENA_CHAN_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL 0x0040 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL_MASK 0x0040 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL_SHIFT 6 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_STARTUP_HPL_WIDTH 1 /* DCS_TRIG_STARTUP_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL 0x0010 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL_MASK 0x0010 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL_SHIFT 4 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_TRIG_SERIES_HPL_WIDTH 1 /* DCS_TRIG_SERIES_HPL */ +#define WM8961_DCS_ENA_CHAN_HPR 0x0008 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_ENA_CHAN_HPR_MASK 0x0008 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_ENA_CHAN_HPR_SHIFT 3 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_ENA_CHAN_HPR_WIDTH 1 /* DCS_ENA_CHAN_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR 0x0004 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR_MASK 0x0004 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR_SHIFT 2 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_STARTUP_HPR_WIDTH 1 /* DCS_TRIG_STARTUP_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR 0x0001 /* DCS_TRIG_SERIES_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR_MASK 0x0001 /* DCS_TRIG_SERIES_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR_SHIFT 0 /* DCS_TRIG_SERIES_HPR */ +#define WM8961_DCS_TRIG_SERIES_HPR_WIDTH 1 /* DCS_TRIG_SERIES_HPR */ + +/* + * R63 (0x3F) - DC Servo 3 + */ +#define WM8961_DCS_FILT_BW_SERIES_MASK 0x0030 /* DCS_FILT_BW_SERIES - [5:4] */ +#define WM8961_DCS_FILT_BW_SERIES_SHIFT 4 /* DCS_FILT_BW_SERIES - [5:4] */ +#define WM8961_DCS_FILT_BW_SERIES_WIDTH 2 /* DCS_FILT_BW_SERIES - [5:4] */ + +/* + * R65 (0x41) - DC Servo 5 + */ +#define WM8961_DCS_SERIES_NO_HP_MASK 0x007F /* DCS_SERIES_NO_HP - [6:0] */ +#define WM8961_DCS_SERIES_NO_HP_SHIFT 0 /* DCS_SERIES_NO_HP - [6:0] */ +#define WM8961_DCS_SERIES_NO_HP_WIDTH 7 /* DCS_SERIES_NO_HP - [6:0] */ + +/* + * R68 (0x44) - Analogue PGA Bias + */ +#define WM8961_HP_PGAS_BIAS_MASK 0x0007 /* HP_PGAS_BIAS - [2:0] */ +#define WM8961_HP_PGAS_BIAS_SHIFT 0 /* HP_PGAS_BIAS - [2:0] */ +#define WM8961_HP_PGAS_BIAS_WIDTH 3 /* HP_PGAS_BIAS - [2:0] */ + +/* + * R69 (0x45) - Analogue HP 0 + */ +#define WM8961_HPL_RMV_SHORT 0x0080 /* HPL_RMV_SHORT */ +#define WM8961_HPL_RMV_SHORT_MASK 0x0080 /* HPL_RMV_SHORT */ +#define WM8961_HPL_RMV_SHORT_SHIFT 7 /* HPL_RMV_SHORT */ +#define WM8961_HPL_RMV_SHORT_WIDTH 1 /* HPL_RMV_SHORT */ +#define WM8961_HPL_ENA_OUTP 0x0040 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_OUTP_MASK 0x0040 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_OUTP_SHIFT 6 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_OUTP_WIDTH 1 /* HPL_ENA_OUTP */ +#define WM8961_HPL_ENA_DLY 0x0020 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA_DLY_MASK 0x0020 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA_DLY_SHIFT 5 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA_DLY_WIDTH 1 /* HPL_ENA_DLY */ +#define WM8961_HPL_ENA 0x0010 /* HPL_ENA */ +#define WM8961_HPL_ENA_MASK 0x0010 /* HPL_ENA */ +#define WM8961_HPL_ENA_SHIFT 4 /* HPL_ENA */ +#define WM8961_HPL_ENA_WIDTH 1 /* HPL_ENA */ +#define WM8961_HPR_RMV_SHORT 0x0008 /* HPR_RMV_SHORT */ +#define WM8961_HPR_RMV_SHORT_MASK 0x0008 /* HPR_RMV_SHORT */ +#define WM8961_HPR_RMV_SHORT_SHIFT 3 /* HPR_RMV_SHORT */ +#define WM8961_HPR_RMV_SHORT_WIDTH 1 /* HPR_RMV_SHORT */ +#define WM8961_HPR_ENA_OUTP 0x0004 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_OUTP_MASK 0x0004 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_OUTP_SHIFT 2 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_OUTP_WIDTH 1 /* HPR_ENA_OUTP */ +#define WM8961_HPR_ENA_DLY 0x0002 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA_DLY_MASK 0x0002 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA_DLY_SHIFT 1 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA_DLY_WIDTH 1 /* HPR_ENA_DLY */ +#define WM8961_HPR_ENA 0x0001 /* HPR_ENA */ +#define WM8961_HPR_ENA_MASK 0x0001 /* HPR_ENA */ +#define WM8961_HPR_ENA_SHIFT 0 /* HPR_ENA */ +#define WM8961_HPR_ENA_WIDTH 1 /* HPR_ENA */ + +/* + * R71 (0x47) - Analogue HP 2 + */ +#define WM8961_HPL_VOL_MASK 0x01C0 /* HPL_VOL - [8:6] */ +#define WM8961_HPL_VOL_SHIFT 6 /* HPL_VOL - [8:6] */ +#define WM8961_HPL_VOL_WIDTH 3 /* HPL_VOL - [8:6] */ +#define WM8961_HPR_VOL_MASK 0x0038 /* HPR_VOL - [5:3] */ +#define WM8961_HPR_VOL_SHIFT 3 /* HPR_VOL - [5:3] */ +#define WM8961_HPR_VOL_WIDTH 3 /* HPR_VOL - [5:3] */ +#define WM8961_HP_BIAS_BOOST_MASK 0x0007 /* HP_BIAS_BOOST - [2:0] */ +#define WM8961_HP_BIAS_BOOST_SHIFT 0 /* HP_BIAS_BOOST - [2:0] */ +#define WM8961_HP_BIAS_BOOST_WIDTH 3 /* HP_BIAS_BOOST - [2:0] */ + +/* + * R72 (0x48) - Charge Pump 1 + */ +#define WM8961_CP_ENA 0x0001 /* CP_ENA */ +#define WM8961_CP_ENA_MASK 0x0001 /* CP_ENA */ +#define WM8961_CP_ENA_SHIFT 0 /* CP_ENA */ +#define WM8961_CP_ENA_WIDTH 1 /* CP_ENA */ + +/* + * R82 (0x52) - Charge Pump B + */ +#define WM8961_CP_DYN_PWR_MASK 0x0003 /* CP_DYN_PWR - [1:0] */ +#define WM8961_CP_DYN_PWR_SHIFT 0 /* CP_DYN_PWR - [1:0] */ +#define WM8961_CP_DYN_PWR_WIDTH 2 /* CP_DYN_PWR - [1:0] */ + +/* + * R87 (0x57) - Write Sequencer 1 + */ +#define WM8961_WSEQ_ENA 0x0020 /* WSEQ_ENA */ +#define WM8961_WSEQ_ENA_MASK 0x0020 /* WSEQ_ENA */ +#define WM8961_WSEQ_ENA_SHIFT 5 /* WSEQ_ENA */ +#define WM8961_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM8961_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8961_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8961_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */ + +/* + * R88 (0x58) - Write Sequencer 2 + */ +#define WM8961_WSEQ_EOS 0x0100 /* WSEQ_EOS */ +#define WM8961_WSEQ_EOS_MASK 0x0100 /* WSEQ_EOS */ +#define WM8961_WSEQ_EOS_SHIFT 8 /* WSEQ_EOS */ +#define WM8961_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */ +#define WM8961_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */ +#define WM8961_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */ +#define WM8961_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */ + +/* + * R89 (0x59) - Write Sequencer 3 + */ +#define WM8961_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */ +#define WM8961_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */ +#define WM8961_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */ + +/* + * R90 (0x5A) - Write Sequencer 4 + */ +#define WM8961_WSEQ_ABORT 0x0100 /* WSEQ_ABORT */ +#define WM8961_WSEQ_ABORT_MASK 0x0100 /* WSEQ_ABORT */ +#define WM8961_WSEQ_ABORT_SHIFT 8 /* WSEQ_ABORT */ +#define WM8961_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM8961_WSEQ_START 0x0080 /* WSEQ_START */ +#define WM8961_WSEQ_START_MASK 0x0080 /* WSEQ_START */ +#define WM8961_WSEQ_START_SHIFT 7 /* WSEQ_START */ +#define WM8961_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM8961_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */ +#define WM8961_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */ +#define WM8961_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */ + +/* + * R91 (0x5B) - Write Sequencer 5 + */ +#define WM8961_WSEQ_DATA_WIDTH_MASK 0x0070 /* WSEQ_DATA_WIDTH - [6:4] */ +#define WM8961_WSEQ_DATA_WIDTH_SHIFT 4 /* WSEQ_DATA_WIDTH - [6:4] */ +#define WM8961_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [6:4] */ +#define WM8961_WSEQ_DATA_START_MASK 0x000F /* WSEQ_DATA_START - [3:0] */ +#define WM8961_WSEQ_DATA_START_SHIFT 0 /* WSEQ_DATA_START - [3:0] */ +#define WM8961_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [3:0] */ + +/* + * R92 (0x5C) - Write Sequencer 6 + */ +#define WM8961_WSEQ_DELAY_MASK 0x000F /* WSEQ_DELAY - [3:0] */ +#define WM8961_WSEQ_DELAY_SHIFT 0 /* WSEQ_DELAY - [3:0] */ +#define WM8961_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [3:0] */ + +/* + * R93 (0x5D) - Write Sequencer 7 + */ +#define WM8961_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM8961_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM8961_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM8961_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R252 (0xFC) - General test 1 + */ +#define WM8961_ARA_ENA 0x0002 /* ARA_ENA */ +#define WM8961_ARA_ENA_MASK 0x0002 /* ARA_ENA */ +#define WM8961_ARA_ENA_SHIFT 1 /* ARA_ENA */ +#define WM8961_ARA_ENA_WIDTH 1 /* ARA_ENA */ +#define WM8961_AUTO_INC 0x0001 /* AUTO_INC */ +#define WM8961_AUTO_INC_MASK 0x0001 /* AUTO_INC */ +#define WM8961_AUTO_INC_SHIFT 0 /* AUTO_INC */ +#define WM8961_AUTO_INC_WIDTH 1 /* AUTO_INC */ + +#endif diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 032dca22dbd3..d66efb0546ea 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -59,44 +59,7 @@ static const u16 wm8971_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; -static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg < WM8971_REG_COUNT) - return cache[reg]; - - return -1; -} - -static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg < WM8971_REG_COUNT) - cache[reg] = value; -} - -static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8971_write_reg_cache (codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0) +#define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; @@ -521,7 +484,7 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8971_write(codec, WM8971_IFACE, iface); + snd_soc_write(codec, WM8971_IFACE, iface); return 0; } @@ -533,8 +496,8 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8971_priv *wm8971 = codec->private_data; - u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3; - u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0; + u16 iface = snd_soc_read(codec, WM8971_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8971_SRATE) & 0x1c0; int coeff = get_coeff(wm8971->sysclk, params_rate(params)); /* bit size */ @@ -553,9 +516,9 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8971_write(codec, WM8971_IFACE, iface); + snd_soc_write(codec, WM8971_IFACE, iface); if (coeff >= 0) - wm8971_write(codec, WM8971_SRATE, srate | + snd_soc_write(codec, WM8971_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -564,33 +527,33 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8971_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8971_ADCDAC) & 0xfff7; if (mute) - wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8971_ADCDAC, mute_reg | 0x8); else - wm8971_write(codec, WM8971_ADCDAC, mute_reg); + snd_soc_write(codec, WM8971_ADCDAC, mute_reg); return 0; } static int wm8971_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; + u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; switch (level) { case SND_SOC_BIAS_ON: /* set vmid to 50k and unmute dac */ - wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: /* mute dac and set vmid to 500k, enable VREF */ - wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140); + snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140); break; case SND_SOC_BIAS_OFF: - wm8971_write(codec, WM8971_PWR1, 0x0001); + snd_soc_write(codec, WM8971_PWR1, 0x0001); break; } codec->bias_level = level; @@ -667,8 +630,8 @@ static int wm8971_resume(struct platform_device *pdev) /* charge wm8971 caps */ if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { - reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; - wm8971_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->delayed_work, msecs_to_jiffies(1000)); @@ -677,15 +640,14 @@ static int wm8971_resume(struct platform_device *pdev) return 0; } -static int wm8971_init(struct snd_soc_device *socdev) +static int wm8971_init(struct snd_soc_device *socdev, + enum snd_soc_control_type control) { struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8971"; codec->owner = THIS_MODULE; - codec->read = wm8971_read_reg_cache; - codec->write = wm8971_write; codec->set_bias_level = wm8971_set_bias_level; codec->dai = &wm8971_dai; codec->reg_cache_size = ARRAY_SIZE(wm8971_reg); @@ -695,42 +657,48 @@ static int wm8971_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); + goto err; + } + wm8971_reset(codec); /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { printk(KERN_ERR "wm8971: failed to create pcms\n"); - goto pcm_err; + goto err; } /* charge output caps - set vmid to 5k for quick power up */ - reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; - wm8971_write(codec, WM8971_PWR1, reg | 0x01c0); + reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; + snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ - reg = wm8971_read_reg_cache(codec, WM8971_LDAC); - wm8971_write(codec, WM8971_LDAC, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_RDAC); - wm8971_write(codec, WM8971_RDAC, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V); - wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V); - wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V); - wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V); - wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100); - - reg = wm8971_read_reg_cache(codec, WM8971_LINVOL); - wm8971_write(codec, WM8971_LINVOL, reg | 0x0100); - reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); - wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_LDAC); + snd_soc_write(codec, WM8971_LDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_RDAC); + snd_soc_write(codec, WM8971_RDAC, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LOUT1V); + snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_ROUT1V); + snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LOUT2V); + snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_ROUT2V); + snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100); + + reg = snd_soc_read(codec, WM8971_LINVOL); + snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8971_RINVOL); + snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100); snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); @@ -745,7 +713,7 @@ static int wm8971_init(struct snd_soc_device *socdev) card_err: snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: +err: kfree(codec->reg_cache); return ret; } @@ -767,7 +735,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, codec->control_data = i2c; - ret = wm8971_init(socdev); + ret = wm8971_init(socdev, SND_SOC_I2C); if (ret < 0) pr_err("failed to initialise WM8971\n"); @@ -877,7 +845,6 @@ static int wm8971_probe(struct platform_device *pdev) #if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; ret = wm8971_add_i2c_device(pdev, setup); } #endif diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c new file mode 100644 index 000000000000..d8a013ab3177 --- /dev/null +++ b/sound/soc/codecs/wm8974.c @@ -0,0 +1,808 @@ +/* + * wm8974.c -- WM8974 ALSA Soc Audio driver + * + * Copyright 2006-2009 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood <linux@wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8974.h" + +static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0050, 0x0000, 0x0140, 0x0000, + 0x0000, 0x0000, 0x0000, 0x00ff, + 0x0000, 0x0000, 0x0100, 0x00ff, + 0x0000, 0x0000, 0x012c, 0x002c, + 0x002c, 0x002c, 0x002c, 0x0000, + 0x0032, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0038, 0x000b, 0x0032, 0x0000, + 0x0008, 0x000c, 0x0093, 0x00e9, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0003, 0x0010, 0x0000, 0x0000, + 0x0000, 0x0002, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0039, 0x0000, + 0x0000, +}; + +#define WM8974_POWER1_BIASEN 0x08 +#define WM8974_POWER1_BUFIOEN 0x10 + +struct wm8974_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8974_CACHEREGNUM]; +}; + +static struct snd_soc_codec *wm8974_codec; + +#define wm8974_reset(c) snd_soc_write(c, WM8974_RESET, 0) + +static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" }; +static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8974_eqmode[] = {"Capture", "Playback" }; +static const char *wm8974_bw[] = {"Narrow", "Wide" }; +static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; +static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; +static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; +static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; +static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; +static const char *wm8974_alc[] = {"ALC", "Limiter" }; + +static const struct soc_enum wm8974_enum[] = { + SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */ + SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */ + SOC_ENUM_SINGLE(WM8974_DAC, 4, 4, wm8974_deemp), + SOC_ENUM_SINGLE(WM8974_EQ1, 8, 2, wm8974_eqmode), + + SOC_ENUM_SINGLE(WM8974_EQ1, 5, 4, wm8974_eq1), + SOC_ENUM_SINGLE(WM8974_EQ2, 8, 2, wm8974_bw), + SOC_ENUM_SINGLE(WM8974_EQ2, 5, 4, wm8974_eq2), + SOC_ENUM_SINGLE(WM8974_EQ3, 8, 2, wm8974_bw), + + SOC_ENUM_SINGLE(WM8974_EQ3, 5, 4, wm8974_eq3), + SOC_ENUM_SINGLE(WM8974_EQ4, 8, 2, wm8974_bw), + SOC_ENUM_SINGLE(WM8974_EQ4, 5, 4, wm8974_eq4), + SOC_ENUM_SINGLE(WM8974_EQ5, 8, 2, wm8974_bw), + + SOC_ENUM_SINGLE(WM8974_EQ5, 5, 4, wm8974_eq5), + SOC_ENUM_SINGLE(WM8974_ALC3, 8, 2, wm8974_alc), +}; + +static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" }; + +static const struct soc_enum wm8974_auxmode = + SOC_ENUM_SINGLE(WM8974_INPUT, 3, 2, wm8974_auxmode_text); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0); +static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0); + +static const struct snd_kcontrol_new wm8974_snd_controls[] = { + +SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0), + +SOC_ENUM("DAC Companding", wm8974_enum[1]), +SOC_ENUM("ADC Companding", wm8974_enum[0]), + +SOC_ENUM("Playback De-emphasis", wm8974_enum[2]), +SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0), + +SOC_SINGLE_TLV("PCM Volume", WM8974_DACVOL, 0, 255, 0, digital_tlv), + +SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0), +SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0), +SOC_SINGLE("ADC Inversion Switch", WM8974_ADC, 0, 1, 0), + +SOC_SINGLE_TLV("Capture Volume", WM8974_ADCVOL, 0, 255, 0, digital_tlv), + +SOC_ENUM("Equaliser Function", wm8974_enum[3]), +SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]), +SOC_SINGLE_TLV("EQ1 Volume", WM8974_EQ1, 0, 24, 1, eq_tlv), + +SOC_ENUM("Equaliser EQ2 Bandwith", wm8974_enum[5]), +SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]), +SOC_SINGLE_TLV("EQ2 Volume", WM8974_EQ2, 0, 24, 1, eq_tlv), + +SOC_ENUM("Equaliser EQ3 Bandwith", wm8974_enum[7]), +SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]), +SOC_SINGLE_TLV("EQ3 Volume", WM8974_EQ3, 0, 24, 1, eq_tlv), + +SOC_ENUM("Equaliser EQ4 Bandwith", wm8974_enum[9]), +SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]), +SOC_SINGLE_TLV("EQ4 Volume", WM8974_EQ4, 0, 24, 1, eq_tlv), + +SOC_ENUM("Equaliser EQ5 Bandwith", wm8974_enum[11]), +SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]), +SOC_SINGLE_TLV("EQ5 Volume", WM8974_EQ5, 0, 24, 1, eq_tlv), + +SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1, 8, 1, 0), +SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1, 4, 15, 0), +SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1, 0, 15, 0), + +SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2, 4, 7, 0), +SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2, 0, 15, 0), + +SOC_SINGLE("ALC Enable Switch", WM8974_ALC1, 8, 1, 0), +SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1, 3, 7, 0), +SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1, 0, 7, 0), + +SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2, 8, 1, 0), +SOC_SINGLE("ALC Capture Hold", WM8974_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Capture Target", WM8974_ALC2, 0, 15, 0), + +SOC_ENUM("ALC Capture Mode", wm8974_enum[13]), +SOC_SINGLE("ALC Capture Decay", WM8974_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack", WM8974_ALC3, 0, 15, 0), + +SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE, 3, 1, 0), +SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE, 0, 7, 0), + +SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA, 7, 1, 0), +SOC_SINGLE_TLV("Capture PGA Volume", WM8974_INPPGA, 0, 63, 0, inpga_tlv), + +SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL, 7, 1, 0), +SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL, 6, 1, 1), +SOC_SINGLE_TLV("Speaker Playback Volume", WM8974_SPKVOL, 0, 63, 0, spk_tlv), + +SOC_ENUM("Aux Mode", wm8974_auxmode), + +SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1), +}; + +/* Speaker Output Mixer */ +static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1), +}; + +/* Mono Output Mixer */ +static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0), +}; + +/* Boost mixer */ +static const struct snd_kcontrol_new wm8974_boost_mixer[] = { +SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0), +}; + +/* Input PGA */ +static const struct snd_kcontrol_new wm8974_inpga[] = { +SOC_DAPM_SINGLE("Aux Switch", WM8974_INPUT, 2, 1, 0), +SOC_DAPM_SINGLE("MicN Switch", WM8974_INPUT, 1, 1, 0), +SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0), +}; + +/* AUX Input boost vol */ +static const struct snd_kcontrol_new wm8974_aux_boost_controls = +SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0); + +/* Mic Input boost vol */ +static const struct snd_kcontrol_new wm8974_mic_boost_controls = +SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0); + +static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0, + &wm8974_speaker_mixer_controls[0], + ARRAY_SIZE(wm8974_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0, + &wm8974_mono_mixer_controls[0], + ARRAY_SIZE(wm8974_mono_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER2, 0, 0), +SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0), + +SND_SOC_DAPM_MIXER("Input PGA", WM8974_POWER2, 2, 0, wm8974_inpga, + ARRAY_SIZE(wm8974_inpga)), +SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0, + wm8974_boost_mixer, ARRAY_SIZE(wm8974_boost_mixer)), + +SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0), + +SND_SOC_DAPM_INPUT("MICN"), +SND_SOC_DAPM_INPUT("MICP"), +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_OUTPUT("MONOOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Mono output mixer */ + {"Mono Mixer", "PCM Playback Switch", "DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, + {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Speaker output mixer */ + {"Speaker Mixer", "PCM Playback Switch", "DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, + {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, + + /* Outputs */ + {"Mono Out", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono Out"}, + {"SpkN Out", NULL, "Speaker Mixer"}, + {"SpkP Out", NULL, "Speaker Mixer"}, + {"SPKOUTN", NULL, "SpkN Out"}, + {"SPKOUTP", NULL, "SpkP Out"}, + + /* Boost Mixer */ + {"ADC", NULL, "Boost Mixer"}, + {"Boost Mixer", "Aux Switch", "Aux Input"}, + {"Boost Mixer", NULL, "Input PGA"}, + {"Boost Mixer", NULL, "MICP"}, + + /* Input PGA */ + {"Input PGA", "Aux Switch", "Aux Input"}, + {"Input PGA", "MicN Switch", "MICN"}, + {"Input PGA", "MicP Switch", "MICP"}, + + /* Inputs */ + {"Aux Input", NULL, "AUX"}, +}; + +static int wm8974_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8974_dapm_widgets, + ARRAY_SIZE(wm8974_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +struct pll_ { + unsigned int pre_div:4; /* prescale - 1 */ + unsigned int n:4; + unsigned int k; +}; + +static struct pll_ pll_div; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static void pll_factors(unsigned int target, unsigned int source) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div.pre_div = 1; + Ndiv = target / source; + } else + pll_div.pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) + printk(KERN_WARNING + "WM8974 N value %u outwith recommended range!\n", + Ndiv); + + pll_div.n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div.k = K; +} + +static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + if (freq_in == 0 || freq_out == 0) { + /* Clock CODEC directly from MCLK */ + reg = snd_soc_read(codec, WM8974_CLOCK); + snd_soc_write(codec, WM8974_CLOCK, reg & 0x0ff); + + /* Turn off PLL */ + reg = snd_soc_read(codec, WM8974_POWER1); + snd_soc_write(codec, WM8974_POWER1, reg & 0x1df); + return 0; + } + + pll_factors(freq_out*4, freq_in); + + snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n); + snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18); + snd_soc_write(codec, WM8974_PLLK2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8974_PLLK3, pll_div.k & 0x1ff); + reg = snd_soc_read(codec, WM8974_POWER1); + snd_soc_write(codec, WM8974_POWER1, reg | 0x020); + + /* Run CODEC from PLL instead of MCLK */ + reg = snd_soc_read(codec, WM8974_CLOCK); + snd_soc_write(codec, WM8974_CLOCK, reg | 0x100); + + return 0; +} + +/* + * Configure WM8974 clock dividers. + */ +static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8974_OPCLKDIV: + reg = snd_soc_read(codec, WM8974_GPIO) & 0x1cf; + snd_soc_write(codec, WM8974_GPIO, reg | div); + break; + case WM8974_MCLKDIV: + reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f; + snd_soc_write(codec, WM8974_CLOCK, reg | div); + break; + case WM8974_ADCCLK: + reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7; + snd_soc_write(codec, WM8974_ADC, reg | div); + break; + case WM8974_DACCLK: + reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7; + snd_soc_write(codec, WM8974_DAC, reg | div); + break; + case WM8974_BCLKDIV: + reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3; + snd_soc_write(codec, WM8974_CLOCK, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + u16 clk = snd_soc_read(codec, WM8974_CLOCK) & 0x1fe; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + clk |= 0x0001; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0010; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0008; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x00018; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0180; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0100; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0080; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, WM8974_IFACE, iface); + snd_soc_write(codec, WM8974_CLOCK, clk); + return 0; +} + +static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u16 iface = snd_soc_read(codec, WM8974_IFACE) & 0x19f; + u16 adn = snd_soc_read(codec, WM8974_ADD) & 0x1f1; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0020; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0040; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x0060; + break; + } + + /* filter coefficient */ + switch (params_rate(params)) { + case SNDRV_PCM_RATE_8000: + adn |= 0x5 << 1; + break; + case SNDRV_PCM_RATE_11025: + adn |= 0x4 << 1; + break; + case SNDRV_PCM_RATE_16000: + adn |= 0x3 << 1; + break; + case SNDRV_PCM_RATE_22050: + adn |= 0x2 << 1; + break; + case SNDRV_PCM_RATE_32000: + adn |= 0x1 << 1; + break; + case SNDRV_PCM_RATE_44100: + case SNDRV_PCM_RATE_48000: + break; + } + + snd_soc_write(codec, WM8974_IFACE, iface); + snd_soc_write(codec, WM8974_ADD, adn); + return 0; +} + +static int wm8974_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + + if (mute) + snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + else + snd_soc_write(codec, WM8974_DAC, mute_reg); + return 0; +} + +/* liam need to make this lower power with dapm */ +static int wm8974_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 power1 = snd_soc_read(codec, WM8974_POWER1) & ~0x3; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + power1 |= 0x1; /* VMID 50k */ + snd_soc_write(codec, WM8974_POWER1, power1); + break; + + case SND_SOC_BIAS_STANDBY: + power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN; + + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Initial cap charge at VMID 5k */ + snd_soc_write(codec, WM8974_POWER1, power1 | 0x3); + mdelay(100); + } + + power1 |= 0x2; /* VMID 500k */ + snd_soc_write(codec, WM8974_POWER1, power1); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, WM8974_POWER1, 0); + snd_soc_write(codec, WM8974_POWER2, 0); + snd_soc_write(codec, WM8974_POWER3, 0); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8974_RATES (SNDRV_PCM_RATE_8000_48000) + +#define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8974_ops = { + .hw_params = wm8974_pcm_hw_params, + .digital_mute = wm8974_mute, + .set_fmt = wm8974_set_dai_fmt, + .set_clkdiv = wm8974_set_dai_clkdiv, + .set_pll = wm8974_set_dai_pll, +}; + +struct snd_soc_dai wm8974_dai = { + .name = "WM8974 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = WM8974_RATES, + .formats = WM8974_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, /* Only 1 channel of data */ + .rates = WM8974_RATES, + .formats = WM8974_FORMATS,}, + .ops = &wm8974_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8974_dai); + +static int wm8974_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8974_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8974_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8974_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static int wm8974_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8974_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8974_codec; + codec = wm8974_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8974_snd_controls, + ARRAY_SIZE(wm8974_snd_controls)); + wm8974_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8974_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8974 = { + .probe = wm8974_probe, + .remove = wm8974_remove, + .suspend = wm8974_suspend, + .resume = wm8974_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8974); + +static __devinit int wm8974_register(struct wm8974_priv *wm8974) +{ + int ret; + struct snd_soc_codec *codec = &wm8974->codec; + + if (wm8974_codec) { + dev_err(codec->dev, "Another WM8974 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8974; + codec->name = "WM8974"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8974_set_bias_level; + codec->dai = &wm8974_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8974_CACHEREGNUM; + codec->reg_cache = &wm8974->reg_cache; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + memcpy(codec->reg_cache, wm8974_reg, sizeof(wm8974_reg)); + + ret = wm8974_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8974_dai.dev = codec->dev; + + wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8974_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8974_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8974); + return ret; +} + +static __devexit void wm8974_unregister(struct wm8974_priv *wm8974) +{ + wm8974_set_bias_level(&wm8974->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8974_dai); + snd_soc_unregister_codec(&wm8974->codec); + kfree(wm8974); + wm8974_codec = NULL; +} + +static __devinit int wm8974_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8974_priv *wm8974; + struct snd_soc_codec *codec; + + wm8974 = kzalloc(sizeof(struct wm8974_priv), GFP_KERNEL); + if (wm8974 == NULL) + return -ENOMEM; + + codec = &wm8974->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8974); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8974_register(wm8974); +} + +static __devexit int wm8974_i2c_remove(struct i2c_client *client) +{ + struct wm8974_priv *wm8974 = i2c_get_clientdata(client); + wm8974_unregister(wm8974); + return 0; +} + +static const struct i2c_device_id wm8974_i2c_id[] = { + { "wm8974", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id); + +static struct i2c_driver wm8974_i2c_driver = { + .driver = { + .name = "WM8974", + .owner = THIS_MODULE, + }, + .probe = wm8974_i2c_probe, + .remove = __devexit_p(wm8974_i2c_remove), + .id_table = wm8974_i2c_id, +}; + +static int __init wm8974_modinit(void) +{ + return i2c_add_driver(&wm8974_i2c_driver); +} +module_init(wm8974_modinit); + +static void __exit wm8974_exit(void) +{ + i2c_del_driver(&wm8974_i2c_driver); +} +module_exit(wm8974_exit); + +MODULE_DESCRIPTION("ASoC WM8974 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8974.h b/sound/soc/codecs/wm8974.h new file mode 100644 index 000000000000..98de9562d4d2 --- /dev/null +++ b/sound/soc/codecs/wm8974.h @@ -0,0 +1,99 @@ +/* + * wm8974.h -- WM8974 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8974_H +#define _WM8974_H + +/* WM8974 register space */ + +#define WM8974_RESET 0x0 +#define WM8974_POWER1 0x1 +#define WM8974_POWER2 0x2 +#define WM8974_POWER3 0x3 +#define WM8974_IFACE 0x4 +#define WM8974_COMP 0x5 +#define WM8974_CLOCK 0x6 +#define WM8974_ADD 0x7 +#define WM8974_GPIO 0x8 +#define WM8974_DAC 0xa +#define WM8974_DACVOL 0xb +#define WM8974_ADC 0xe +#define WM8974_ADCVOL 0xf +#define WM8974_EQ1 0x12 +#define WM8974_EQ2 0x13 +#define WM8974_EQ3 0x14 +#define WM8974_EQ4 0x15 +#define WM8974_EQ5 0x16 +#define WM8974_DACLIM1 0x18 +#define WM8974_DACLIM2 0x19 +#define WM8974_NOTCH1 0x1b +#define WM8974_NOTCH2 0x1c +#define WM8974_NOTCH3 0x1d +#define WM8974_NOTCH4 0x1e +#define WM8974_ALC1 0x20 +#define WM8974_ALC2 0x21 +#define WM8974_ALC3 0x22 +#define WM8974_NGATE 0x23 +#define WM8974_PLLN 0x24 +#define WM8974_PLLK1 0x25 +#define WM8974_PLLK2 0x26 +#define WM8974_PLLK3 0x27 +#define WM8974_ATTEN 0x28 +#define WM8974_INPUT 0x2c +#define WM8974_INPPGA 0x2d +#define WM8974_ADCBOOST 0x2f +#define WM8974_OUTPUT 0x31 +#define WM8974_SPKMIX 0x32 +#define WM8974_SPKVOL 0x36 +#define WM8974_MONOMIX 0x38 + +#define WM8974_CACHEREGNUM 57 + +/* Clock divider Id's */ +#define WM8974_OPCLKDIV 0 +#define WM8974_MCLKDIV 1 +#define WM8974_ADCCLK 2 +#define WM8974_DACCLK 3 +#define WM8974_BCLKDIV 4 + +/* DAC clock dividers */ +#define WM8974_DACCLK_F2 (1 << 3) +#define WM8974_DACCLK_F4 (0 << 3) + +/* ADC clock dividers */ +#define WM8974_ADCCLK_F2 (1 << 3) +#define WM8974_ADCCLK_F4 (0 << 3) + +/* PLL Out dividers */ +#define WM8974_OPCLKDIV_1 (0 << 4) +#define WM8974_OPCLKDIV_2 (1 << 4) +#define WM8974_OPCLKDIV_3 (2 << 4) +#define WM8974_OPCLKDIV_4 (3 << 4) + +/* BCLK clock dividers */ +#define WM8974_BCLKDIV_1 (0 << 2) +#define WM8974_BCLKDIV_2 (1 << 2) +#define WM8974_BCLKDIV_4 (2 << 2) +#define WM8974_BCLKDIV_8 (3 << 2) +#define WM8974_BCLKDIV_16 (4 << 2) +#define WM8974_BCLKDIV_32 (5 << 2) + +/* MCLK clock dividers */ +#define WM8974_MCLKDIV_1 (0 << 5) +#define WM8974_MCLKDIV_1_5 (1 << 5) +#define WM8974_MCLKDIV_2 (2 << 5) +#define WM8974_MCLKDIV_3 (3 << 5) +#define WM8974_MCLKDIV_4 (4 << 5) +#define WM8974_MCLKDIV_6 (5 << 5) +#define WM8974_MCLKDIV_8 (6 << 5) +#define WM8974_MCLKDIV_12 (7 << 5) + +extern struct snd_soc_dai wm8974_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8974; + +#endif diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 8c0fdf84aac3..3f530f8a972a 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -57,50 +57,7 @@ struct wm8988_priv { }; -/* - * read wm8988 register cache - */ -static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8988_NUM_REG) - return -1; - return cache[reg]; -} - -/* - * write wm8988 register cache - */ -static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - if (reg > WM8988_NUM_REG) - return; - cache[reg] = value; -} - -static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - /* data is - * D15..D9 WM8753 register offset - * D8...D0 register data - */ - data[0] = (reg << 1) | ((value >> 8) & 0x0001); - data[1] = value & 0x00ff; - - wm8988_write_reg_cache(codec, reg, value); - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} - -#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0) +#define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) /* * WM8988 Controls @@ -226,15 +183,15 @@ static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2); + u16 adctl2 = snd_soc_read(codec, WM8988_ADCTL2); /* Use the DAC to gate LRC if active, otherwise use ADC */ - if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180) + if (snd_soc_read(codec, WM8988_PWR2) & 0x180) adctl2 &= ~0x4; else adctl2 |= 0x4; - return wm8988_write(codec, WM8988_ADCTL2, adctl2); + return snd_soc_write(codec, WM8988_ADCTL2, adctl2); } static const char *wm8988_line_texts[] = { @@ -619,7 +576,7 @@ static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8988_write(codec, WM8988_IFACE, iface); + snd_soc_write(codec, WM8988_IFACE, iface); return 0; } @@ -653,8 +610,8 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; struct wm8988_priv *wm8988 = codec->private_data; - u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3; - u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180; + u16 iface = snd_soc_read(codec, WM8988_IFACE) & 0x1f3; + u16 srate = snd_soc_read(codec, WM8988_SRATE) & 0x180; int coeff; coeff = get_coeff(wm8988->sysclk, params_rate(params)); @@ -685,9 +642,9 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, } /* set iface & srate */ - wm8988_write(codec, WM8988_IFACE, iface); + snd_soc_write(codec, WM8988_IFACE, iface); if (coeff >= 0) - wm8988_write(codec, WM8988_SRATE, srate | + snd_soc_write(codec, WM8988_SRATE, srate | (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); return 0; @@ -696,19 +653,19 @@ static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, static int wm8988_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7; + u16 mute_reg = snd_soc_read(codec, WM8988_ADCDAC) & 0xfff7; if (mute) - wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8); + snd_soc_write(codec, WM8988_ADCDAC, mute_reg | 0x8); else - wm8988_write(codec, WM8988_ADCDAC, mute_reg); + snd_soc_write(codec, WM8988_ADCDAC, mute_reg); return 0; } static int wm8988_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1; + u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1; switch (level) { case SND_SOC_BIAS_ON: @@ -716,24 +673,24 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VREF, VMID=2x50k, digital enabled */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* VREF, VMID=2x5k */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); /* Charge caps */ msleep(100); } /* VREF, VMID=2*500k, digital stopped */ - wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141); + snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x0141); break; case SND_SOC_BIAS_OFF: - wm8988_write(codec, WM8988_PWR1, 0x0000); + snd_soc_write(codec, WM8988_PWR1, 0x0000); break; } codec->bias_level = level; @@ -868,7 +825,8 @@ struct snd_soc_codec_device soc_codec_dev_wm8988 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988); -static int wm8988_register(struct wm8988_priv *wm8988) +static int wm8988_register(struct wm8988_priv *wm8988, + enum snd_soc_control_type control) { struct snd_soc_codec *codec = &wm8988->codec; int ret; @@ -887,8 +845,6 @@ static int wm8988_register(struct wm8988_priv *wm8988) codec->private_data = wm8988; codec->name = "WM8988"; codec->owner = THIS_MODULE; - codec->read = wm8988_read_reg_cache; - codec->write = wm8988_write; codec->dai = &wm8988_dai; codec->num_dai = 1; codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache); @@ -899,23 +855,29 @@ static int wm8988_register(struct wm8988_priv *wm8988) memcpy(codec->reg_cache, wm8988_reg, sizeof(wm8988_reg)); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + ret = wm8988_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); - return ret; + goto err; } /* set the update bits (we always update left then right) */ - reg = wm8988_read_reg_cache(codec, WM8988_RADC); - wm8988_write(codec, WM8988_RADC, reg | 0x100); - reg = wm8988_read_reg_cache(codec, WM8988_RDAC); - wm8988_write(codec, WM8988_RDAC, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V); - wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V); - wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100); - reg = wm8988_read_reg_cache(codec, WM8988_RINVOL); - wm8988_write(codec, WM8988_RINVOL, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_RADC); + snd_soc_write(codec, WM8988_RADC, reg | 0x100); + reg = snd_soc_read(codec, WM8988_RDAC); + snd_soc_write(codec, WM8988_RDAC, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_ROUT1V); + snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_ROUT2V); + snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100); + reg = snd_soc_read(codec, WM8988_RINVOL); + snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100); wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY); @@ -926,18 +888,20 @@ static int wm8988_register(struct wm8988_priv *wm8988) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - return ret; + goto err; } ret = snd_soc_register_dai(&wm8988_dai); if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); - return ret; + goto err_codec; } return 0; +err_codec: + snd_soc_unregister_codec(codec); err: kfree(wm8988); return ret; @@ -964,14 +928,13 @@ static int wm8988_i2c_probe(struct i2c_client *i2c, return -ENOMEM; codec = &wm8988->codec; - codec->hw_write = (hw_write_t)i2c_master_send; i2c_set_clientdata(i2c, wm8988); codec->control_data = i2c; codec->dev = &i2c->dev; - return wm8988_register(wm8988); + return wm8988_register(wm8988, SND_SOC_I2C); } static int wm8988_i2c_remove(struct i2c_client *client) @@ -981,6 +944,21 @@ static int wm8988_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm8988_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm8988_i2c_suspend NULL +#define wm8988_i2c_resume NULL +#endif + static const struct i2c_device_id wm8988_i2c_id[] = { { "wm8988", 0 }, { } @@ -994,35 +972,13 @@ static struct i2c_driver wm8988_i2c_driver = { }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, + .suspend = wm8988_i2c_suspend, + .resume = wm8988_i2c_resume, .id_table = wm8988_i2c_id, }; #endif #if defined(CONFIG_SPI_MASTER) -static int wm8988_spi_write(struct spi_device *spi, const char *data, int len) -{ - struct spi_transfer t; - struct spi_message m; - u8 msg[2]; - - if (len <= 0) - return 0; - - msg[0] = data[0]; - msg[1] = data[1]; - - spi_message_init(&m); - memset(&t, 0, (sizeof t)); - - t.tx_buf = &msg[0]; - t.len = len; - - spi_message_add_tail(&t, &m); - spi_sync(spi, &m); - - return len; -} - static int __devinit wm8988_spi_probe(struct spi_device *spi) { struct wm8988_priv *wm8988; @@ -1033,13 +989,12 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) return -ENOMEM; codec = &wm8988->codec; - codec->hw_write = (hw_write_t)wm8988_spi_write; codec->control_data = spi; codec->dev = &spi->dev; dev_set_drvdata(&spi->dev, wm8988); - return wm8988_register(wm8988); + return wm8988_register(wm8988, SND_SOC_SPI); } static int __devexit wm8988_spi_remove(struct spi_device *spi) @@ -1051,6 +1006,21 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) return 0; } +#ifdef CONFIG_PM +static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8988_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8988_spi_suspend NULL +#define wm8988_spi_resume NULL +#endif + static struct spi_driver wm8988_spi_driver = { .driver = { .name = "wm8988", @@ -1059,6 +1029,8 @@ static struct spi_driver wm8988_spi_driver = { }, .probe = wm8988_spi_probe, .remove = __devexit_p(wm8988_spi_remove), + .suspend = wm8988_spi_suspend, + .resume = wm8988_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index d029818350e9..2d702db4131d 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -108,53 +108,7 @@ static const u16 wm8990_reg[] = { 0x0000, /* R63 - Driver internal */ }; -/* - * read wm8990 register cache - */ -static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg >= ARRAY_SIZE(wm8990_reg)); - return cache[reg]; -} - -/* - * write wm8990 register cache - */ -static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - u16 *cache = codec->reg_cache; - - /* Reset register and reserved registers are uncached */ - if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg)) - return; - - cache[reg] = value; -} - -/* - * write to the wm8990 register space - */ -static int wm8990_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - - data[0] = reg & 0xFF; - data[1] = (value >> 8) & 0xFF; - data[2] = value & 0xFF; - - wm8990_write_reg_cache(codec, reg, value); - - if (codec->hw_write(codec->control_data, data, 3) == 2) - return 0; - else - return -EIO; -} - -#define wm8990_reset(c) wm8990_write(c, WM8990_RESET, 0) +#define wm8990_reset(c) snd_soc_write(c, WM8990_RESET, 0) static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); @@ -187,8 +141,8 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, return ret; /* now hit the volume update bits (always bit 8) */ - val = wm8990_read_reg_cache(codec, reg); - return wm8990_write(codec, reg, val | 0x0100); + val = snd_soc_read(codec, reg); + return snd_soc_write(codec, reg, val | 0x0100); } #define SOC_WM899X_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert,\ @@ -427,8 +381,8 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, { u16 reg, fakepower; - reg = wm8990_read_reg_cache(w->codec, WM8990_POWER_MANAGEMENT_2); - fakepower = wm8990_read_reg_cache(w->codec, WM8990_INTDRIVBITS); + reg = snd_soc_read(w->codec, WM8990_POWER_MANAGEMENT_2); + fakepower = snd_soc_read(w->codec, WM8990_INTDRIVBITS); if (fakepower & ((1 << WM8990_INMIXL_PWR_BIT) | (1 << WM8990_AINLMUX_PWR_BIT))) { @@ -443,7 +397,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w, } else { reg &= ~WM8990_AINL_ENA; } - wm8990_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg); return 0; } @@ -457,7 +411,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, switch (reg_shift) { case WM8990_SPEAKER_MIXER | (WM8990_LDSPK_BIT << 8) : - reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER1); + reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER1); if (reg & WM8990_LDLO) { printk(KERN_WARNING "Cannot set as Output Mixer 1 LDLO Set\n"); @@ -465,7 +419,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, } break; case WM8990_SPEAKER_MIXER | (WM8990_RDSPK_BIT << 8): - reg = wm8990_read_reg_cache(w->codec, WM8990_OUTPUT_MIXER2); + reg = snd_soc_read(w->codec, WM8990_OUTPUT_MIXER2); if (reg & WM8990_RDRO) { printk(KERN_WARNING "Cannot set as Output Mixer 2 RDRO Set\n"); @@ -473,7 +427,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, } break; case WM8990_OUTPUT_MIXER1 | (WM8990_LDLO_BIT << 8): - reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER); if (reg & WM8990_LDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer LDSPK Set\n"); @@ -481,7 +435,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, } break; case WM8990_OUTPUT_MIXER2 | (WM8990_RDRO_BIT << 8): - reg = wm8990_read_reg_cache(w->codec, WM8990_SPEAKER_MIXER); + reg = snd_soc_read(w->codec, WM8990_SPEAKER_MIXER); if (reg & WM8990_RDSPK) { printk(KERN_WARNING "Cannot set as Speaker Mixer RDSPK Set\n"); @@ -1029,24 +983,24 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, pll_factors(&pll_div, freq_out * 4, freq_in); /* Turn on PLL */ - reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); reg |= WM8990_PLL_ENA; - wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); /* sysclk comes from PLL */ - reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2); - wm8990_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); + reg = snd_soc_read(codec, WM8990_CLOCKING_2); + snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC); /* set up N , fractional mode and pre-divisor if neccessary */ - wm8990_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | + snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM | (pll_div.div2?WM8990_PRESCALE:0)); - wm8990_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); - wm8990_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); + snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8)); + snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF)); } else { /* Turn on PLL */ - reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); + reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); reg &= ~WM8990_PLL_ENA; - wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg); } return 0; } @@ -1073,8 +1027,8 @@ static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; u16 audio1, audio3; - audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); - audio3 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_3); + audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); + audio3 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_3); /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1115,8 +1069,8 @@ static int wm8990_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); - wm8990_write(codec, WM8990_AUDIO_INTERFACE_3, audio3); + snd_soc_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8990_AUDIO_INTERFACE_3, audio3); return 0; } @@ -1128,24 +1082,24 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div_id) { case WM8990_MCLK_DIV: - reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + reg = snd_soc_read(codec, WM8990_CLOCKING_2) & ~WM8990_MCLK_DIV_MASK; - wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); break; case WM8990_DACCLK_DIV: - reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + reg = snd_soc_read(codec, WM8990_CLOCKING_2) & ~WM8990_DAC_CLKDIV_MASK; - wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); break; case WM8990_ADCCLK_DIV: - reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_2) & + reg = snd_soc_read(codec, WM8990_CLOCKING_2) & ~WM8990_ADC_CLKDIV_MASK; - wm8990_write(codec, WM8990_CLOCKING_2, reg | div); + snd_soc_write(codec, WM8990_CLOCKING_2, reg | div); break; case WM8990_BCLK_DIV: - reg = wm8990_read_reg_cache(codec, WM8990_CLOCKING_1) & + reg = snd_soc_read(codec, WM8990_CLOCKING_1) & ~WM8990_BCLK_DIV_MASK; - wm8990_write(codec, WM8990_CLOCKING_1, reg | div); + snd_soc_write(codec, WM8990_CLOCKING_1, reg | div); break; default: return -EINVAL; @@ -1164,7 +1118,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); + u16 audio1 = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; /* bit size */ @@ -1182,7 +1136,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream, break; } - wm8990_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); + snd_soc_write(codec, WM8990_AUDIO_INTERFACE_1, audio1); return 0; } @@ -1191,12 +1145,12 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_codec *codec = dai->codec; u16 val; - val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE; + val = snd_soc_read(codec, WM8990_DAC_CTRL) & ~WM8990_DAC_MUTE; if (mute) - wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); else - wm8990_write(codec, WM8990_DAC_CTRL, val); + snd_soc_write(codec, WM8990_DAC_CTRL, val); return 0; } @@ -1212,21 +1166,21 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*50k */ - val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & ~WM8990_VMID_MODE_MASK; - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); break; case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable all output discharge bits */ - wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | WM8990_DIS_RLINE | WM8990_DIS_OUT3 | WM8990_DIS_OUT4 | WM8990_DIS_LOUT | WM8990_DIS_ROUT); /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ - wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | WM8990_BUFDCOPEN | WM8990_POBCTRL | WM8990_VMIDTOG); @@ -1234,83 +1188,83 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, msleep(msecs_to_jiffies(300)); /* Disable VMIDTOG */ - wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | WM8990_BUFDCOPEN | WM8990_POBCTRL); /* disable all output discharge bits */ - wm8990_write(codec, WM8990_ANTIPOP1, 0); + snd_soc_write(codec, WM8990_ANTIPOP1, 0); /* Enable outputs */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1b00); msleep(msecs_to_jiffies(50)); /* Enable VMID at 2x50k */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f02); msleep(msecs_to_jiffies(100)); /* Enable VREF */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); msleep(msecs_to_jiffies(600)); /* Enable BUFIOEN */ - wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | WM8990_BUFDCOPEN | WM8990_POBCTRL | WM8990_BUFIOEN); /* Disable outputs */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x3); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); + snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); /* Enable workaround for ADC clocking issue. */ - wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2); - wm8990_write(codec, WM8990_EXT_CTL1, 0xa003); - wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0); + snd_soc_write(codec, WM8990_EXT_ACCESS_ENA, 0x2); + snd_soc_write(codec, WM8990_EXT_CTL1, 0xa003); + snd_soc_write(codec, WM8990_EXT_ACCESS_ENA, 0); } /* VMID=2*250k */ - val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) & ~WM8990_VMID_MODE_MASK; - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: /* Enable POBCTRL and SOFT_ST */ - wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | WM8990_POBCTRL | WM8990_BUFIOEN); /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8990_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | + snd_soc_write(codec, WM8990_ANTIPOP2, WM8990_SOFTST | WM8990_BUFDCOPEN | WM8990_POBCTRL | WM8990_BUFIOEN); /* mute DAC */ - val = wm8990_read_reg_cache(codec, WM8990_DAC_CTRL); - wm8990_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); + val = snd_soc_read(codec, WM8990_DAC_CTRL); + snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE); /* Enable any disabled outputs */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03); /* Disable VMID */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f01); msleep(msecs_to_jiffies(300)); /* Enable all output discharge bits */ - wm8990_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | + snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE | WM8990_DIS_RLINE | WM8990_DIS_OUT3 | WM8990_DIS_OUT4 | WM8990_DIS_LOUT | WM8990_DIS_ROUT); /* Disable VREF */ - wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x0); /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ - wm8990_write(codec, WM8990_ANTIPOP2, 0x0); + snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); break; } @@ -1411,8 +1365,6 @@ static int wm8990_init(struct snd_soc_device *socdev) codec->name = "WM8990"; codec->owner = THIS_MODULE; - codec->read = wm8990_read_reg_cache; - codec->write = wm8990_write; codec->set_bias_level = wm8990_set_bias_level; codec->dai = &wm8990_dai; codec->num_dai = 2; @@ -1422,6 +1374,12 @@ static int wm8990_init(struct snd_soc_device *socdev) if (codec->reg_cache == NULL) return -ENOMEM; + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + if (ret < 0) { + printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); + goto pcm_err; + } + wm8990_reset(codec); /* register pcms */ @@ -1435,18 +1393,18 @@ static int wm8990_init(struct snd_soc_device *socdev) codec->bias_level = SND_SOC_BIAS_OFF; wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - reg = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_4); - wm8990_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); + reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4); + snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1); - reg = wm8990_read_reg_cache(codec, WM8990_GPIO1_GPIO2) & + reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) & ~WM8990_GPIO1_SEL_MASK; - wm8990_write(codec, WM8990_GPIO1_GPIO2, reg | 1); + snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1); - reg = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_2); - wm8990_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); + reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2); + snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA); - wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_add_controls(codec, wm8990_snd_controls, ARRAY_SIZE(wm8990_snd_controls)); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c new file mode 100644 index 000000000000..d9987999e92c --- /dev/null +++ b/sound/soc/codecs/wm8993.c @@ -0,0 +1,1675 @@ +/* + * wm8993.c -- WM8993 ALSA SoC audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/wm8993.h> + +#include "wm8993.h" +#include "wm_hubs.h" + +static u16 wm8993_reg_defaults[WM8993_REGISTER_COUNT] = { + 0x8993, /* R0 - Software Reset */ + 0x0000, /* R1 - Power Management (1) */ + 0x6000, /* R2 - Power Management (2) */ + 0x0000, /* R3 - Power Management (3) */ + 0x4050, /* R4 - Audio Interface (1) */ + 0x4000, /* R5 - Audio Interface (2) */ + 0x01C8, /* R6 - Clocking 1 */ + 0x0000, /* R7 - Clocking 2 */ + 0x0000, /* R8 - Audio Interface (3) */ + 0x0040, /* R9 - Audio Interface (4) */ + 0x0004, /* R10 - DAC CTRL */ + 0x00C0, /* R11 - Left DAC Digital Volume */ + 0x00C0, /* R12 - Right DAC Digital Volume */ + 0x0000, /* R13 - Digital Side Tone */ + 0x0300, /* R14 - ADC CTRL */ + 0x00C0, /* R15 - Left ADC Digital Volume */ + 0x00C0, /* R16 - Right ADC Digital Volume */ + 0x0000, /* R17 */ + 0x0000, /* R18 - GPIO CTRL 1 */ + 0x0010, /* R19 - GPIO1 */ + 0x0000, /* R20 - IRQ_DEBOUNCE */ + 0x0000, /* R21 */ + 0x8000, /* R22 - GPIOCTRL 2 */ + 0x0800, /* R23 - GPIO_POL */ + 0x008B, /* R24 - Left Line Input 1&2 Volume */ + 0x008B, /* R25 - Left Line Input 3&4 Volume */ + 0x008B, /* R26 - Right Line Input 1&2 Volume */ + 0x008B, /* R27 - Right Line Input 3&4 Volume */ + 0x006D, /* R28 - Left Output Volume */ + 0x006D, /* R29 - Right Output Volume */ + 0x0066, /* R30 - Line Outputs Volume */ + 0x0020, /* R31 - HPOUT2 Volume */ + 0x0079, /* R32 - Left OPGA Volume */ + 0x0079, /* R33 - Right OPGA Volume */ + 0x0003, /* R34 - SPKMIXL Attenuation */ + 0x0003, /* R35 - SPKMIXR Attenuation */ + 0x0011, /* R36 - SPKOUT Mixers */ + 0x0100, /* R37 - SPKOUT Boost */ + 0x0079, /* R38 - Speaker Volume Left */ + 0x0079, /* R39 - Speaker Volume Right */ + 0x0000, /* R40 - Input Mixer2 */ + 0x0000, /* R41 - Input Mixer3 */ + 0x0000, /* R42 - Input Mixer4 */ + 0x0000, /* R43 - Input Mixer5 */ + 0x0000, /* R44 - Input Mixer6 */ + 0x0000, /* R45 - Output Mixer1 */ + 0x0000, /* R46 - Output Mixer2 */ + 0x0000, /* R47 - Output Mixer3 */ + 0x0000, /* R48 - Output Mixer4 */ + 0x0000, /* R49 - Output Mixer5 */ + 0x0000, /* R50 - Output Mixer6 */ + 0x0000, /* R51 - HPOUT2 Mixer */ + 0x0000, /* R52 - Line Mixer1 */ + 0x0000, /* R53 - Line Mixer2 */ + 0x0000, /* R54 - Speaker Mixer */ + 0x0000, /* R55 - Additional Control */ + 0x0000, /* R56 - AntiPOP1 */ + 0x0000, /* R57 - AntiPOP2 */ + 0x0000, /* R58 - MICBIAS */ + 0x0000, /* R59 */ + 0x0000, /* R60 - FLL Control 1 */ + 0x0000, /* R61 - FLL Control 2 */ + 0x0000, /* R62 - FLL Control 3 */ + 0x2EE0, /* R63 - FLL Control 4 */ + 0x0002, /* R64 - FLL Control 5 */ + 0x2287, /* R65 - Clocking 3 */ + 0x025F, /* R66 - Clocking 4 */ + 0x0000, /* R67 - MW Slave Control */ + 0x0000, /* R68 */ + 0x0002, /* R69 - Bus Control 1 */ + 0x0000, /* R70 - Write Sequencer 0 */ + 0x0000, /* R71 - Write Sequencer 1 */ + 0x0000, /* R72 - Write Sequencer 2 */ + 0x0000, /* R73 - Write Sequencer 3 */ + 0x0000, /* R74 - Write Sequencer 4 */ + 0x0000, /* R75 - Write Sequencer 5 */ + 0x1F25, /* R76 - Charge Pump 1 */ + 0x0000, /* R77 */ + 0x0000, /* R78 */ + 0x0000, /* R79 */ + 0x0000, /* R80 */ + 0x0000, /* R81 - Class W 0 */ + 0x0000, /* R82 */ + 0x0000, /* R83 */ + 0x0000, /* R84 - DC Servo 0 */ + 0x054A, /* R85 - DC Servo 1 */ + 0x0000, /* R86 */ + 0x0000, /* R87 - DC Servo 3 */ + 0x0000, /* R88 - DC Servo Readback 0 */ + 0x0000, /* R89 - DC Servo Readback 1 */ + 0x0000, /* R90 - DC Servo Readback 2 */ + 0x0000, /* R91 */ + 0x0000, /* R92 */ + 0x0000, /* R93 */ + 0x0000, /* R94 */ + 0x0000, /* R95 */ + 0x0100, /* R96 - Analogue HP 0 */ + 0x0000, /* R97 */ + 0x0000, /* R98 - EQ1 */ + 0x000C, /* R99 - EQ2 */ + 0x000C, /* R100 - EQ3 */ + 0x000C, /* R101 - EQ4 */ + 0x000C, /* R102 - EQ5 */ + 0x000C, /* R103 - EQ6 */ + 0x0FCA, /* R104 - EQ7 */ + 0x0400, /* R105 - EQ8 */ + 0x00D8, /* R106 - EQ9 */ + 0x1EB5, /* R107 - EQ10 */ + 0xF145, /* R108 - EQ11 */ + 0x0B75, /* R109 - EQ12 */ + 0x01C5, /* R110 - EQ13 */ + 0x1C58, /* R111 - EQ14 */ + 0xF373, /* R112 - EQ15 */ + 0x0A54, /* R113 - EQ16 */ + 0x0558, /* R114 - EQ17 */ + 0x168E, /* R115 - EQ18 */ + 0xF829, /* R116 - EQ19 */ + 0x07AD, /* R117 - EQ20 */ + 0x1103, /* R118 - EQ21 */ + 0x0564, /* R119 - EQ22 */ + 0x0559, /* R120 - EQ23 */ + 0x4000, /* R121 - EQ24 */ + 0x0000, /* R122 - Digital Pulls */ + 0x0F08, /* R123 - DRC Control 1 */ + 0x0000, /* R124 - DRC Control 2 */ + 0x0080, /* R125 - DRC Control 3 */ + 0x0000, /* R126 - DRC Control 4 */ +}; + +static struct { + int ratio; + int clk_sys_rate; +} clk_sys_rates[] = { + { 64, 0 }, + { 128, 1 }, + { 192, 2 }, + { 256, 3 }, + { 384, 4 }, + { 512, 5 }, + { 768, 6 }, + { 1024, 7 }, + { 1408, 8 }, + { 1536, 9 }, +}; + +static struct { + int rate; + int sample_rate; +} sample_rates[] = { + { 8000, 0 }, + { 11025, 1 }, + { 12000, 1 }, + { 16000, 2 }, + { 22050, 3 }, + { 24000, 3 }, + { 32000, 4 }, + { 44100, 5 }, + { 48000, 5 }, +}; + +static struct { + int div; /* *10 due to .5s */ + int bclk_div; +} bclk_divs[] = { + { 10, 0 }, + { 15, 1 }, + { 20, 2 }, + { 30, 3 }, + { 40, 4 }, + { 55, 5 }, + { 60, 6 }, + { 80, 7 }, + { 110, 8 }, + { 120, 9 }, + { 160, 10 }, + { 220, 11 }, + { 240, 12 }, + { 320, 13 }, + { 440, 14 }, + { 480, 15 }, +}; + +struct wm8993_priv { + u16 reg_cache[WM8993_REGISTER_COUNT]; + struct wm8993_platform_data pdata; + struct snd_soc_codec codec; + int master; + int sysclk_source; + int tdm_slots; + int tdm_width; + unsigned int mclk_rate; + unsigned int sysclk_rate; + unsigned int fs; + unsigned int bclk; + int class_w_users; + unsigned int fll_fref; + unsigned int fll_fout; +}; + +static unsigned int wm8993_read_hw(struct snd_soc_codec *codec, u8 reg) +{ + struct i2c_msg xfer[2]; + u16 data; + int ret; + struct i2c_client *i2c = codec->control_data; + + /* Write register */ + xfer[0].addr = i2c->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = i2c->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(i2c->adapter, xfer, 2); + if (ret != 2) { + dev_err(codec->dev, "Failed to read 0x%x: %d\n", reg, ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} + +static int wm8993_volatile(unsigned int reg) +{ + switch (reg) { + case WM8993_SOFTWARE_RESET: + case WM8993_DC_SERVO_0: + case WM8993_DC_SERVO_READBACK_0: + case WM8993_DC_SERVO_READBACK_1: + case WM8993_DC_SERVO_READBACK_2: + return 1; + default: + return 0; + } +} + +static unsigned int wm8993_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + BUG_ON(reg > WM8993_MAX_REGISTER); + + if (wm8993_volatile(reg)) + return wm8993_read_hw(codec, reg); + else + return reg_cache[reg]; +} + +static int wm8993_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *reg_cache = codec->reg_cache; + u8 data[3]; + int ret; + + BUG_ON(reg > WM8993_MAX_REGISTER); + + /* data is + * D15..D9 WM8993 register offset + * D8...D0 register data + */ + data[0] = reg; + data[1] = value >> 8; + data[2] = value & 0x00ff; + + if (!wm8993_volatile(reg)) + reg_cache[reg] = value; + + ret = codec->hw_write(codec->control_data, data, 3); + + if (ret == 3) + return 0; + if (ret < 0) + return ret; + return -EIO; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_clk_ref_div; + u16 n; + u16 k; +}; + +/* The size in bits of the FLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + unsigned int div; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_clk_ref_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_clk_ref_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 0; + target = Fout * 2; + while (target < 90000000) { + div++; + target *= 2; + if (div > 7) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + fll_div->fll_outdiv = div; + + pr_debug("Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + target /= fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + /* Now, calculate N.K */ + Ndiv = target / Fref; + + fll_div->n = Ndiv; + Nmod = target % Fref; + pr_debug("Nmod=%d\n", Nmod); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, Fref); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + fll_div->k = K / 10; + + pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n", + fll_div->n, fll_div->k, + fll_div->fll_fratio, fll_div->fll_outdiv, + fll_div->fll_clk_ref_div); + + return 0; +} + +static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, + unsigned int Fref, unsigned int Fout) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8993_priv *wm8993 = codec->private_data; + u16 reg1, reg4, reg5; + struct _fll_div fll_div; + int ret; + + /* Any change? */ + if (Fref == wm8993->fll_fref && Fout == wm8993->fll_fout) + return 0; + + /* Disable the FLL */ + if (Fout == 0) { + dev_dbg(codec->dev, "FLL disabled\n"); + wm8993->fll_fref = 0; + wm8993->fll_fout = 0; + + reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1); + reg1 &= ~WM8993_FLL_ENA; + wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + + return 0; + } + + ret = fll_factors(&fll_div, Fref, Fout); + if (ret != 0) + return ret; + + reg5 = wm8993_read(codec, WM8993_FLL_CONTROL_5); + reg5 &= ~WM8993_FLL_CLK_SRC_MASK; + + switch (fll_id) { + case WM8993_FLL_MCLK: + break; + + case WM8993_FLL_LRCLK: + reg5 |= 1; + break; + + case WM8993_FLL_BCLK: + reg5 |= 2; + break; + + default: + dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id); + return -EINVAL; + } + + /* Any FLL configuration change requires that the FLL be + * disabled first. */ + reg1 = wm8993_read(codec, WM8993_FLL_CONTROL_1); + reg1 &= ~WM8993_FLL_ENA; + wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + + /* Apply the configuration */ + if (fll_div.k) + reg1 |= WM8993_FLL_FRAC_MASK; + else + reg1 &= ~WM8993_FLL_FRAC_MASK; + wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1); + + wm8993_write(codec, WM8993_FLL_CONTROL_2, + (fll_div.fll_outdiv << WM8993_FLL_OUTDIV_SHIFT) | + (fll_div.fll_fratio << WM8993_FLL_FRATIO_SHIFT)); + wm8993_write(codec, WM8993_FLL_CONTROL_3, fll_div.k); + + reg4 = wm8993_read(codec, WM8993_FLL_CONTROL_4); + reg4 &= ~WM8993_FLL_N_MASK; + reg4 |= fll_div.n << WM8993_FLL_N_SHIFT; + wm8993_write(codec, WM8993_FLL_CONTROL_4, reg4); + + reg5 &= ~WM8993_FLL_CLK_REF_DIV_MASK; + reg5 |= fll_div.fll_clk_ref_div << WM8993_FLL_CLK_REF_DIV_SHIFT; + wm8993_write(codec, WM8993_FLL_CONTROL_5, reg5); + + /* Enable the FLL */ + wm8993_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); + + dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); + + wm8993->fll_fref = Fref; + wm8993->fll_fout = Fout; + + return 0; +} + +static int configure_clock(struct snd_soc_codec *codec) +{ + struct wm8993_priv *wm8993 = codec->private_data; + unsigned int reg; + + /* This should be done on init() for bypass paths */ + switch (wm8993->sysclk_source) { + case WM8993_SYSCLK_MCLK: + dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8993->mclk_rate); + + reg = wm8993_read(codec, WM8993_CLOCKING_2); + reg &= ~(WM8993_MCLK_DIV | WM8993_SYSCLK_SRC); + if (wm8993->mclk_rate > 13500000) { + reg |= WM8993_MCLK_DIV; + wm8993->sysclk_rate = wm8993->mclk_rate / 2; + } else { + reg &= ~WM8993_MCLK_DIV; + wm8993->sysclk_rate = wm8993->mclk_rate; + } + wm8993_write(codec, WM8993_CLOCKING_2, reg); + break; + + case WM8993_SYSCLK_FLL: + dev_dbg(codec->dev, "Using %dHz FLL clock\n", + wm8993->fll_fout); + + reg = wm8993_read(codec, WM8993_CLOCKING_2); + reg |= WM8993_SYSCLK_SRC; + if (wm8993->fll_fout > 13500000) { + reg |= WM8993_MCLK_DIV; + wm8993->sysclk_rate = wm8993->fll_fout / 2; + } else { + reg &= ~WM8993_MCLK_DIV; + wm8993->sysclk_rate = wm8993->fll_fout; + } + wm8993_write(codec, WM8993_CLOCKING_2, reg); + break; + + default: + dev_err(codec->dev, "System clock not configured\n"); + return -EINVAL; + } + + dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm8993->sysclk_rate); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0); +static const DECLARE_TLV_DB_SCALE(drc_comp_threash, -4500, 75, 0); +static const DECLARE_TLV_DB_SCALE(drc_comp_amp, -2250, 75, 0); +static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0); +static const unsigned int drc_max_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0), + 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0); +static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -1800, 300, 0); +static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0); + +static const char *dac_deemph_text[] = { + "None", + "32kHz", + "44.1kHz", + "48kHz", +}; + +static const struct soc_enum dac_deemph = + SOC_ENUM_SINGLE(WM8993_DAC_CTRL, 4, 4, dac_deemph_text); + +static const char *adc_hpf_text[] = { + "Hi-Fi", + "Voice 1", + "Voice 2", + "Voice 3", +}; + +static const struct soc_enum adc_hpf = + SOC_ENUM_SINGLE(WM8993_ADC_CTRL, 5, 4, adc_hpf_text); + +static const char *drc_path_text[] = { + "ADC", + "DAC" +}; + +static const struct soc_enum drc_path = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 14, 2, drc_path_text); + +static const char *drc_r0_text[] = { + "1", + "1/2", + "1/4", + "1/8", + "1/16", + "0", +}; + +static const struct soc_enum drc_r0 = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 8, 6, drc_r0_text); + +static const char *drc_r1_text[] = { + "1", + "1/2", + "1/4", + "1/8", + "0", +}; + +static const struct soc_enum drc_r1 = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_4, 13, 5, drc_r1_text); + +static const char *drc_attack_text[] = { + "Reserved", + "181us", + "363us", + "726us", + "1.45ms", + "2.9ms", + "5.8ms", + "11.6ms", + "23.2ms", + "46.4ms", + "92.8ms", + "185.6ms", +}; + +static const struct soc_enum drc_attack = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 12, 12, drc_attack_text); + +static const char *drc_decay_text[] = { + "186ms", + "372ms", + "743ms", + "1.49s", + "2.97ms", + "5.94ms", + "11.89ms", + "23.78ms", + "47.56ms", +}; + +static const struct soc_enum drc_decay = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 8, 9, drc_decay_text); + +static const char *drc_ff_text[] = { + "5 samples", + "9 samples", +}; + +static const struct soc_enum drc_ff = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 7, 2, drc_ff_text); + +static const char *drc_qr_rate_text[] = { + "0.725ms", + "1.45ms", + "5.8ms", +}; + +static const struct soc_enum drc_qr_rate = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 0, 3, drc_qr_rate_text); + +static const char *drc_smooth_text[] = { + "Low", + "Medium", + "High", +}; + +static const struct soc_enum drc_smooth = + SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 4, 3, drc_smooth_text); + +static const struct snd_kcontrol_new wm8993_snd_controls[] = { +SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE, + 5, 9, 12, 0, sidetone_tlv), + +SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0), +SOC_ENUM("DRC Path", drc_path), +SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2, + 2, 60, 1, drc_comp_threash), +SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3, + 11, 30, 1, drc_comp_amp), +SOC_ENUM("DRC R0", drc_r0), +SOC_ENUM("DRC R1", drc_r1), +SOC_SINGLE_TLV("DRC Minimum Volume", WM8993_DRC_CONTROL_1, 2, 3, 1, + drc_min_tlv), +SOC_SINGLE_TLV("DRC Maximum Volume", WM8993_DRC_CONTROL_1, 0, 3, 0, + drc_max_tlv), +SOC_ENUM("DRC Attack Rate", drc_attack), +SOC_ENUM("DRC Decay Rate", drc_decay), +SOC_ENUM("DRC FF Delay", drc_ff), +SOC_SINGLE("DRC Anti-clip Switch", WM8993_DRC_CONTROL_1, 9, 1, 0), +SOC_SINGLE("DRC Quick Release Switch", WM8993_DRC_CONTROL_1, 10, 1, 0), +SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0, + drc_qr_tlv), +SOC_ENUM("DRC Quick Release Rate", drc_qr_rate), +SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0), +SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0), +SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth), +SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0, + drc_startup_tlv), + +SOC_SINGLE("EQ Switch", WM8993_EQ1, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Volume", WM8993_LEFT_ADC_DIGITAL_VOLUME, + WM8993_RIGHT_ADC_DIGITAL_VOLUME, 1, 96, 0, digital_tlv), +SOC_SINGLE("ADC High Pass Filter Switch", WM8993_ADC_CTRL, 8, 1, 0), +SOC_ENUM("ADC High Pass Filter Mode", adc_hpf), + +SOC_DOUBLE_R_TLV("Playback Volume", WM8993_LEFT_DAC_DIGITAL_VOLUME, + WM8993_RIGHT_DAC_DIGITAL_VOLUME, 1, 96, 0, digital_tlv), +SOC_SINGLE_TLV("Playback Boost Volume", WM8993_AUDIO_INTERFACE_2, 10, 3, 0, + dac_boost_tlv), +SOC_ENUM("DAC Deemphasis", dac_deemph), + +SOC_SINGLE_TLV("SPKL DAC Volume", WM8993_SPKMIXL_ATTENUATION, + 2, 1, 1, wm_hubs_spkmix_tlv), + +SOC_SINGLE_TLV("SPKR DAC Volume", WM8993_SPKMIXR_ATTENUATION, + 2, 1, 1, wm_hubs_spkmix_tlv), +}; + +static const struct snd_kcontrol_new wm8993_eq_controls[] = { +SOC_SINGLE_TLV("EQ1 Volume", WM8993_EQ2, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 Volume", WM8993_EQ3, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 Volume", WM8993_EQ4, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 Volume", WM8993_EQ5, 0, 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ5 Volume", WM8993_EQ6, 0, 24, 0, eq_tlv), +}; + +static int clk_sys_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + return configure_clock(codec); + + case SND_SOC_DAPM_POST_PMD: + break; + } + + return 0; +} + +/* + * When used with DAC outputs only the WM8993 charge pump supports + * operation in class W mode, providing very low power consumption + * when used with digital sources. Enable and disable this mode + * automatically depending on the mixer configuration. + * + * Currently the only supported paths are the direct DAC->headphone + * paths (which provide minimum power consumption anyway). + */ +static int class_w_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct snd_soc_codec *codec = widget->codec; + struct wm8993_priv *wm8993 = codec->private_data; + int ret; + + /* Turn it off if we're using the main output mixer */ + if (ucontrol->value.integer.value[0] == 0) { + if (wm8993->class_w_users == 0) { + dev_dbg(codec->dev, "Disabling Class W\n"); + snd_soc_update_bits(codec, WM8993_CLASS_W_0, + WM8993_CP_DYN_FREQ | + WM8993_CP_DYN_V, + 0); + } + wm8993->class_w_users++; + } + + /* Implement the change */ + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + /* Enable it if we're using the direct DAC path */ + if (ucontrol->value.integer.value[0] == 1) { + if (wm8993->class_w_users == 1) { + dev_dbg(codec->dev, "Enabling Class W\n"); + snd_soc_update_bits(codec, WM8993_CLASS_W_0, + WM8993_CP_DYN_FREQ | + WM8993_CP_DYN_V, + WM8993_CP_DYN_FREQ | + WM8993_CP_DYN_V); + } + wm8993->class_w_users--; + } + + dev_dbg(codec->dev, "Indirect DAC use count now %d\n", + wm8993->class_w_users); + + return ret; +} + +#define SOC_DAPM_ENUM_W(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_double, \ + .put = class_w_put, \ + .private_value = (unsigned long)&xenum } + +static const char *hp_mux_text[] = { + "Mixer", + "DAC", +}; + +static const struct soc_enum hpl_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); + +static const struct snd_kcontrol_new hpl_mux = + SOC_DAPM_ENUM_W("Left Headphone Mux", hpl_enum); + +static const struct soc_enum hpr_enum = + SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); + +static const struct snd_kcontrol_new hpr_mux = + SOC_DAPM_ENUM_W("Right Headphone Mux", hpr_enum); + +static const struct snd_kcontrol_new left_speaker_mixer[] = { +SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 7, 1, 0), +SOC_DAPM_SINGLE("IN1LP Switch", WM8993_SPEAKER_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8993_SPEAKER_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8993_SPEAKER_MIXER, 6, 1, 0), +}; + +static const struct snd_kcontrol_new right_speaker_mixer[] = { +SOC_DAPM_SINGLE("Input Switch", WM8993_SPEAKER_MIXER, 6, 1, 0), +SOC_DAPM_SINGLE("IN1RP Switch", WM8993_SPEAKER_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8993_SPEAKER_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8993_SPEAKER_MIXER, 0, 1, 0), +}; + +static const char *aif_text[] = { + "Left", "Right" +}; + +static const struct soc_enum aifoutl_enum = + SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text); + +static const struct snd_kcontrol_new aifoutl_mux = + SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); + +static const struct soc_enum aifoutr_enum = + SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text); + +static const struct snd_kcontrol_new aifoutr_mux = + SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); + +static const struct soc_enum aifinl_enum = + SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text); + +static const struct snd_kcontrol_new aifinl_mux = + SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); + +static const struct soc_enum aifinr_enum = + SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text); + +static const struct snd_kcontrol_new aifinr_mux = + SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); + +static const char *sidetone_text[] = { + "None", "Left", "Right" +}; + +static const struct soc_enum sidetonel_enum = + SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text); + +static const struct snd_kcontrol_new sidetonel_mux = + SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum); + +static const struct soc_enum sidetoner_enum = + SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text); + +static const struct snd_kcontrol_new sidetoner_mux = + SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum); + +static const struct snd_soc_dapm_widget wm8993_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8993_BUS_CONTROL_1, 1, 0, clk_sys_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("TOCLK", WM8993_CLOCKING_1, 14, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8993_CLOCKING_3, 0, 0, NULL, 0), + +SND_SOC_DAPM_ADC("ADCL", NULL, WM8993_POWER_MANAGEMENT_2, 1, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, WM8993_POWER_MANAGEMENT_2, 0, 0), + +SND_SOC_DAPM_MUX("AIFOUTL Mux", SND_SOC_NOPM, 0, 0, &aifoutl_mux), +SND_SOC_DAPM_MUX("AIFOUTR Mux", SND_SOC_NOPM, 0, 0, &aifoutr_mux), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &aifinl_mux), +SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &aifinr_mux), + +SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &sidetonel_mux), +SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &sidetoner_mux), + +SND_SOC_DAPM_DAC("DACL", NULL, WM8993_POWER_MANAGEMENT_3, 1, 0), +SND_SOC_DAPM_DAC("DACR", NULL, WM8993_POWER_MANAGEMENT_3, 0, 0), + +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), + +SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), + +}; + +static const struct snd_soc_dapm_route routes[] = { + { "ADCL", NULL, "CLK_SYS" }, + { "ADCL", NULL, "CLK_DSP" }, + { "ADCR", NULL, "CLK_SYS" }, + { "ADCR", NULL, "CLK_DSP" }, + + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, + + { "DACL Mux", "Left", "AIFINL" }, + { "DACL Mux", "Right", "AIFINR" }, + { "DACR Mux", "Left", "AIFINL" }, + { "DACR Mux", "Right", "AIFINR" }, + + { "DACL Sidetone", "Left", "ADCL" }, + { "DACL Sidetone", "Right", "ADCR" }, + { "DACR Sidetone", "Left", "ADCL" }, + { "DACR Sidetone", "Right", "ADCR" }, + + { "DACL", NULL, "CLK_SYS" }, + { "DACL", NULL, "CLK_DSP" }, + { "DACL", NULL, "DACL Mux" }, + { "DACL", NULL, "DACL Sidetone" }, + { "DACR", NULL, "CLK_SYS" }, + { "DACR", NULL, "CLK_DSP" }, + { "DACR", NULL, "DACR Mux" }, + { "DACR", NULL, "DACR Sidetone" }, + + { "Left Output Mixer", "DAC Switch", "DACL" }, + + { "Right Output Mixer", "DAC Switch", "DACR" }, + + { "Left Output PGA", NULL, "CLK_SYS" }, + + { "Right Output PGA", NULL, "CLK_SYS" }, + + { "SPKL", "DAC Switch", "DACL" }, + { "SPKL", NULL, "CLK_SYS" }, + + { "SPKR", "DAC Switch", "DACR" }, + { "SPKR", NULL, "CLK_SYS" }, + + { "Left Headphone Mux", "DAC", "DACL" }, + { "Right Headphone Mux", "DAC", "DACR" }, +}; + +static int wm8993_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8993_priv *wm8993 = codec->private_data; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + /* VMID=2*40k */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_VMID_SEL_MASK, 0x2); + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_2, + WM8993_TSHUT_ENA, WM8993_TSHUT_ENA); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Bring up VMID with fast soft start */ + snd_soc_update_bits(codec, WM8993_ANTIPOP2, + WM8993_STARTUP_BIAS_ENA | + WM8993_VMID_BUF_ENA | + WM8993_VMID_RAMP_MASK | + WM8993_BIAS_SRC, + WM8993_STARTUP_BIAS_ENA | + WM8993_VMID_BUF_ENA | + WM8993_VMID_RAMP_MASK | + WM8993_BIAS_SRC); + + /* If either line output is single ended we + * need the VMID buffer */ + if (!wm8993->pdata.lineout1_diff || + !wm8993->pdata.lineout2_diff) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, + WM8993_LINEOUT_VMID_BUF_ENA); + + /* VMID=2*40k */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_VMID_SEL_MASK | + WM8993_BIAS_ENA, + WM8993_BIAS_ENA | 0x2); + msleep(32); + + /* Switch to normal bias */ + snd_soc_update_bits(codec, WM8993_ANTIPOP2, + WM8993_BIAS_SRC | + WM8993_STARTUP_BIAS_ENA, 0); + } + + /* VMID=2*240k */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_VMID_SEL_MASK, 0x4); + + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_2, + WM8993_TSHUT_ENA, 0); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, 0); + + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_VMID_SEL_MASK | WM8993_BIAS_ENA, + 0); + break; + } + + codec->bias_level = level; + + return 0; +} + +static int wm8993_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8993_priv *wm8993 = codec->private_data; + + switch (clk_id) { + case WM8993_SYSCLK_MCLK: + wm8993->mclk_rate = freq; + case WM8993_SYSCLK_FLL: + wm8993->sysclk_source = clk_id; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8993_priv *wm8993 = codec->private_data; + unsigned int aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1); + unsigned int aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4); + + aif1 &= ~(WM8993_BCLK_DIR | WM8993_AIF_BCLK_INV | + WM8993_AIF_LRCLK_INV | WM8993_AIF_FMT_MASK); + aif4 &= ~WM8993_LRCLK_DIR; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + wm8993->master = 0; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif4 |= WM8993_LRCLK_DIR; + wm8993->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif1 |= WM8993_BCLK_DIR; + wm8993->master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif1 |= WM8993_BCLK_DIR; + aif4 |= WM8993_LRCLK_DIR; + wm8993->master = 1; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + aif1 |= WM8993_AIF_LRCLK_INV; + case SND_SOC_DAIFMT_DSP_A: + aif1 |= 0x18; + break; + case SND_SOC_DAIFMT_I2S: + aif1 |= 0x10; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + aif1 |= 0x8; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + /* frame inversion not valid for DSP modes */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8993_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif1 |= WM8993_AIF_BCLK_INV | WM8993_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif1 |= WM8993_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif1 |= WM8993_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); + wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); + + return 0; +} + +static int wm8993_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8993_priv *wm8993 = codec->private_data; + int ret, i, best, best_val, cur_val; + unsigned int clocking1, clocking3, aif1, aif4; + + clocking1 = wm8993_read(codec, WM8993_CLOCKING_1); + clocking1 &= ~WM8993_BCLK_DIV_MASK; + + clocking3 = wm8993_read(codec, WM8993_CLOCKING_3); + clocking3 &= ~(WM8993_CLK_SYS_RATE_MASK | WM8993_SAMPLE_RATE_MASK); + + aif1 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_1); + aif1 &= ~WM8993_AIF_WL_MASK; + + aif4 = wm8993_read(codec, WM8993_AUDIO_INTERFACE_4); + aif4 &= ~WM8993_LRCLK_RATE_MASK; + + /* What BCLK do we need? */ + wm8993->fs = params_rate(params); + wm8993->bclk = 2 * wm8993->fs; + if (wm8993->tdm_slots) { + dev_dbg(codec->dev, "Configuring for %d %d bit TDM slots\n", + wm8993->tdm_slots, wm8993->tdm_width); + wm8993->bclk *= wm8993->tdm_width * wm8993->tdm_slots; + } else { + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wm8993->bclk *= 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wm8993->bclk *= 20; + aif1 |= 0x8; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wm8993->bclk *= 24; + aif1 |= 0x10; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wm8993->bclk *= 32; + aif1 |= 0x18; + break; + default: + return -EINVAL; + } + } + + dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8993->bclk); + + ret = configure_clock(codec); + if (ret != 0) + return ret; + + /* Select nearest CLK_SYS_RATE */ + best = 0; + best_val = abs((wm8993->sysclk_rate / clk_sys_rates[0].ratio) + - wm8993->fs); + for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) { + cur_val = abs((wm8993->sysclk_rate / + clk_sys_rates[i].ratio) - wm8993->fs);; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n", + clk_sys_rates[best].ratio); + clocking3 |= (clk_sys_rates[best].clk_sys_rate + << WM8993_CLK_SYS_RATE_SHIFT); + + /* SAMPLE_RATE */ + best = 0; + best_val = abs(wm8993->fs - sample_rates[0].rate); + for (i = 1; i < ARRAY_SIZE(sample_rates); i++) { + /* Closest match */ + cur_val = abs(wm8993->fs - sample_rates[i].rate); + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n", + sample_rates[best].rate); + clocking3 |= (sample_rates[best].sample_rate + << WM8993_SAMPLE_RATE_SHIFT); + + /* BCLK_DIV */ + best = 0; + best_val = INT_MAX; + for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) { + cur_val = ((wm8993->sysclk_rate * 10) / bclk_divs[i].div) + - wm8993->bclk; + if (cur_val < 0) /* Table is sorted */ + break; + if (cur_val < best_val) { + best = i; + best_val = cur_val; + } + } + wm8993->bclk = (wm8993->sysclk_rate * 10) / bclk_divs[best].div; + dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n", + bclk_divs[best].div, wm8993->bclk); + clocking1 |= bclk_divs[best].bclk_div << WM8993_BCLK_DIV_SHIFT; + + /* LRCLK is a simple fraction of BCLK */ + dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8993->bclk / wm8993->fs); + aif4 |= wm8993->bclk / wm8993->fs; + + wm8993_write(codec, WM8993_CLOCKING_1, clocking1); + wm8993_write(codec, WM8993_CLOCKING_3, clocking3); + wm8993_write(codec, WM8993_AUDIO_INTERFACE_1, aif1); + wm8993_write(codec, WM8993_AUDIO_INTERFACE_4, aif4); + + /* ReTune Mobile? */ + if (wm8993->pdata.num_retune_configs) { + u16 eq1 = wm8993_read(codec, WM8993_EQ1); + struct wm8993_retune_mobile_setting *s; + + best = 0; + best_val = abs(wm8993->pdata.retune_configs[0].rate + - wm8993->fs); + for (i = 0; i < wm8993->pdata.num_retune_configs; i++) { + cur_val = abs(wm8993->pdata.retune_configs[i].rate + - wm8993->fs); + if (cur_val < best_val) { + best_val = cur_val; + best = i; + } + } + s = &wm8993->pdata.retune_configs[best]; + + dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", + s->name, s->rate); + + /* Disable EQ while we reconfigure */ + snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, 0); + + for (i = 1; i < ARRAY_SIZE(s->config); i++) + wm8993_write(codec, WM8993_EQ1 + i, s->config[i]); + + snd_soc_update_bits(codec, WM8993_EQ1, WM8993_EQ_ENA, eq1); + } + + return 0; +} + +static int wm8993_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg; + + reg = wm8993_read(codec, WM8993_DAC_CTRL); + + if (mute) + reg |= WM8993_DAC_MUTE; + else + reg &= ~WM8993_DAC_MUTE; + + wm8993_write(codec, WM8993_DAC_CTRL, reg); + + return 0; +} + +static int wm8993_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8993_priv *wm8993 = codec->private_data; + int aif1 = 0; + int aif2 = 0; + + /* Don't need to validate anything if we're turning off TDM */ + if (slots == 0) { + wm8993->tdm_slots = 0; + goto out; + } + + /* Note that we allow configurations we can't handle ourselves - + * for example, we can generate clocks for slots 2 and up even if + * we can't use those slots ourselves. + */ + aif1 |= WM8993_AIFADC_TDM; + aif2 |= WM8993_AIFDAC_TDM; + + switch (rx_mask) { + case 3: + break; + case 0xc: + aif1 |= WM8993_AIFADC_TDM_CHAN; + break; + default: + return -EINVAL; + } + + + switch (tx_mask) { + case 3: + break; + case 0xc: + aif2 |= WM8993_AIFDAC_TDM_CHAN; + break; + default: + return -EINVAL; + } + +out: + wm8993->tdm_width = slot_width; + wm8993->tdm_slots = slots / 2; + + snd_soc_update_bits(codec, WM8993_AUDIO_INTERFACE_1, + WM8993_AIFADC_TDM | WM8993_AIFADC_TDM_CHAN, aif1); + snd_soc_update_bits(codec, WM8993_AUDIO_INTERFACE_2, + WM8993_AIFDAC_TDM | WM8993_AIFDAC_TDM_CHAN, aif2); + + return 0; +} + +static struct snd_soc_dai_ops wm8993_ops = { + .set_sysclk = wm8993_set_sysclk, + .set_fmt = wm8993_set_dai_fmt, + .hw_params = wm8993_hw_params, + .digital_mute = wm8993_digital_mute, + .set_pll = wm8993_set_fll, + .set_tdm_slot = wm8993_set_tdm_slot, +}; + +#define WM8993_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8993_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai wm8993_dai = { + .name = "WM8993", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8993_RATES, + .formats = WM8993_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8993_RATES, + .formats = WM8993_FORMATS, + }, + .ops = &wm8993_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8993_dai); + +static struct snd_soc_codec *wm8993_codec; + +static int wm8993_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct wm8993_priv *wm8993; + int ret = 0; + + if (!wm8993_codec) { + dev_err(&pdev->dev, "I2C device not yet probed\n"); + goto err; + } + + socdev->card->codec = wm8993_codec; + codec = wm8993_codec; + wm8993 = codec->private_data; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8993_snd_controls, + ARRAY_SIZE(wm8993_snd_controls)); + if (wm8993->pdata.num_retune_configs != 0) { + dev_dbg(codec->dev, "Using ReTune Mobile\n"); + } else { + dev_dbg(codec->dev, "No ReTune Mobile, using normal EQ\n"); + snd_soc_add_controls(codec, wm8993_eq_controls, + ARRAY_SIZE(wm8993_eq_controls)); + } + + snd_soc_dapm_new_controls(codec, wm8993_dapm_widgets, + ARRAY_SIZE(wm8993_dapm_widgets)); + wm_hubs_add_analogue_controls(codec); + + snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, + wm8993->pdata.lineout2_diff); + + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +err: + return ret; +} + +static int wm8993_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8993 = { + .probe = wm8993_probe, + .remove = wm8993_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8993); + +static int wm8993_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8993_priv *wm8993; + struct snd_soc_codec *codec; + unsigned int val; + int ret; + + if (wm8993_codec) { + dev_err(&i2c->dev, "A WM8993 is already registered\n"); + return -EINVAL; + } + + wm8993 = kzalloc(sizeof(struct wm8993_priv), GFP_KERNEL); + if (wm8993 == NULL) + return -ENOMEM; + + codec = &wm8993->codec; + if (i2c->dev.platform_data) + memcpy(&wm8993->pdata, i2c->dev.platform_data, + sizeof(wm8993->pdata)); + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "WM8993"; + codec->read = wm8993_read; + codec->write = wm8993_write; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->reg_cache = wm8993->reg_cache; + codec->reg_cache_size = ARRAY_SIZE(wm8993->reg_cache); + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8993_set_bias_level; + codec->dai = &wm8993_dai; + codec->num_dai = 1; + codec->private_data = wm8993; + + memcpy(wm8993->reg_cache, wm8993_reg_defaults, + sizeof(wm8993->reg_cache)); + + i2c_set_clientdata(i2c, wm8993); + codec->control_data = i2c; + wm8993_codec = codec; + + codec->dev = &i2c->dev; + + val = wm8993_read_hw(codec, WM8993_SOFTWARE_RESET); + if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) { + dev_err(codec->dev, "Invalid ID register value %x\n", val); + ret = -EINVAL; + goto err; + } + + ret = wm8993_write(codec, WM8993_SOFTWARE_RESET, 0xffff); + if (ret != 0) + goto err; + + /* By default we're using the output mixers */ + wm8993->class_w_users = 2; + + /* Latch volume update bits and default ZC on */ + snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, + WM8993_DAC_VU, WM8993_DAC_VU); + snd_soc_update_bits(codec, WM8993_RIGHT_ADC_DIGITAL_VOLUME, + WM8993_ADC_VU, WM8993_ADC_VU); + + /* Manualy manage the HPOUT sequencing for independent stereo + * control. */ + snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0, + WM8993_HPOUT1_AUTO_PU, 0); + + /* Use automatic clock configuration */ + snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0); + + if (!wm8993->pdata.lineout1_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER1, + WM8993_LINEOUT1_MODE, + WM8993_LINEOUT1_MODE); + if (!wm8993->pdata.lineout2_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER2, + WM8993_LINEOUT2_MODE, + WM8993_LINEOUT2_MODE); + + if (wm8993->pdata.lineout1fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); + + if (wm8993->pdata.lineout2fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + + /* Apply the microphone bias/detection configuration - the + * platform data is directly applicable to the register. */ + snd_soc_update_bits(codec, WM8993_MICBIAS, + WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | + WM8993_MICB1_LVL | WM8993_MICB2_LVL, + wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT | + wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT | + wm8993->pdata.micbias1_lvl | + wm8993->pdata.micbias1_lvl << 1); + + ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret != 0) + goto err; + + wm8993_dai.dev = codec->dev; + + ret = snd_soc_register_dai(&wm8993_dai); + if (ret != 0) + goto err_bias; + + ret = snd_soc_register_codec(codec); + + return 0; + +err_bias: + wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); +err: + wm8993_codec = NULL; + kfree(wm8993); + return ret; +} + +static int wm8993_i2c_remove(struct i2c_client *client) +{ + struct wm8993_priv *wm8993 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&wm8993->codec); + snd_soc_unregister_dai(&wm8993_dai); + + wm8993_set_bias_level(&wm8993->codec, SND_SOC_BIAS_OFF); + kfree(wm8993); + + return 0; +} + +static const struct i2c_device_id wm8993_i2c_id[] = { + { "wm8993", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8993_i2c_id); + +static struct i2c_driver wm8993_i2c_driver = { + .driver = { + .name = "WM8993", + .owner = THIS_MODULE, + }, + .probe = wm8993_i2c_probe, + .remove = wm8993_i2c_remove, + .id_table = wm8993_i2c_id, +}; + + +static int __init wm8993_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8993_i2c_driver); + if (ret != 0) + pr_err("WM8993: Unable to register I2C driver: %d\n", ret); + + return ret; +} +module_init(wm8993_modinit); + +static void __exit wm8993_exit(void) +{ + i2c_del_driver(&wm8993_i2c_driver); +} +module_exit(wm8993_exit); + + +MODULE_DESCRIPTION("ASoC WM8993 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8993.h b/sound/soc/codecs/wm8993.h new file mode 100644 index 000000000000..30e71ca88dad --- /dev/null +++ b/sound/soc/codecs/wm8993.h @@ -0,0 +1,2132 @@ +#ifndef WM8993_H +#define WM8993_H + +extern struct snd_soc_dai wm8993_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8993; + +#define WM8993_SYSCLK_MCLK 1 +#define WM8993_SYSCLK_FLL 2 + +#define WM8993_FLL_MCLK 1 +#define WM8993_FLL_BCLK 2 +#define WM8993_FLL_LRCLK 3 + +/* + * Register values. + */ +#define WM8993_SOFTWARE_RESET 0x00 +#define WM8993_POWER_MANAGEMENT_1 0x01 +#define WM8993_POWER_MANAGEMENT_2 0x02 +#define WM8993_POWER_MANAGEMENT_3 0x03 +#define WM8993_AUDIO_INTERFACE_1 0x04 +#define WM8993_AUDIO_INTERFACE_2 0x05 +#define WM8993_CLOCKING_1 0x06 +#define WM8993_CLOCKING_2 0x07 +#define WM8993_AUDIO_INTERFACE_3 0x08 +#define WM8993_AUDIO_INTERFACE_4 0x09 +#define WM8993_DAC_CTRL 0x0A +#define WM8993_LEFT_DAC_DIGITAL_VOLUME 0x0B +#define WM8993_RIGHT_DAC_DIGITAL_VOLUME 0x0C +#define WM8993_DIGITAL_SIDE_TONE 0x0D +#define WM8993_ADC_CTRL 0x0E +#define WM8993_LEFT_ADC_DIGITAL_VOLUME 0x0F +#define WM8993_RIGHT_ADC_DIGITAL_VOLUME 0x10 +#define WM8993_GPIO_CTRL_1 0x12 +#define WM8993_GPIO1 0x13 +#define WM8993_IRQ_DEBOUNCE 0x14 +#define WM8993_GPIOCTRL_2 0x16 +#define WM8993_GPIO_POL 0x17 +#define WM8993_LEFT_LINE_INPUT_1_2_VOLUME 0x18 +#define WM8993_LEFT_LINE_INPUT_3_4_VOLUME 0x19 +#define WM8993_RIGHT_LINE_INPUT_1_2_VOLUME 0x1A +#define WM8993_RIGHT_LINE_INPUT_3_4_VOLUME 0x1B +#define WM8993_LEFT_OUTPUT_VOLUME 0x1C +#define WM8993_RIGHT_OUTPUT_VOLUME 0x1D +#define WM8993_LINE_OUTPUTS_VOLUME 0x1E +#define WM8993_HPOUT2_VOLUME 0x1F +#define WM8993_LEFT_OPGA_VOLUME 0x20 +#define WM8993_RIGHT_OPGA_VOLUME 0x21 +#define WM8993_SPKMIXL_ATTENUATION 0x22 +#define WM8993_SPKMIXR_ATTENUATION 0x23 +#define WM8993_SPKOUT_MIXERS 0x24 +#define WM8993_SPKOUT_BOOST 0x25 +#define WM8993_SPEAKER_VOLUME_LEFT 0x26 +#define WM8993_SPEAKER_VOLUME_RIGHT 0x27 +#define WM8993_INPUT_MIXER2 0x28 +#define WM8993_INPUT_MIXER3 0x29 +#define WM8993_INPUT_MIXER4 0x2A +#define WM8993_INPUT_MIXER5 0x2B +#define WM8993_INPUT_MIXER6 0x2C +#define WM8993_OUTPUT_MIXER1 0x2D +#define WM8993_OUTPUT_MIXER2 0x2E +#define WM8993_OUTPUT_MIXER3 0x2F +#define WM8993_OUTPUT_MIXER4 0x30 +#define WM8993_OUTPUT_MIXER5 0x31 +#define WM8993_OUTPUT_MIXER6 0x32 +#define WM8993_HPOUT2_MIXER 0x33 +#define WM8993_LINE_MIXER1 0x34 +#define WM8993_LINE_MIXER2 0x35 +#define WM8993_SPEAKER_MIXER 0x36 +#define WM8993_ADDITIONAL_CONTROL 0x37 +#define WM8993_ANTIPOP1 0x38 +#define WM8993_ANTIPOP2 0x39 +#define WM8993_MICBIAS 0x3A +#define WM8993_FLL_CONTROL_1 0x3C +#define WM8993_FLL_CONTROL_2 0x3D +#define WM8993_FLL_CONTROL_3 0x3E +#define WM8993_FLL_CONTROL_4 0x3F +#define WM8993_FLL_CONTROL_5 0x40 +#define WM8993_CLOCKING_3 0x41 +#define WM8993_CLOCKING_4 0x42 +#define WM8993_MW_SLAVE_CONTROL 0x43 +#define WM8993_BUS_CONTROL_1 0x45 +#define WM8993_WRITE_SEQUENCER_0 0x46 +#define WM8993_WRITE_SEQUENCER_1 0x47 +#define WM8993_WRITE_SEQUENCER_2 0x48 +#define WM8993_WRITE_SEQUENCER_3 0x49 +#define WM8993_WRITE_SEQUENCER_4 0x4A +#define WM8993_WRITE_SEQUENCER_5 0x4B +#define WM8993_CHARGE_PUMP_1 0x4C +#define WM8993_CLASS_W_0 0x51 +#define WM8993_DC_SERVO_0 0x54 +#define WM8993_DC_SERVO_1 0x55 +#define WM8993_DC_SERVO_3 0x57 +#define WM8993_DC_SERVO_READBACK_0 0x58 +#define WM8993_DC_SERVO_READBACK_1 0x59 +#define WM8993_DC_SERVO_READBACK_2 0x5A +#define WM8993_ANALOGUE_HP_0 0x60 +#define WM8993_EQ1 0x62 +#define WM8993_EQ2 0x63 +#define WM8993_EQ3 0x64 +#define WM8993_EQ4 0x65 +#define WM8993_EQ5 0x66 +#define WM8993_EQ6 0x67 +#define WM8993_EQ7 0x68 +#define WM8993_EQ8 0x69 +#define WM8993_EQ9 0x6A +#define WM8993_EQ10 0x6B +#define WM8993_EQ11 0x6C +#define WM8993_EQ12 0x6D +#define WM8993_EQ13 0x6E +#define WM8993_EQ14 0x6F +#define WM8993_EQ15 0x70 +#define WM8993_EQ16 0x71 +#define WM8993_EQ17 0x72 +#define WM8993_EQ18 0x73 +#define WM8993_EQ19 0x74 +#define WM8993_EQ20 0x75 +#define WM8993_EQ21 0x76 +#define WM8993_EQ22 0x77 +#define WM8993_EQ23 0x78 +#define WM8993_EQ24 0x79 +#define WM8993_DIGITAL_PULLS 0x7A +#define WM8993_DRC_CONTROL_1 0x7B +#define WM8993_DRC_CONTROL_2 0x7C +#define WM8993_DRC_CONTROL_3 0x7D +#define WM8993_DRC_CONTROL_4 0x7E + +#define WM8993_REGISTER_COUNT 0x7F +#define WM8993_MAX_REGISTER 0x7E + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - Software Reset + */ +#define WM8993_SW_RESET_MASK 0xFFFF /* SW_RESET - [15:0] */ +#define WM8993_SW_RESET_SHIFT 0 /* SW_RESET - [15:0] */ +#define WM8993_SW_RESET_WIDTH 16 /* SW_RESET - [15:0] */ + +/* + * R1 (0x01) - Power Management (1) + */ +#define WM8993_SPKOUTR_ENA 0x2000 /* SPKOUTR_ENA */ +#define WM8993_SPKOUTR_ENA_MASK 0x2000 /* SPKOUTR_ENA */ +#define WM8993_SPKOUTR_ENA_SHIFT 13 /* SPKOUTR_ENA */ +#define WM8993_SPKOUTR_ENA_WIDTH 1 /* SPKOUTR_ENA */ +#define WM8993_SPKOUTL_ENA 0x1000 /* SPKOUTL_ENA */ +#define WM8993_SPKOUTL_ENA_MASK 0x1000 /* SPKOUTL_ENA */ +#define WM8993_SPKOUTL_ENA_SHIFT 12 /* SPKOUTL_ENA */ +#define WM8993_SPKOUTL_ENA_WIDTH 1 /* SPKOUTL_ENA */ +#define WM8993_HPOUT2_ENA 0x0800 /* HPOUT2_ENA */ +#define WM8993_HPOUT2_ENA_MASK 0x0800 /* HPOUT2_ENA */ +#define WM8993_HPOUT2_ENA_SHIFT 11 /* HPOUT2_ENA */ +#define WM8993_HPOUT2_ENA_WIDTH 1 /* HPOUT2_ENA */ +#define WM8993_HPOUT1L_ENA 0x0200 /* HPOUT1L_ENA */ +#define WM8993_HPOUT1L_ENA_MASK 0x0200 /* HPOUT1L_ENA */ +#define WM8993_HPOUT1L_ENA_SHIFT 9 /* HPOUT1L_ENA */ +#define WM8993_HPOUT1L_ENA_WIDTH 1 /* HPOUT1L_ENA */ +#define WM8993_HPOUT1R_ENA 0x0100 /* HPOUT1R_ENA */ +#define WM8993_HPOUT1R_ENA_MASK 0x0100 /* HPOUT1R_ENA */ +#define WM8993_HPOUT1R_ENA_SHIFT 8 /* HPOUT1R_ENA */ +#define WM8993_HPOUT1R_ENA_WIDTH 1 /* HPOUT1R_ENA */ +#define WM8993_MICB2_ENA 0x0020 /* MICB2_ENA */ +#define WM8993_MICB2_ENA_MASK 0x0020 /* MICB2_ENA */ +#define WM8993_MICB2_ENA_SHIFT 5 /* MICB2_ENA */ +#define WM8993_MICB2_ENA_WIDTH 1 /* MICB2_ENA */ +#define WM8993_MICB1_ENA 0x0010 /* MICB1_ENA */ +#define WM8993_MICB1_ENA_MASK 0x0010 /* MICB1_ENA */ +#define WM8993_MICB1_ENA_SHIFT 4 /* MICB1_ENA */ +#define WM8993_MICB1_ENA_WIDTH 1 /* MICB1_ENA */ +#define WM8993_VMID_SEL_MASK 0x0006 /* VMID_SEL - [2:1] */ +#define WM8993_VMID_SEL_SHIFT 1 /* VMID_SEL - [2:1] */ +#define WM8993_VMID_SEL_WIDTH 2 /* VMID_SEL - [2:1] */ +#define WM8993_BIAS_ENA 0x0001 /* BIAS_ENA */ +#define WM8993_BIAS_ENA_MASK 0x0001 /* BIAS_ENA */ +#define WM8993_BIAS_ENA_SHIFT 0 /* BIAS_ENA */ +#define WM8993_BIAS_ENA_WIDTH 1 /* BIAS_ENA */ + +/* + * R2 (0x02) - Power Management (2) + */ +#define WM8993_TSHUT_ENA 0x4000 /* TSHUT_ENA */ +#define WM8993_TSHUT_ENA_MASK 0x4000 /* TSHUT_ENA */ +#define WM8993_TSHUT_ENA_SHIFT 14 /* TSHUT_ENA */ +#define WM8993_TSHUT_ENA_WIDTH 1 /* TSHUT_ENA */ +#define WM8993_TSHUT_OPDIS 0x2000 /* TSHUT_OPDIS */ +#define WM8993_TSHUT_OPDIS_MASK 0x2000 /* TSHUT_OPDIS */ +#define WM8993_TSHUT_OPDIS_SHIFT 13 /* TSHUT_OPDIS */ +#define WM8993_TSHUT_OPDIS_WIDTH 1 /* TSHUT_OPDIS */ +#define WM8993_OPCLK_ENA 0x0800 /* OPCLK_ENA */ +#define WM8993_OPCLK_ENA_MASK 0x0800 /* OPCLK_ENA */ +#define WM8993_OPCLK_ENA_SHIFT 11 /* OPCLK_ENA */ +#define WM8993_OPCLK_ENA_WIDTH 1 /* OPCLK_ENA */ +#define WM8993_MIXINL_ENA 0x0200 /* MIXINL_ENA */ +#define WM8993_MIXINL_ENA_MASK 0x0200 /* MIXINL_ENA */ +#define WM8993_MIXINL_ENA_SHIFT 9 /* MIXINL_ENA */ +#define WM8993_MIXINL_ENA_WIDTH 1 /* MIXINL_ENA */ +#define WM8993_MIXINR_ENA 0x0100 /* MIXINR_ENA */ +#define WM8993_MIXINR_ENA_MASK 0x0100 /* MIXINR_ENA */ +#define WM8993_MIXINR_ENA_SHIFT 8 /* MIXINR_ENA */ +#define WM8993_MIXINR_ENA_WIDTH 1 /* MIXINR_ENA */ +#define WM8993_IN2L_ENA 0x0080 /* IN2L_ENA */ +#define WM8993_IN2L_ENA_MASK 0x0080 /* IN2L_ENA */ +#define WM8993_IN2L_ENA_SHIFT 7 /* IN2L_ENA */ +#define WM8993_IN2L_ENA_WIDTH 1 /* IN2L_ENA */ +#define WM8993_IN1L_ENA 0x0040 /* IN1L_ENA */ +#define WM8993_IN1L_ENA_MASK 0x0040 /* IN1L_ENA */ +#define WM8993_IN1L_ENA_SHIFT 6 /* IN1L_ENA */ +#define WM8993_IN1L_ENA_WIDTH 1 /* IN1L_ENA */ +#define WM8993_IN2R_ENA 0x0020 /* IN2R_ENA */ +#define WM8993_IN2R_ENA_MASK 0x0020 /* IN2R_ENA */ +#define WM8993_IN2R_ENA_SHIFT 5 /* IN2R_ENA */ +#define WM8993_IN2R_ENA_WIDTH 1 /* IN2R_ENA */ +#define WM8993_IN1R_ENA 0x0010 /* IN1R_ENA */ +#define WM8993_IN1R_ENA_MASK 0x0010 /* IN1R_ENA */ +#define WM8993_IN1R_ENA_SHIFT 4 /* IN1R_ENA */ +#define WM8993_IN1R_ENA_WIDTH 1 /* IN1R_ENA */ +#define WM8993_ADCL_ENA 0x0002 /* ADCL_ENA */ +#define WM8993_ADCL_ENA_MASK 0x0002 /* ADCL_ENA */ +#define WM8993_ADCL_ENA_SHIFT 1 /* ADCL_ENA */ +#define WM8993_ADCL_ENA_WIDTH 1 /* ADCL_ENA */ +#define WM8993_ADCR_ENA 0x0001 /* ADCR_ENA */ +#define WM8993_ADCR_ENA_MASK 0x0001 /* ADCR_ENA */ +#define WM8993_ADCR_ENA_SHIFT 0 /* ADCR_ENA */ +#define WM8993_ADCR_ENA_WIDTH 1 /* ADCR_ENA */ + +/* + * R3 (0x03) - Power Management (3) + */ +#define WM8993_LINEOUT1N_ENA 0x2000 /* LINEOUT1N_ENA */ +#define WM8993_LINEOUT1N_ENA_MASK 0x2000 /* LINEOUT1N_ENA */ +#define WM8993_LINEOUT1N_ENA_SHIFT 13 /* LINEOUT1N_ENA */ +#define WM8993_LINEOUT1N_ENA_WIDTH 1 /* LINEOUT1N_ENA */ +#define WM8993_LINEOUT1P_ENA 0x1000 /* LINEOUT1P_ENA */ +#define WM8993_LINEOUT1P_ENA_MASK 0x1000 /* LINEOUT1P_ENA */ +#define WM8993_LINEOUT1P_ENA_SHIFT 12 /* LINEOUT1P_ENA */ +#define WM8993_LINEOUT1P_ENA_WIDTH 1 /* LINEOUT1P_ENA */ +#define WM8993_LINEOUT2N_ENA 0x0800 /* LINEOUT2N_ENA */ +#define WM8993_LINEOUT2N_ENA_MASK 0x0800 /* LINEOUT2N_ENA */ +#define WM8993_LINEOUT2N_ENA_SHIFT 11 /* LINEOUT2N_ENA */ +#define WM8993_LINEOUT2N_ENA_WIDTH 1 /* LINEOUT2N_ENA */ +#define WM8993_LINEOUT2P_ENA 0x0400 /* LINEOUT2P_ENA */ +#define WM8993_LINEOUT2P_ENA_MASK 0x0400 /* LINEOUT2P_ENA */ +#define WM8993_LINEOUT2P_ENA_SHIFT 10 /* LINEOUT2P_ENA */ +#define WM8993_LINEOUT2P_ENA_WIDTH 1 /* LINEOUT2P_ENA */ +#define WM8993_SPKRVOL_ENA 0x0200 /* SPKRVOL_ENA */ +#define WM8993_SPKRVOL_ENA_MASK 0x0200 /* SPKRVOL_ENA */ +#define WM8993_SPKRVOL_ENA_SHIFT 9 /* SPKRVOL_ENA */ +#define WM8993_SPKRVOL_ENA_WIDTH 1 /* SPKRVOL_ENA */ +#define WM8993_SPKLVOL_ENA 0x0100 /* SPKLVOL_ENA */ +#define WM8993_SPKLVOL_ENA_MASK 0x0100 /* SPKLVOL_ENA */ +#define WM8993_SPKLVOL_ENA_SHIFT 8 /* SPKLVOL_ENA */ +#define WM8993_SPKLVOL_ENA_WIDTH 1 /* SPKLVOL_ENA */ +#define WM8993_MIXOUTLVOL_ENA 0x0080 /* MIXOUTLVOL_ENA */ +#define WM8993_MIXOUTLVOL_ENA_MASK 0x0080 /* MIXOUTLVOL_ENA */ +#define WM8993_MIXOUTLVOL_ENA_SHIFT 7 /* MIXOUTLVOL_ENA */ +#define WM8993_MIXOUTLVOL_ENA_WIDTH 1 /* MIXOUTLVOL_ENA */ +#define WM8993_MIXOUTRVOL_ENA 0x0040 /* MIXOUTRVOL_ENA */ +#define WM8993_MIXOUTRVOL_ENA_MASK 0x0040 /* MIXOUTRVOL_ENA */ +#define WM8993_MIXOUTRVOL_ENA_SHIFT 6 /* MIXOUTRVOL_ENA */ +#define WM8993_MIXOUTRVOL_ENA_WIDTH 1 /* MIXOUTRVOL_ENA */ +#define WM8993_MIXOUTL_ENA 0x0020 /* MIXOUTL_ENA */ +#define WM8993_MIXOUTL_ENA_MASK 0x0020 /* MIXOUTL_ENA */ +#define WM8993_MIXOUTL_ENA_SHIFT 5 /* MIXOUTL_ENA */ +#define WM8993_MIXOUTL_ENA_WIDTH 1 /* MIXOUTL_ENA */ +#define WM8993_MIXOUTR_ENA 0x0010 /* MIXOUTR_ENA */ +#define WM8993_MIXOUTR_ENA_MASK 0x0010 /* MIXOUTR_ENA */ +#define WM8993_MIXOUTR_ENA_SHIFT 4 /* MIXOUTR_ENA */ +#define WM8993_MIXOUTR_ENA_WIDTH 1 /* MIXOUTR_ENA */ +#define WM8993_DACL_ENA 0x0002 /* DACL_ENA */ +#define WM8993_DACL_ENA_MASK 0x0002 /* DACL_ENA */ +#define WM8993_DACL_ENA_SHIFT 1 /* DACL_ENA */ +#define WM8993_DACL_ENA_WIDTH 1 /* DACL_ENA */ +#define WM8993_DACR_ENA 0x0001 /* DACR_ENA */ +#define WM8993_DACR_ENA_MASK 0x0001 /* DACR_ENA */ +#define WM8993_DACR_ENA_SHIFT 0 /* DACR_ENA */ +#define WM8993_DACR_ENA_WIDTH 1 /* DACR_ENA */ + +/* + * R4 (0x04) - Audio Interface (1) + */ +#define WM8993_AIFADCL_SRC 0x8000 /* AIFADCL_SRC */ +#define WM8993_AIFADCL_SRC_MASK 0x8000 /* AIFADCL_SRC */ +#define WM8993_AIFADCL_SRC_SHIFT 15 /* AIFADCL_SRC */ +#define WM8993_AIFADCL_SRC_WIDTH 1 /* AIFADCL_SRC */ +#define WM8993_AIFADCR_SRC 0x4000 /* AIFADCR_SRC */ +#define WM8993_AIFADCR_SRC_MASK 0x4000 /* AIFADCR_SRC */ +#define WM8993_AIFADCR_SRC_SHIFT 14 /* AIFADCR_SRC */ +#define WM8993_AIFADCR_SRC_WIDTH 1 /* AIFADCR_SRC */ +#define WM8993_AIFADC_TDM 0x2000 /* AIFADC_TDM */ +#define WM8993_AIFADC_TDM_MASK 0x2000 /* AIFADC_TDM */ +#define WM8993_AIFADC_TDM_SHIFT 13 /* AIFADC_TDM */ +#define WM8993_AIFADC_TDM_WIDTH 1 /* AIFADC_TDM */ +#define WM8993_AIFADC_TDM_CHAN 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8993_AIFADC_TDM_CHAN_MASK 0x1000 /* AIFADC_TDM_CHAN */ +#define WM8993_AIFADC_TDM_CHAN_SHIFT 12 /* AIFADC_TDM_CHAN */ +#define WM8993_AIFADC_TDM_CHAN_WIDTH 1 /* AIFADC_TDM_CHAN */ +#define WM8993_BCLK_DIR 0x0200 /* BCLK_DIR */ +#define WM8993_BCLK_DIR_MASK 0x0200 /* BCLK_DIR */ +#define WM8993_BCLK_DIR_SHIFT 9 /* BCLK_DIR */ +#define WM8993_BCLK_DIR_WIDTH 1 /* BCLK_DIR */ +#define WM8993_AIF_BCLK_INV 0x0100 /* AIF_BCLK_INV */ +#define WM8993_AIF_BCLK_INV_MASK 0x0100 /* AIF_BCLK_INV */ +#define WM8993_AIF_BCLK_INV_SHIFT 8 /* AIF_BCLK_INV */ +#define WM8993_AIF_BCLK_INV_WIDTH 1 /* AIF_BCLK_INV */ +#define WM8993_AIF_LRCLK_INV 0x0080 /* AIF_LRCLK_INV */ +#define WM8993_AIF_LRCLK_INV_MASK 0x0080 /* AIF_LRCLK_INV */ +#define WM8993_AIF_LRCLK_INV_SHIFT 7 /* AIF_LRCLK_INV */ +#define WM8993_AIF_LRCLK_INV_WIDTH 1 /* AIF_LRCLK_INV */ +#define WM8993_AIF_WL_MASK 0x0060 /* AIF_WL - [6:5] */ +#define WM8993_AIF_WL_SHIFT 5 /* AIF_WL - [6:5] */ +#define WM8993_AIF_WL_WIDTH 2 /* AIF_WL - [6:5] */ +#define WM8993_AIF_FMT_MASK 0x0018 /* AIF_FMT - [4:3] */ +#define WM8993_AIF_FMT_SHIFT 3 /* AIF_FMT - [4:3] */ +#define WM8993_AIF_FMT_WIDTH 2 /* AIF_FMT - [4:3] */ + +/* + * R5 (0x05) - Audio Interface (2) + */ +#define WM8993_AIFDACL_SRC 0x8000 /* AIFDACL_SRC */ +#define WM8993_AIFDACL_SRC_MASK 0x8000 /* AIFDACL_SRC */ +#define WM8993_AIFDACL_SRC_SHIFT 15 /* AIFDACL_SRC */ +#define WM8993_AIFDACL_SRC_WIDTH 1 /* AIFDACL_SRC */ +#define WM8993_AIFDACR_SRC 0x4000 /* AIFDACR_SRC */ +#define WM8993_AIFDACR_SRC_MASK 0x4000 /* AIFDACR_SRC */ +#define WM8993_AIFDACR_SRC_SHIFT 14 /* AIFDACR_SRC */ +#define WM8993_AIFDACR_SRC_WIDTH 1 /* AIFDACR_SRC */ +#define WM8993_AIFDAC_TDM 0x2000 /* AIFDAC_TDM */ +#define WM8993_AIFDAC_TDM_MASK 0x2000 /* AIFDAC_TDM */ +#define WM8993_AIFDAC_TDM_SHIFT 13 /* AIFDAC_TDM */ +#define WM8993_AIFDAC_TDM_WIDTH 1 /* AIFDAC_TDM */ +#define WM8993_AIFDAC_TDM_CHAN 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8993_AIFDAC_TDM_CHAN_MASK 0x1000 /* AIFDAC_TDM_CHAN */ +#define WM8993_AIFDAC_TDM_CHAN_SHIFT 12 /* AIFDAC_TDM_CHAN */ +#define WM8993_AIFDAC_TDM_CHAN_WIDTH 1 /* AIFDAC_TDM_CHAN */ +#define WM8993_DAC_BOOST_MASK 0x0C00 /* DAC_BOOST - [11:10] */ +#define WM8993_DAC_BOOST_SHIFT 10 /* DAC_BOOST - [11:10] */ +#define WM8993_DAC_BOOST_WIDTH 2 /* DAC_BOOST - [11:10] */ +#define WM8993_DAC_COMP 0x0010 /* DAC_COMP */ +#define WM8993_DAC_COMP_MASK 0x0010 /* DAC_COMP */ +#define WM8993_DAC_COMP_SHIFT 4 /* DAC_COMP */ +#define WM8993_DAC_COMP_WIDTH 1 /* DAC_COMP */ +#define WM8993_DAC_COMPMODE 0x0008 /* DAC_COMPMODE */ +#define WM8993_DAC_COMPMODE_MASK 0x0008 /* DAC_COMPMODE */ +#define WM8993_DAC_COMPMODE_SHIFT 3 /* DAC_COMPMODE */ +#define WM8993_DAC_COMPMODE_WIDTH 1 /* DAC_COMPMODE */ +#define WM8993_ADC_COMP 0x0004 /* ADC_COMP */ +#define WM8993_ADC_COMP_MASK 0x0004 /* ADC_COMP */ +#define WM8993_ADC_COMP_SHIFT 2 /* ADC_COMP */ +#define WM8993_ADC_COMP_WIDTH 1 /* ADC_COMP */ +#define WM8993_ADC_COMPMODE 0x0002 /* ADC_COMPMODE */ +#define WM8993_ADC_COMPMODE_MASK 0x0002 /* ADC_COMPMODE */ +#define WM8993_ADC_COMPMODE_SHIFT 1 /* ADC_COMPMODE */ +#define WM8993_ADC_COMPMODE_WIDTH 1 /* ADC_COMPMODE */ +#define WM8993_LOOPBACK 0x0001 /* LOOPBACK */ +#define WM8993_LOOPBACK_MASK 0x0001 /* LOOPBACK */ +#define WM8993_LOOPBACK_SHIFT 0 /* LOOPBACK */ +#define WM8993_LOOPBACK_WIDTH 1 /* LOOPBACK */ + +/* + * R6 (0x06) - Clocking 1 + */ +#define WM8993_TOCLK_RATE 0x8000 /* TOCLK_RATE */ +#define WM8993_TOCLK_RATE_MASK 0x8000 /* TOCLK_RATE */ +#define WM8993_TOCLK_RATE_SHIFT 15 /* TOCLK_RATE */ +#define WM8993_TOCLK_RATE_WIDTH 1 /* TOCLK_RATE */ +#define WM8993_TOCLK_ENA 0x4000 /* TOCLK_ENA */ +#define WM8993_TOCLK_ENA_MASK 0x4000 /* TOCLK_ENA */ +#define WM8993_TOCLK_ENA_SHIFT 14 /* TOCLK_ENA */ +#define WM8993_TOCLK_ENA_WIDTH 1 /* TOCLK_ENA */ +#define WM8993_OPCLK_DIV_MASK 0x1E00 /* OPCLK_DIV - [12:9] */ +#define WM8993_OPCLK_DIV_SHIFT 9 /* OPCLK_DIV - [12:9] */ +#define WM8993_OPCLK_DIV_WIDTH 4 /* OPCLK_DIV - [12:9] */ +#define WM8993_DCLK_DIV_MASK 0x01C0 /* DCLK_DIV - [8:6] */ +#define WM8993_DCLK_DIV_SHIFT 6 /* DCLK_DIV - [8:6] */ +#define WM8993_DCLK_DIV_WIDTH 3 /* DCLK_DIV - [8:6] */ +#define WM8993_BCLK_DIV_MASK 0x001E /* BCLK_DIV - [4:1] */ +#define WM8993_BCLK_DIV_SHIFT 1 /* BCLK_DIV - [4:1] */ +#define WM8993_BCLK_DIV_WIDTH 4 /* BCLK_DIV - [4:1] */ + +/* + * R7 (0x07) - Clocking 2 + */ +#define WM8993_MCLK_SRC 0x8000 /* MCLK_SRC */ +#define WM8993_MCLK_SRC_MASK 0x8000 /* MCLK_SRC */ +#define WM8993_MCLK_SRC_SHIFT 15 /* MCLK_SRC */ +#define WM8993_MCLK_SRC_WIDTH 1 /* MCLK_SRC */ +#define WM8993_SYSCLK_SRC 0x4000 /* SYSCLK_SRC */ +#define WM8993_SYSCLK_SRC_MASK 0x4000 /* SYSCLK_SRC */ +#define WM8993_SYSCLK_SRC_SHIFT 14 /* SYSCLK_SRC */ +#define WM8993_SYSCLK_SRC_WIDTH 1 /* SYSCLK_SRC */ +#define WM8993_MCLK_DIV 0x1000 /* MCLK_DIV */ +#define WM8993_MCLK_DIV_MASK 0x1000 /* MCLK_DIV */ +#define WM8993_MCLK_DIV_SHIFT 12 /* MCLK_DIV */ +#define WM8993_MCLK_DIV_WIDTH 1 /* MCLK_DIV */ +#define WM8993_MCLK_INV 0x0400 /* MCLK_INV */ +#define WM8993_MCLK_INV_MASK 0x0400 /* MCLK_INV */ +#define WM8993_MCLK_INV_SHIFT 10 /* MCLK_INV */ +#define WM8993_MCLK_INV_WIDTH 1 /* MCLK_INV */ +#define WM8993_ADC_DIV_MASK 0x00E0 /* ADC_DIV - [7:5] */ +#define WM8993_ADC_DIV_SHIFT 5 /* ADC_DIV - [7:5] */ +#define WM8993_ADC_DIV_WIDTH 3 /* ADC_DIV - [7:5] */ +#define WM8993_DAC_DIV_MASK 0x001C /* DAC_DIV - [4:2] */ +#define WM8993_DAC_DIV_SHIFT 2 /* DAC_DIV - [4:2] */ +#define WM8993_DAC_DIV_WIDTH 3 /* DAC_DIV - [4:2] */ + +/* + * R8 (0x08) - Audio Interface (3) + */ +#define WM8993_AIF_MSTR1 0x8000 /* AIF_MSTR1 */ +#define WM8993_AIF_MSTR1_MASK 0x8000 /* AIF_MSTR1 */ +#define WM8993_AIF_MSTR1_SHIFT 15 /* AIF_MSTR1 */ +#define WM8993_AIF_MSTR1_WIDTH 1 /* AIF_MSTR1 */ + +/* + * R9 (0x09) - Audio Interface (4) + */ +#define WM8993_AIF_TRIS 0x2000 /* AIF_TRIS */ +#define WM8993_AIF_TRIS_MASK 0x2000 /* AIF_TRIS */ +#define WM8993_AIF_TRIS_SHIFT 13 /* AIF_TRIS */ +#define WM8993_AIF_TRIS_WIDTH 1 /* AIF_TRIS */ +#define WM8993_LRCLK_DIR 0x0800 /* LRCLK_DIR */ +#define WM8993_LRCLK_DIR_MASK 0x0800 /* LRCLK_DIR */ +#define WM8993_LRCLK_DIR_SHIFT 11 /* LRCLK_DIR */ +#define WM8993_LRCLK_DIR_WIDTH 1 /* LRCLK_DIR */ +#define WM8993_LRCLK_RATE_MASK 0x07FF /* LRCLK_RATE - [10:0] */ +#define WM8993_LRCLK_RATE_SHIFT 0 /* LRCLK_RATE - [10:0] */ +#define WM8993_LRCLK_RATE_WIDTH 11 /* LRCLK_RATE - [10:0] */ + +/* + * R10 (0x0A) - DAC CTRL + */ +#define WM8993_DAC_OSR128 0x2000 /* DAC_OSR128 */ +#define WM8993_DAC_OSR128_MASK 0x2000 /* DAC_OSR128 */ +#define WM8993_DAC_OSR128_SHIFT 13 /* DAC_OSR128 */ +#define WM8993_DAC_OSR128_WIDTH 1 /* DAC_OSR128 */ +#define WM8993_DAC_MONO 0x0200 /* DAC_MONO */ +#define WM8993_DAC_MONO_MASK 0x0200 /* DAC_MONO */ +#define WM8993_DAC_MONO_SHIFT 9 /* DAC_MONO */ +#define WM8993_DAC_MONO_WIDTH 1 /* DAC_MONO */ +#define WM8993_DAC_SB_FILT 0x0100 /* DAC_SB_FILT */ +#define WM8993_DAC_SB_FILT_MASK 0x0100 /* DAC_SB_FILT */ +#define WM8993_DAC_SB_FILT_SHIFT 8 /* DAC_SB_FILT */ +#define WM8993_DAC_SB_FILT_WIDTH 1 /* DAC_SB_FILT */ +#define WM8993_DAC_MUTERATE 0x0080 /* DAC_MUTERATE */ +#define WM8993_DAC_MUTERATE_MASK 0x0080 /* DAC_MUTERATE */ +#define WM8993_DAC_MUTERATE_SHIFT 7 /* DAC_MUTERATE */ +#define WM8993_DAC_MUTERATE_WIDTH 1 /* DAC_MUTERATE */ +#define WM8993_DAC_UNMUTE_RAMP 0x0040 /* DAC_UNMUTE_RAMP */ +#define WM8993_DAC_UNMUTE_RAMP_MASK 0x0040 /* DAC_UNMUTE_RAMP */ +#define WM8993_DAC_UNMUTE_RAMP_SHIFT 6 /* DAC_UNMUTE_RAMP */ +#define WM8993_DAC_UNMUTE_RAMP_WIDTH 1 /* DAC_UNMUTE_RAMP */ +#define WM8993_DEEMPH_MASK 0x0030 /* DEEMPH - [5:4] */ +#define WM8993_DEEMPH_SHIFT 4 /* DEEMPH - [5:4] */ +#define WM8993_DEEMPH_WIDTH 2 /* DEEMPH - [5:4] */ +#define WM8993_DAC_MUTE 0x0004 /* DAC_MUTE */ +#define WM8993_DAC_MUTE_MASK 0x0004 /* DAC_MUTE */ +#define WM8993_DAC_MUTE_SHIFT 2 /* DAC_MUTE */ +#define WM8993_DAC_MUTE_WIDTH 1 /* DAC_MUTE */ +#define WM8993_DACL_DATINV 0x0002 /* DACL_DATINV */ +#define WM8993_DACL_DATINV_MASK 0x0002 /* DACL_DATINV */ +#define WM8993_DACL_DATINV_SHIFT 1 /* DACL_DATINV */ +#define WM8993_DACL_DATINV_WIDTH 1 /* DACL_DATINV */ +#define WM8993_DACR_DATINV 0x0001 /* DACR_DATINV */ +#define WM8993_DACR_DATINV_MASK 0x0001 /* DACR_DATINV */ +#define WM8993_DACR_DATINV_SHIFT 0 /* DACR_DATINV */ +#define WM8993_DACR_DATINV_WIDTH 1 /* DACR_DATINV */ + +/* + * R11 (0x0B) - Left DAC Digital Volume + */ +#define WM8993_DAC_VU 0x0100 /* DAC_VU */ +#define WM8993_DAC_VU_MASK 0x0100 /* DAC_VU */ +#define WM8993_DAC_VU_SHIFT 8 /* DAC_VU */ +#define WM8993_DAC_VU_WIDTH 1 /* DAC_VU */ +#define WM8993_DACL_VOL_MASK 0x00FF /* DACL_VOL - [7:0] */ +#define WM8993_DACL_VOL_SHIFT 0 /* DACL_VOL - [7:0] */ +#define WM8993_DACL_VOL_WIDTH 8 /* DACL_VOL - [7:0] */ + +/* + * R12 (0x0C) - Right DAC Digital Volume + */ +#define WM8993_DAC_VU 0x0100 /* DAC_VU */ +#define WM8993_DAC_VU_MASK 0x0100 /* DAC_VU */ +#define WM8993_DAC_VU_SHIFT 8 /* DAC_VU */ +#define WM8993_DAC_VU_WIDTH 1 /* DAC_VU */ +#define WM8993_DACR_VOL_MASK 0x00FF /* DACR_VOL - [7:0] */ +#define WM8993_DACR_VOL_SHIFT 0 /* DACR_VOL - [7:0] */ +#define WM8993_DACR_VOL_WIDTH 8 /* DACR_VOL - [7:0] */ + +/* + * R13 (0x0D) - Digital Side Tone + */ +#define WM8993_ADCL_DAC_SVOL_MASK 0x1E00 /* ADCL_DAC_SVOL - [12:9] */ +#define WM8993_ADCL_DAC_SVOL_SHIFT 9 /* ADCL_DAC_SVOL - [12:9] */ +#define WM8993_ADCL_DAC_SVOL_WIDTH 4 /* ADCL_DAC_SVOL - [12:9] */ +#define WM8993_ADCR_DAC_SVOL_MASK 0x01E0 /* ADCR_DAC_SVOL - [8:5] */ +#define WM8993_ADCR_DAC_SVOL_SHIFT 5 /* ADCR_DAC_SVOL - [8:5] */ +#define WM8993_ADCR_DAC_SVOL_WIDTH 4 /* ADCR_DAC_SVOL - [8:5] */ +#define WM8993_ADC_TO_DACL_MASK 0x000C /* ADC_TO_DACL - [3:2] */ +#define WM8993_ADC_TO_DACL_SHIFT 2 /* ADC_TO_DACL - [3:2] */ +#define WM8993_ADC_TO_DACL_WIDTH 2 /* ADC_TO_DACL - [3:2] */ +#define WM8993_ADC_TO_DACR_MASK 0x0003 /* ADC_TO_DACR - [1:0] */ +#define WM8993_ADC_TO_DACR_SHIFT 0 /* ADC_TO_DACR - [1:0] */ +#define WM8993_ADC_TO_DACR_WIDTH 2 /* ADC_TO_DACR - [1:0] */ + +/* + * R14 (0x0E) - ADC CTRL + */ +#define WM8993_ADC_OSR128 0x0200 /* ADC_OSR128 */ +#define WM8993_ADC_OSR128_MASK 0x0200 /* ADC_OSR128 */ +#define WM8993_ADC_OSR128_SHIFT 9 /* ADC_OSR128 */ +#define WM8993_ADC_OSR128_WIDTH 1 /* ADC_OSR128 */ +#define WM8993_ADC_HPF 0x0100 /* ADC_HPF */ +#define WM8993_ADC_HPF_MASK 0x0100 /* ADC_HPF */ +#define WM8993_ADC_HPF_SHIFT 8 /* ADC_HPF */ +#define WM8993_ADC_HPF_WIDTH 1 /* ADC_HPF */ +#define WM8993_ADC_HPF_CUT_MASK 0x0060 /* ADC_HPF_CUT - [6:5] */ +#define WM8993_ADC_HPF_CUT_SHIFT 5 /* ADC_HPF_CUT - [6:5] */ +#define WM8993_ADC_HPF_CUT_WIDTH 2 /* ADC_HPF_CUT - [6:5] */ +#define WM8993_ADCL_DATINV 0x0002 /* ADCL_DATINV */ +#define WM8993_ADCL_DATINV_MASK 0x0002 /* ADCL_DATINV */ +#define WM8993_ADCL_DATINV_SHIFT 1 /* ADCL_DATINV */ +#define WM8993_ADCL_DATINV_WIDTH 1 /* ADCL_DATINV */ +#define WM8993_ADCR_DATINV 0x0001 /* ADCR_DATINV */ +#define WM8993_ADCR_DATINV_MASK 0x0001 /* ADCR_DATINV */ +#define WM8993_ADCR_DATINV_SHIFT 0 /* ADCR_DATINV */ +#define WM8993_ADCR_DATINV_WIDTH 1 /* ADCR_DATINV */ + +/* + * R15 (0x0F) - Left ADC Digital Volume + */ +#define WM8993_ADC_VU 0x0100 /* ADC_VU */ +#define WM8993_ADC_VU_MASK 0x0100 /* ADC_VU */ +#define WM8993_ADC_VU_SHIFT 8 /* ADC_VU */ +#define WM8993_ADC_VU_WIDTH 1 /* ADC_VU */ +#define WM8993_ADCL_VOL_MASK 0x00FF /* ADCL_VOL - [7:0] */ +#define WM8993_ADCL_VOL_SHIFT 0 /* ADCL_VOL - [7:0] */ +#define WM8993_ADCL_VOL_WIDTH 8 /* ADCL_VOL - [7:0] */ + +/* + * R16 (0x10) - Right ADC Digital Volume + */ +#define WM8993_ADC_VU 0x0100 /* ADC_VU */ +#define WM8993_ADC_VU_MASK 0x0100 /* ADC_VU */ +#define WM8993_ADC_VU_SHIFT 8 /* ADC_VU */ +#define WM8993_ADC_VU_WIDTH 1 /* ADC_VU */ +#define WM8993_ADCR_VOL_MASK 0x00FF /* ADCR_VOL - [7:0] */ +#define WM8993_ADCR_VOL_SHIFT 0 /* ADCR_VOL - [7:0] */ +#define WM8993_ADCR_VOL_WIDTH 8 /* ADCR_VOL - [7:0] */ + +/* + * R18 (0x12) - GPIO CTRL 1 + */ +#define WM8993_JD2_SC_EINT 0x8000 /* JD2_SC_EINT */ +#define WM8993_JD2_SC_EINT_MASK 0x8000 /* JD2_SC_EINT */ +#define WM8993_JD2_SC_EINT_SHIFT 15 /* JD2_SC_EINT */ +#define WM8993_JD2_SC_EINT_WIDTH 1 /* JD2_SC_EINT */ +#define WM8993_JD2_EINT 0x4000 /* JD2_EINT */ +#define WM8993_JD2_EINT_MASK 0x4000 /* JD2_EINT */ +#define WM8993_JD2_EINT_SHIFT 14 /* JD2_EINT */ +#define WM8993_JD2_EINT_WIDTH 1 /* JD2_EINT */ +#define WM8993_WSEQ_EINT 0x2000 /* WSEQ_EINT */ +#define WM8993_WSEQ_EINT_MASK 0x2000 /* WSEQ_EINT */ +#define WM8993_WSEQ_EINT_SHIFT 13 /* WSEQ_EINT */ +#define WM8993_WSEQ_EINT_WIDTH 1 /* WSEQ_EINT */ +#define WM8993_IRQ 0x1000 /* IRQ */ +#define WM8993_IRQ_MASK 0x1000 /* IRQ */ +#define WM8993_IRQ_SHIFT 12 /* IRQ */ +#define WM8993_IRQ_WIDTH 1 /* IRQ */ +#define WM8993_TEMPOK_EINT 0x0800 /* TEMPOK_EINT */ +#define WM8993_TEMPOK_EINT_MASK 0x0800 /* TEMPOK_EINT */ +#define WM8993_TEMPOK_EINT_SHIFT 11 /* TEMPOK_EINT */ +#define WM8993_TEMPOK_EINT_WIDTH 1 /* TEMPOK_EINT */ +#define WM8993_JD1_SC_EINT 0x0400 /* JD1_SC_EINT */ +#define WM8993_JD1_SC_EINT_MASK 0x0400 /* JD1_SC_EINT */ +#define WM8993_JD1_SC_EINT_SHIFT 10 /* JD1_SC_EINT */ +#define WM8993_JD1_SC_EINT_WIDTH 1 /* JD1_SC_EINT */ +#define WM8993_JD1_EINT 0x0200 /* JD1_EINT */ +#define WM8993_JD1_EINT_MASK 0x0200 /* JD1_EINT */ +#define WM8993_JD1_EINT_SHIFT 9 /* JD1_EINT */ +#define WM8993_JD1_EINT_WIDTH 1 /* JD1_EINT */ +#define WM8993_FLL_LOCK_EINT 0x0100 /* FLL_LOCK_EINT */ +#define WM8993_FLL_LOCK_EINT_MASK 0x0100 /* FLL_LOCK_EINT */ +#define WM8993_FLL_LOCK_EINT_SHIFT 8 /* FLL_LOCK_EINT */ +#define WM8993_FLL_LOCK_EINT_WIDTH 1 /* FLL_LOCK_EINT */ +#define WM8993_GPI8_EINT 0x0080 /* GPI8_EINT */ +#define WM8993_GPI8_EINT_MASK 0x0080 /* GPI8_EINT */ +#define WM8993_GPI8_EINT_SHIFT 7 /* GPI8_EINT */ +#define WM8993_GPI8_EINT_WIDTH 1 /* GPI8_EINT */ +#define WM8993_GPI7_EINT 0x0040 /* GPI7_EINT */ +#define WM8993_GPI7_EINT_MASK 0x0040 /* GPI7_EINT */ +#define WM8993_GPI7_EINT_SHIFT 6 /* GPI7_EINT */ +#define WM8993_GPI7_EINT_WIDTH 1 /* GPI7_EINT */ +#define WM8993_GPIO1_EINT 0x0001 /* GPIO1_EINT */ +#define WM8993_GPIO1_EINT_MASK 0x0001 /* GPIO1_EINT */ +#define WM8993_GPIO1_EINT_SHIFT 0 /* GPIO1_EINT */ +#define WM8993_GPIO1_EINT_WIDTH 1 /* GPIO1_EINT */ + +/* + * R19 (0x13) - GPIO1 + */ +#define WM8993_GPIO1_PU 0x0020 /* GPIO1_PU */ +#define WM8993_GPIO1_PU_MASK 0x0020 /* GPIO1_PU */ +#define WM8993_GPIO1_PU_SHIFT 5 /* GPIO1_PU */ +#define WM8993_GPIO1_PU_WIDTH 1 /* GPIO1_PU */ +#define WM8993_GPIO1_PD 0x0010 /* GPIO1_PD */ +#define WM8993_GPIO1_PD_MASK 0x0010 /* GPIO1_PD */ +#define WM8993_GPIO1_PD_SHIFT 4 /* GPIO1_PD */ +#define WM8993_GPIO1_PD_WIDTH 1 /* GPIO1_PD */ +#define WM8993_GPIO1_SEL_MASK 0x000F /* GPIO1_SEL - [3:0] */ +#define WM8993_GPIO1_SEL_SHIFT 0 /* GPIO1_SEL - [3:0] */ +#define WM8993_GPIO1_SEL_WIDTH 4 /* GPIO1_SEL - [3:0] */ + +/* + * R20 (0x14) - IRQ_DEBOUNCE + */ +#define WM8993_JD2_SC_DB 0x8000 /* JD2_SC_DB */ +#define WM8993_JD2_SC_DB_MASK 0x8000 /* JD2_SC_DB */ +#define WM8993_JD2_SC_DB_SHIFT 15 /* JD2_SC_DB */ +#define WM8993_JD2_SC_DB_WIDTH 1 /* JD2_SC_DB */ +#define WM8993_JD2_DB 0x4000 /* JD2_DB */ +#define WM8993_JD2_DB_MASK 0x4000 /* JD2_DB */ +#define WM8993_JD2_DB_SHIFT 14 /* JD2_DB */ +#define WM8993_JD2_DB_WIDTH 1 /* JD2_DB */ +#define WM8993_WSEQ_DB 0x2000 /* WSEQ_DB */ +#define WM8993_WSEQ_DB_MASK 0x2000 /* WSEQ_DB */ +#define WM8993_WSEQ_DB_SHIFT 13 /* WSEQ_DB */ +#define WM8993_WSEQ_DB_WIDTH 1 /* WSEQ_DB */ +#define WM8993_TEMPOK_DB 0x0800 /* TEMPOK_DB */ +#define WM8993_TEMPOK_DB_MASK 0x0800 /* TEMPOK_DB */ +#define WM8993_TEMPOK_DB_SHIFT 11 /* TEMPOK_DB */ +#define WM8993_TEMPOK_DB_WIDTH 1 /* TEMPOK_DB */ +#define WM8993_JD1_SC_DB 0x0400 /* JD1_SC_DB */ +#define WM8993_JD1_SC_DB_MASK 0x0400 /* JD1_SC_DB */ +#define WM8993_JD1_SC_DB_SHIFT 10 /* JD1_SC_DB */ +#define WM8993_JD1_SC_DB_WIDTH 1 /* JD1_SC_DB */ +#define WM8993_JD1_DB 0x0200 /* JD1_DB */ +#define WM8993_JD1_DB_MASK 0x0200 /* JD1_DB */ +#define WM8993_JD1_DB_SHIFT 9 /* JD1_DB */ +#define WM8993_JD1_DB_WIDTH 1 /* JD1_DB */ +#define WM8993_FLL_LOCK_DB 0x0100 /* FLL_LOCK_DB */ +#define WM8993_FLL_LOCK_DB_MASK 0x0100 /* FLL_LOCK_DB */ +#define WM8993_FLL_LOCK_DB_SHIFT 8 /* FLL_LOCK_DB */ +#define WM8993_FLL_LOCK_DB_WIDTH 1 /* FLL_LOCK_DB */ +#define WM8993_GPI8_DB 0x0080 /* GPI8_DB */ +#define WM8993_GPI8_DB_MASK 0x0080 /* GPI8_DB */ +#define WM8993_GPI8_DB_SHIFT 7 /* GPI8_DB */ +#define WM8993_GPI8_DB_WIDTH 1 /* GPI8_DB */ +#define WM8993_GPI7_DB 0x0008 /* GPI7_DB */ +#define WM8993_GPI7_DB_MASK 0x0008 /* GPI7_DB */ +#define WM8993_GPI7_DB_SHIFT 3 /* GPI7_DB */ +#define WM8993_GPI7_DB_WIDTH 1 /* GPI7_DB */ +#define WM8993_GPIO1_DB 0x0001 /* GPIO1_DB */ +#define WM8993_GPIO1_DB_MASK 0x0001 /* GPIO1_DB */ +#define WM8993_GPIO1_DB_SHIFT 0 /* GPIO1_DB */ +#define WM8993_GPIO1_DB_WIDTH 1 /* GPIO1_DB */ + +/* + * R22 (0x16) - GPIOCTRL 2 + */ +#define WM8993_IM_JD2_EINT 0x2000 /* IM_JD2_EINT */ +#define WM8993_IM_JD2_EINT_MASK 0x2000 /* IM_JD2_EINT */ +#define WM8993_IM_JD2_EINT_SHIFT 13 /* IM_JD2_EINT */ +#define WM8993_IM_JD2_EINT_WIDTH 1 /* IM_JD2_EINT */ +#define WM8993_IM_JD2_SC_EINT 0x1000 /* IM_JD2_SC_EINT */ +#define WM8993_IM_JD2_SC_EINT_MASK 0x1000 /* IM_JD2_SC_EINT */ +#define WM8993_IM_JD2_SC_EINT_SHIFT 12 /* IM_JD2_SC_EINT */ +#define WM8993_IM_JD2_SC_EINT_WIDTH 1 /* IM_JD2_SC_EINT */ +#define WM8993_IM_TEMPOK_EINT 0x0800 /* IM_TEMPOK_EINT */ +#define WM8993_IM_TEMPOK_EINT_MASK 0x0800 /* IM_TEMPOK_EINT */ +#define WM8993_IM_TEMPOK_EINT_SHIFT 11 /* IM_TEMPOK_EINT */ +#define WM8993_IM_TEMPOK_EINT_WIDTH 1 /* IM_TEMPOK_EINT */ +#define WM8993_IM_JD1_SC_EINT 0x0400 /* IM_JD1_SC_EINT */ +#define WM8993_IM_JD1_SC_EINT_MASK 0x0400 /* IM_JD1_SC_EINT */ +#define WM8993_IM_JD1_SC_EINT_SHIFT 10 /* IM_JD1_SC_EINT */ +#define WM8993_IM_JD1_SC_EINT_WIDTH 1 /* IM_JD1_SC_EINT */ +#define WM8993_IM_JD1_EINT 0x0200 /* IM_JD1_EINT */ +#define WM8993_IM_JD1_EINT_MASK 0x0200 /* IM_JD1_EINT */ +#define WM8993_IM_JD1_EINT_SHIFT 9 /* IM_JD1_EINT */ +#define WM8993_IM_JD1_EINT_WIDTH 1 /* IM_JD1_EINT */ +#define WM8993_IM_FLL_LOCK_EINT 0x0100 /* IM_FLL_LOCK_EINT */ +#define WM8993_IM_FLL_LOCK_EINT_MASK 0x0100 /* IM_FLL_LOCK_EINT */ +#define WM8993_IM_FLL_LOCK_EINT_SHIFT 8 /* IM_FLL_LOCK_EINT */ +#define WM8993_IM_FLL_LOCK_EINT_WIDTH 1 /* IM_FLL_LOCK_EINT */ +#define WM8993_IM_GPI8_EINT 0x0040 /* IM_GPI8_EINT */ +#define WM8993_IM_GPI8_EINT_MASK 0x0040 /* IM_GPI8_EINT */ +#define WM8993_IM_GPI8_EINT_SHIFT 6 /* IM_GPI8_EINT */ +#define WM8993_IM_GPI8_EINT_WIDTH 1 /* IM_GPI8_EINT */ +#define WM8993_IM_GPIO1_EINT 0x0020 /* IM_GPIO1_EINT */ +#define WM8993_IM_GPIO1_EINT_MASK 0x0020 /* IM_GPIO1_EINT */ +#define WM8993_IM_GPIO1_EINT_SHIFT 5 /* IM_GPIO1_EINT */ +#define WM8993_IM_GPIO1_EINT_WIDTH 1 /* IM_GPIO1_EINT */ +#define WM8993_GPI8_ENA 0x0010 /* GPI8_ENA */ +#define WM8993_GPI8_ENA_MASK 0x0010 /* GPI8_ENA */ +#define WM8993_GPI8_ENA_SHIFT 4 /* GPI8_ENA */ +#define WM8993_GPI8_ENA_WIDTH 1 /* GPI8_ENA */ +#define WM8993_IM_GPI7_EINT 0x0004 /* IM_GPI7_EINT */ +#define WM8993_IM_GPI7_EINT_MASK 0x0004 /* IM_GPI7_EINT */ +#define WM8993_IM_GPI7_EINT_SHIFT 2 /* IM_GPI7_EINT */ +#define WM8993_IM_GPI7_EINT_WIDTH 1 /* IM_GPI7_EINT */ +#define WM8993_IM_WSEQ_EINT 0x0002 /* IM_WSEQ_EINT */ +#define WM8993_IM_WSEQ_EINT_MASK 0x0002 /* IM_WSEQ_EINT */ +#define WM8993_IM_WSEQ_EINT_SHIFT 1 /* IM_WSEQ_EINT */ +#define WM8993_IM_WSEQ_EINT_WIDTH 1 /* IM_WSEQ_EINT */ +#define WM8993_GPI7_ENA 0x0001 /* GPI7_ENA */ +#define WM8993_GPI7_ENA_MASK 0x0001 /* GPI7_ENA */ +#define WM8993_GPI7_ENA_SHIFT 0 /* GPI7_ENA */ +#define WM8993_GPI7_ENA_WIDTH 1 /* GPI7_ENA */ + +/* + * R23 (0x17) - GPIO_POL + */ +#define WM8993_JD2_SC_POL 0x8000 /* JD2_SC_POL */ +#define WM8993_JD2_SC_POL_MASK 0x8000 /* JD2_SC_POL */ +#define WM8993_JD2_SC_POL_SHIFT 15 /* JD2_SC_POL */ +#define WM8993_JD2_SC_POL_WIDTH 1 /* JD2_SC_POL */ +#define WM8993_JD2_POL 0x4000 /* JD2_POL */ +#define WM8993_JD2_POL_MASK 0x4000 /* JD2_POL */ +#define WM8993_JD2_POL_SHIFT 14 /* JD2_POL */ +#define WM8993_JD2_POL_WIDTH 1 /* JD2_POL */ +#define WM8993_WSEQ_POL 0x2000 /* WSEQ_POL */ +#define WM8993_WSEQ_POL_MASK 0x2000 /* WSEQ_POL */ +#define WM8993_WSEQ_POL_SHIFT 13 /* WSEQ_POL */ +#define WM8993_WSEQ_POL_WIDTH 1 /* WSEQ_POL */ +#define WM8993_IRQ_POL 0x1000 /* IRQ_POL */ +#define WM8993_IRQ_POL_MASK 0x1000 /* IRQ_POL */ +#define WM8993_IRQ_POL_SHIFT 12 /* IRQ_POL */ +#define WM8993_IRQ_POL_WIDTH 1 /* IRQ_POL */ +#define WM8993_TEMPOK_POL 0x0800 /* TEMPOK_POL */ +#define WM8993_TEMPOK_POL_MASK 0x0800 /* TEMPOK_POL */ +#define WM8993_TEMPOK_POL_SHIFT 11 /* TEMPOK_POL */ +#define WM8993_TEMPOK_POL_WIDTH 1 /* TEMPOK_POL */ +#define WM8993_JD1_SC_POL 0x0400 /* JD1_SC_POL */ +#define WM8993_JD1_SC_POL_MASK 0x0400 /* JD1_SC_POL */ +#define WM8993_JD1_SC_POL_SHIFT 10 /* JD1_SC_POL */ +#define WM8993_JD1_SC_POL_WIDTH 1 /* JD1_SC_POL */ +#define WM8993_JD1_POL 0x0200 /* JD1_POL */ +#define WM8993_JD1_POL_MASK 0x0200 /* JD1_POL */ +#define WM8993_JD1_POL_SHIFT 9 /* JD1_POL */ +#define WM8993_JD1_POL_WIDTH 1 /* JD1_POL */ +#define WM8993_FLL_LOCK_POL 0x0100 /* FLL_LOCK_POL */ +#define WM8993_FLL_LOCK_POL_MASK 0x0100 /* FLL_LOCK_POL */ +#define WM8993_FLL_LOCK_POL_SHIFT 8 /* FLL_LOCK_POL */ +#define WM8993_FLL_LOCK_POL_WIDTH 1 /* FLL_LOCK_POL */ +#define WM8993_GPI8_POL 0x0080 /* GPI8_POL */ +#define WM8993_GPI8_POL_MASK 0x0080 /* GPI8_POL */ +#define WM8993_GPI8_POL_SHIFT 7 /* GPI8_POL */ +#define WM8993_GPI8_POL_WIDTH 1 /* GPI8_POL */ +#define WM8993_GPI7_POL 0x0040 /* GPI7_POL */ +#define WM8993_GPI7_POL_MASK 0x0040 /* GPI7_POL */ +#define WM8993_GPI7_POL_SHIFT 6 /* GPI7_POL */ +#define WM8993_GPI7_POL_WIDTH 1 /* GPI7_POL */ +#define WM8993_GPIO1_POL 0x0001 /* GPIO1_POL */ +#define WM8993_GPIO1_POL_MASK 0x0001 /* GPIO1_POL */ +#define WM8993_GPIO1_POL_SHIFT 0 /* GPIO1_POL */ +#define WM8993_GPIO1_POL_WIDTH 1 /* GPIO1_POL */ + +/* + * R24 (0x18) - Left Line Input 1&2 Volume + */ +#define WM8993_IN1_VU 0x0100 /* IN1_VU */ +#define WM8993_IN1_VU_MASK 0x0100 /* IN1_VU */ +#define WM8993_IN1_VU_SHIFT 8 /* IN1_VU */ +#define WM8993_IN1_VU_WIDTH 1 /* IN1_VU */ +#define WM8993_IN1L_MUTE 0x0080 /* IN1L_MUTE */ +#define WM8993_IN1L_MUTE_MASK 0x0080 /* IN1L_MUTE */ +#define WM8993_IN1L_MUTE_SHIFT 7 /* IN1L_MUTE */ +#define WM8993_IN1L_MUTE_WIDTH 1 /* IN1L_MUTE */ +#define WM8993_IN1L_ZC 0x0040 /* IN1L_ZC */ +#define WM8993_IN1L_ZC_MASK 0x0040 /* IN1L_ZC */ +#define WM8993_IN1L_ZC_SHIFT 6 /* IN1L_ZC */ +#define WM8993_IN1L_ZC_WIDTH 1 /* IN1L_ZC */ +#define WM8993_IN1L_VOL_MASK 0x001F /* IN1L_VOL - [4:0] */ +#define WM8993_IN1L_VOL_SHIFT 0 /* IN1L_VOL - [4:0] */ +#define WM8993_IN1L_VOL_WIDTH 5 /* IN1L_VOL - [4:0] */ + +/* + * R25 (0x19) - Left Line Input 3&4 Volume + */ +#define WM8993_IN2_VU 0x0100 /* IN2_VU */ +#define WM8993_IN2_VU_MASK 0x0100 /* IN2_VU */ +#define WM8993_IN2_VU_SHIFT 8 /* IN2_VU */ +#define WM8993_IN2_VU_WIDTH 1 /* IN2_VU */ +#define WM8993_IN2L_MUTE 0x0080 /* IN2L_MUTE */ +#define WM8993_IN2L_MUTE_MASK 0x0080 /* IN2L_MUTE */ +#define WM8993_IN2L_MUTE_SHIFT 7 /* IN2L_MUTE */ +#define WM8993_IN2L_MUTE_WIDTH 1 /* IN2L_MUTE */ +#define WM8993_IN2L_ZC 0x0040 /* IN2L_ZC */ +#define WM8993_IN2L_ZC_MASK 0x0040 /* IN2L_ZC */ +#define WM8993_IN2L_ZC_SHIFT 6 /* IN2L_ZC */ +#define WM8993_IN2L_ZC_WIDTH 1 /* IN2L_ZC */ +#define WM8993_IN2L_VOL_MASK 0x001F /* IN2L_VOL - [4:0] */ +#define WM8993_IN2L_VOL_SHIFT 0 /* IN2L_VOL - [4:0] */ +#define WM8993_IN2L_VOL_WIDTH 5 /* IN2L_VOL - [4:0] */ + +/* + * R26 (0x1A) - Right Line Input 1&2 Volume + */ +#define WM8993_IN1_VU 0x0100 /* IN1_VU */ +#define WM8993_IN1_VU_MASK 0x0100 /* IN1_VU */ +#define WM8993_IN1_VU_SHIFT 8 /* IN1_VU */ +#define WM8993_IN1_VU_WIDTH 1 /* IN1_VU */ +#define WM8993_IN1R_MUTE 0x0080 /* IN1R_MUTE */ +#define WM8993_IN1R_MUTE_MASK 0x0080 /* IN1R_MUTE */ +#define WM8993_IN1R_MUTE_SHIFT 7 /* IN1R_MUTE */ +#define WM8993_IN1R_MUTE_WIDTH 1 /* IN1R_MUTE */ +#define WM8993_IN1R_ZC 0x0040 /* IN1R_ZC */ +#define WM8993_IN1R_ZC_MASK 0x0040 /* IN1R_ZC */ +#define WM8993_IN1R_ZC_SHIFT 6 /* IN1R_ZC */ +#define WM8993_IN1R_ZC_WIDTH 1 /* IN1R_ZC */ +#define WM8993_IN1R_VOL_MASK 0x001F /* IN1R_VOL - [4:0] */ +#define WM8993_IN1R_VOL_SHIFT 0 /* IN1R_VOL - [4:0] */ +#define WM8993_IN1R_VOL_WIDTH 5 /* IN1R_VOL - [4:0] */ + +/* + * R27 (0x1B) - Right Line Input 3&4 Volume + */ +#define WM8993_IN2_VU 0x0100 /* IN2_VU */ +#define WM8993_IN2_VU_MASK 0x0100 /* IN2_VU */ +#define WM8993_IN2_VU_SHIFT 8 /* IN2_VU */ +#define WM8993_IN2_VU_WIDTH 1 /* IN2_VU */ +#define WM8993_IN2R_MUTE 0x0080 /* IN2R_MUTE */ +#define WM8993_IN2R_MUTE_MASK 0x0080 /* IN2R_MUTE */ +#define WM8993_IN2R_MUTE_SHIFT 7 /* IN2R_MUTE */ +#define WM8993_IN2R_MUTE_WIDTH 1 /* IN2R_MUTE */ +#define WM8993_IN2R_ZC 0x0040 /* IN2R_ZC */ +#define WM8993_IN2R_ZC_MASK 0x0040 /* IN2R_ZC */ +#define WM8993_IN2R_ZC_SHIFT 6 /* IN2R_ZC */ +#define WM8993_IN2R_ZC_WIDTH 1 /* IN2R_ZC */ +#define WM8993_IN2R_VOL_MASK 0x001F /* IN2R_VOL - [4:0] */ +#define WM8993_IN2R_VOL_SHIFT 0 /* IN2R_VOL - [4:0] */ +#define WM8993_IN2R_VOL_WIDTH 5 /* IN2R_VOL - [4:0] */ + +/* + * R28 (0x1C) - Left Output Volume + */ +#define WM8993_HPOUT1_VU 0x0100 /* HPOUT1_VU */ +#define WM8993_HPOUT1_VU_MASK 0x0100 /* HPOUT1_VU */ +#define WM8993_HPOUT1_VU_SHIFT 8 /* HPOUT1_VU */ +#define WM8993_HPOUT1_VU_WIDTH 1 /* HPOUT1_VU */ +#define WM8993_HPOUT1L_ZC 0x0080 /* HPOUT1L_ZC */ +#define WM8993_HPOUT1L_ZC_MASK 0x0080 /* HPOUT1L_ZC */ +#define WM8993_HPOUT1L_ZC_SHIFT 7 /* HPOUT1L_ZC */ +#define WM8993_HPOUT1L_ZC_WIDTH 1 /* HPOUT1L_ZC */ +#define WM8993_HPOUT1L_MUTE_N 0x0040 /* HPOUT1L_MUTE_N */ +#define WM8993_HPOUT1L_MUTE_N_MASK 0x0040 /* HPOUT1L_MUTE_N */ +#define WM8993_HPOUT1L_MUTE_N_SHIFT 6 /* HPOUT1L_MUTE_N */ +#define WM8993_HPOUT1L_MUTE_N_WIDTH 1 /* HPOUT1L_MUTE_N */ +#define WM8993_HPOUT1L_VOL_MASK 0x003F /* HPOUT1L_VOL - [5:0] */ +#define WM8993_HPOUT1L_VOL_SHIFT 0 /* HPOUT1L_VOL - [5:0] */ +#define WM8993_HPOUT1L_VOL_WIDTH 6 /* HPOUT1L_VOL - [5:0] */ + +/* + * R29 (0x1D) - Right Output Volume + */ +#define WM8993_HPOUT1_VU 0x0100 /* HPOUT1_VU */ +#define WM8993_HPOUT1_VU_MASK 0x0100 /* HPOUT1_VU */ +#define WM8993_HPOUT1_VU_SHIFT 8 /* HPOUT1_VU */ +#define WM8993_HPOUT1_VU_WIDTH 1 /* HPOUT1_VU */ +#define WM8993_HPOUT1R_ZC 0x0080 /* HPOUT1R_ZC */ +#define WM8993_HPOUT1R_ZC_MASK 0x0080 /* HPOUT1R_ZC */ +#define WM8993_HPOUT1R_ZC_SHIFT 7 /* HPOUT1R_ZC */ +#define WM8993_HPOUT1R_ZC_WIDTH 1 /* HPOUT1R_ZC */ +#define WM8993_HPOUT1R_MUTE_N 0x0040 /* HPOUT1R_MUTE_N */ +#define WM8993_HPOUT1R_MUTE_N_MASK 0x0040 /* HPOUT1R_MUTE_N */ +#define WM8993_HPOUT1R_MUTE_N_SHIFT 6 /* HPOUT1R_MUTE_N */ +#define WM8993_HPOUT1R_MUTE_N_WIDTH 1 /* HPOUT1R_MUTE_N */ +#define WM8993_HPOUT1R_VOL_MASK 0x003F /* HPOUT1R_VOL - [5:0] */ +#define WM8993_HPOUT1R_VOL_SHIFT 0 /* HPOUT1R_VOL - [5:0] */ +#define WM8993_HPOUT1R_VOL_WIDTH 6 /* HPOUT1R_VOL - [5:0] */ + +/* + * R30 (0x1E) - Line Outputs Volume + */ +#define WM8993_LINEOUT1N_MUTE 0x0040 /* LINEOUT1N_MUTE */ +#define WM8993_LINEOUT1N_MUTE_MASK 0x0040 /* LINEOUT1N_MUTE */ +#define WM8993_LINEOUT1N_MUTE_SHIFT 6 /* LINEOUT1N_MUTE */ +#define WM8993_LINEOUT1N_MUTE_WIDTH 1 /* LINEOUT1N_MUTE */ +#define WM8993_LINEOUT1P_MUTE 0x0020 /* LINEOUT1P_MUTE */ +#define WM8993_LINEOUT1P_MUTE_MASK 0x0020 /* LINEOUT1P_MUTE */ +#define WM8993_LINEOUT1P_MUTE_SHIFT 5 /* LINEOUT1P_MUTE */ +#define WM8993_LINEOUT1P_MUTE_WIDTH 1 /* LINEOUT1P_MUTE */ +#define WM8993_LINEOUT1_VOL 0x0010 /* LINEOUT1_VOL */ +#define WM8993_LINEOUT1_VOL_MASK 0x0010 /* LINEOUT1_VOL */ +#define WM8993_LINEOUT1_VOL_SHIFT 4 /* LINEOUT1_VOL */ +#define WM8993_LINEOUT1_VOL_WIDTH 1 /* LINEOUT1_VOL */ +#define WM8993_LINEOUT2N_MUTE 0x0004 /* LINEOUT2N_MUTE */ +#define WM8993_LINEOUT2N_MUTE_MASK 0x0004 /* LINEOUT2N_MUTE */ +#define WM8993_LINEOUT2N_MUTE_SHIFT 2 /* LINEOUT2N_MUTE */ +#define WM8993_LINEOUT2N_MUTE_WIDTH 1 /* LINEOUT2N_MUTE */ +#define WM8993_LINEOUT2P_MUTE 0x0002 /* LINEOUT2P_MUTE */ +#define WM8993_LINEOUT2P_MUTE_MASK 0x0002 /* LINEOUT2P_MUTE */ +#define WM8993_LINEOUT2P_MUTE_SHIFT 1 /* LINEOUT2P_MUTE */ +#define WM8993_LINEOUT2P_MUTE_WIDTH 1 /* LINEOUT2P_MUTE */ +#define WM8993_LINEOUT2_VOL 0x0001 /* LINEOUT2_VOL */ +#define WM8993_LINEOUT2_VOL_MASK 0x0001 /* LINEOUT2_VOL */ +#define WM8993_LINEOUT2_VOL_SHIFT 0 /* LINEOUT2_VOL */ +#define WM8993_LINEOUT2_VOL_WIDTH 1 /* LINEOUT2_VOL */ + +/* + * R31 (0x1F) - HPOUT2 Volume + */ +#define WM8993_HPOUT2_MUTE 0x0020 /* HPOUT2_MUTE */ +#define WM8993_HPOUT2_MUTE_MASK 0x0020 /* HPOUT2_MUTE */ +#define WM8993_HPOUT2_MUTE_SHIFT 5 /* HPOUT2_MUTE */ +#define WM8993_HPOUT2_MUTE_WIDTH 1 /* HPOUT2_MUTE */ +#define WM8993_HPOUT2_VOL 0x0010 /* HPOUT2_VOL */ +#define WM8993_HPOUT2_VOL_MASK 0x0010 /* HPOUT2_VOL */ +#define WM8993_HPOUT2_VOL_SHIFT 4 /* HPOUT2_VOL */ +#define WM8993_HPOUT2_VOL_WIDTH 1 /* HPOUT2_VOL */ + +/* + * R32 (0x20) - Left OPGA Volume + */ +#define WM8993_MIXOUT_VU 0x0100 /* MIXOUT_VU */ +#define WM8993_MIXOUT_VU_MASK 0x0100 /* MIXOUT_VU */ +#define WM8993_MIXOUT_VU_SHIFT 8 /* MIXOUT_VU */ +#define WM8993_MIXOUT_VU_WIDTH 1 /* MIXOUT_VU */ +#define WM8993_MIXOUTL_ZC 0x0080 /* MIXOUTL_ZC */ +#define WM8993_MIXOUTL_ZC_MASK 0x0080 /* MIXOUTL_ZC */ +#define WM8993_MIXOUTL_ZC_SHIFT 7 /* MIXOUTL_ZC */ +#define WM8993_MIXOUTL_ZC_WIDTH 1 /* MIXOUTL_ZC */ +#define WM8993_MIXOUTL_MUTE_N 0x0040 /* MIXOUTL_MUTE_N */ +#define WM8993_MIXOUTL_MUTE_N_MASK 0x0040 /* MIXOUTL_MUTE_N */ +#define WM8993_MIXOUTL_MUTE_N_SHIFT 6 /* MIXOUTL_MUTE_N */ +#define WM8993_MIXOUTL_MUTE_N_WIDTH 1 /* MIXOUTL_MUTE_N */ +#define WM8993_MIXOUTL_VOL_MASK 0x003F /* MIXOUTL_VOL - [5:0] */ +#define WM8993_MIXOUTL_VOL_SHIFT 0 /* MIXOUTL_VOL - [5:0] */ +#define WM8993_MIXOUTL_VOL_WIDTH 6 /* MIXOUTL_VOL - [5:0] */ + +/* + * R33 (0x21) - Right OPGA Volume + */ +#define WM8993_MIXOUT_VU 0x0100 /* MIXOUT_VU */ +#define WM8993_MIXOUT_VU_MASK 0x0100 /* MIXOUT_VU */ +#define WM8993_MIXOUT_VU_SHIFT 8 /* MIXOUT_VU */ +#define WM8993_MIXOUT_VU_WIDTH 1 /* MIXOUT_VU */ +#define WM8993_MIXOUTR_ZC 0x0080 /* MIXOUTR_ZC */ +#define WM8993_MIXOUTR_ZC_MASK 0x0080 /* MIXOUTR_ZC */ +#define WM8993_MIXOUTR_ZC_SHIFT 7 /* MIXOUTR_ZC */ +#define WM8993_MIXOUTR_ZC_WIDTH 1 /* MIXOUTR_ZC */ +#define WM8993_MIXOUTR_MUTE_N 0x0040 /* MIXOUTR_MUTE_N */ +#define WM8993_MIXOUTR_MUTE_N_MASK 0x0040 /* MIXOUTR_MUTE_N */ +#define WM8993_MIXOUTR_MUTE_N_SHIFT 6 /* MIXOUTR_MUTE_N */ +#define WM8993_MIXOUTR_MUTE_N_WIDTH 1 /* MIXOUTR_MUTE_N */ +#define WM8993_MIXOUTR_VOL_MASK 0x003F /* MIXOUTR_VOL - [5:0] */ +#define WM8993_MIXOUTR_VOL_SHIFT 0 /* MIXOUTR_VOL - [5:0] */ +#define WM8993_MIXOUTR_VOL_WIDTH 6 /* MIXOUTR_VOL - [5:0] */ + +/* + * R34 (0x22) - SPKMIXL Attenuation + */ +#define WM8993_MIXINL_SPKMIXL_VOL 0x0020 /* MIXINL_SPKMIXL_VOL */ +#define WM8993_MIXINL_SPKMIXL_VOL_MASK 0x0020 /* MIXINL_SPKMIXL_VOL */ +#define WM8993_MIXINL_SPKMIXL_VOL_SHIFT 5 /* MIXINL_SPKMIXL_VOL */ +#define WM8993_MIXINL_SPKMIXL_VOL_WIDTH 1 /* MIXINL_SPKMIXL_VOL */ +#define WM8993_IN1LP_SPKMIXL_VOL 0x0010 /* IN1LP_SPKMIXL_VOL */ +#define WM8993_IN1LP_SPKMIXL_VOL_MASK 0x0010 /* IN1LP_SPKMIXL_VOL */ +#define WM8993_IN1LP_SPKMIXL_VOL_SHIFT 4 /* IN1LP_SPKMIXL_VOL */ +#define WM8993_IN1LP_SPKMIXL_VOL_WIDTH 1 /* IN1LP_SPKMIXL_VOL */ +#define WM8993_MIXOUTL_SPKMIXL_VOL 0x0008 /* MIXOUTL_SPKMIXL_VOL */ +#define WM8993_MIXOUTL_SPKMIXL_VOL_MASK 0x0008 /* MIXOUTL_SPKMIXL_VOL */ +#define WM8993_MIXOUTL_SPKMIXL_VOL_SHIFT 3 /* MIXOUTL_SPKMIXL_VOL */ +#define WM8993_MIXOUTL_SPKMIXL_VOL_WIDTH 1 /* MIXOUTL_SPKMIXL_VOL */ +#define WM8993_DACL_SPKMIXL_VOL 0x0004 /* DACL_SPKMIXL_VOL */ +#define WM8993_DACL_SPKMIXL_VOL_MASK 0x0004 /* DACL_SPKMIXL_VOL */ +#define WM8993_DACL_SPKMIXL_VOL_SHIFT 2 /* DACL_SPKMIXL_VOL */ +#define WM8993_DACL_SPKMIXL_VOL_WIDTH 1 /* DACL_SPKMIXL_VOL */ +#define WM8993_SPKMIXL_VOL_MASK 0x0003 /* SPKMIXL_VOL - [1:0] */ +#define WM8993_SPKMIXL_VOL_SHIFT 0 /* SPKMIXL_VOL - [1:0] */ +#define WM8993_SPKMIXL_VOL_WIDTH 2 /* SPKMIXL_VOL - [1:0] */ + +/* + * R35 (0x23) - SPKMIXR Attenuation + */ +#define WM8993_SPKOUT_CLASSAB_MODE 0x0100 /* SPKOUT_CLASSAB_MODE */ +#define WM8993_SPKOUT_CLASSAB_MODE_MASK 0x0100 /* SPKOUT_CLASSAB_MODE */ +#define WM8993_SPKOUT_CLASSAB_MODE_SHIFT 8 /* SPKOUT_CLASSAB_MODE */ +#define WM8993_SPKOUT_CLASSAB_MODE_WIDTH 1 /* SPKOUT_CLASSAB_MODE */ +#define WM8993_MIXINR_SPKMIXR_VOL 0x0020 /* MIXINR_SPKMIXR_VOL */ +#define WM8993_MIXINR_SPKMIXR_VOL_MASK 0x0020 /* MIXINR_SPKMIXR_VOL */ +#define WM8993_MIXINR_SPKMIXR_VOL_SHIFT 5 /* MIXINR_SPKMIXR_VOL */ +#define WM8993_MIXINR_SPKMIXR_VOL_WIDTH 1 /* MIXINR_SPKMIXR_VOL */ +#define WM8993_IN1RP_SPKMIXR_VOL 0x0010 /* IN1RP_SPKMIXR_VOL */ +#define WM8993_IN1RP_SPKMIXR_VOL_MASK 0x0010 /* IN1RP_SPKMIXR_VOL */ +#define WM8993_IN1RP_SPKMIXR_VOL_SHIFT 4 /* IN1RP_SPKMIXR_VOL */ +#define WM8993_IN1RP_SPKMIXR_VOL_WIDTH 1 /* IN1RP_SPKMIXR_VOL */ +#define WM8993_MIXOUTR_SPKMIXR_VOL 0x0008 /* MIXOUTR_SPKMIXR_VOL */ +#define WM8993_MIXOUTR_SPKMIXR_VOL_MASK 0x0008 /* MIXOUTR_SPKMIXR_VOL */ +#define WM8993_MIXOUTR_SPKMIXR_VOL_SHIFT 3 /* MIXOUTR_SPKMIXR_VOL */ +#define WM8993_MIXOUTR_SPKMIXR_VOL_WIDTH 1 /* MIXOUTR_SPKMIXR_VOL */ +#define WM8993_DACR_SPKMIXR_VOL 0x0004 /* DACR_SPKMIXR_VOL */ +#define WM8993_DACR_SPKMIXR_VOL_MASK 0x0004 /* DACR_SPKMIXR_VOL */ +#define WM8993_DACR_SPKMIXR_VOL_SHIFT 2 /* DACR_SPKMIXR_VOL */ +#define WM8993_DACR_SPKMIXR_VOL_WIDTH 1 /* DACR_SPKMIXR_VOL */ +#define WM8993_SPKMIXR_VOL_MASK 0x0003 /* SPKMIXR_VOL - [1:0] */ +#define WM8993_SPKMIXR_VOL_SHIFT 0 /* SPKMIXR_VOL - [1:0] */ +#define WM8993_SPKMIXR_VOL_WIDTH 2 /* SPKMIXR_VOL - [1:0] */ + +/* + * R36 (0x24) - SPKOUT Mixers + */ +#define WM8993_VRX_TO_SPKOUTL 0x0020 /* VRX_TO_SPKOUTL */ +#define WM8993_VRX_TO_SPKOUTL_MASK 0x0020 /* VRX_TO_SPKOUTL */ +#define WM8993_VRX_TO_SPKOUTL_SHIFT 5 /* VRX_TO_SPKOUTL */ +#define WM8993_VRX_TO_SPKOUTL_WIDTH 1 /* VRX_TO_SPKOUTL */ +#define WM8993_SPKMIXL_TO_SPKOUTL 0x0010 /* SPKMIXL_TO_SPKOUTL */ +#define WM8993_SPKMIXL_TO_SPKOUTL_MASK 0x0010 /* SPKMIXL_TO_SPKOUTL */ +#define WM8993_SPKMIXL_TO_SPKOUTL_SHIFT 4 /* SPKMIXL_TO_SPKOUTL */ +#define WM8993_SPKMIXL_TO_SPKOUTL_WIDTH 1 /* SPKMIXL_TO_SPKOUTL */ +#define WM8993_SPKMIXR_TO_SPKOUTL 0x0008 /* SPKMIXR_TO_SPKOUTL */ +#define WM8993_SPKMIXR_TO_SPKOUTL_MASK 0x0008 /* SPKMIXR_TO_SPKOUTL */ +#define WM8993_SPKMIXR_TO_SPKOUTL_SHIFT 3 /* SPKMIXR_TO_SPKOUTL */ +#define WM8993_SPKMIXR_TO_SPKOUTL_WIDTH 1 /* SPKMIXR_TO_SPKOUTL */ +#define WM8993_VRX_TO_SPKOUTR 0x0004 /* VRX_TO_SPKOUTR */ +#define WM8993_VRX_TO_SPKOUTR_MASK 0x0004 /* VRX_TO_SPKOUTR */ +#define WM8993_VRX_TO_SPKOUTR_SHIFT 2 /* VRX_TO_SPKOUTR */ +#define WM8993_VRX_TO_SPKOUTR_WIDTH 1 /* VRX_TO_SPKOUTR */ +#define WM8993_SPKMIXL_TO_SPKOUTR 0x0002 /* SPKMIXL_TO_SPKOUTR */ +#define WM8993_SPKMIXL_TO_SPKOUTR_MASK 0x0002 /* SPKMIXL_TO_SPKOUTR */ +#define WM8993_SPKMIXL_TO_SPKOUTR_SHIFT 1 /* SPKMIXL_TO_SPKOUTR */ +#define WM8993_SPKMIXL_TO_SPKOUTR_WIDTH 1 /* SPKMIXL_TO_SPKOUTR */ +#define WM8993_SPKMIXR_TO_SPKOUTR 0x0001 /* SPKMIXR_TO_SPKOUTR */ +#define WM8993_SPKMIXR_TO_SPKOUTR_MASK 0x0001 /* SPKMIXR_TO_SPKOUTR */ +#define WM8993_SPKMIXR_TO_SPKOUTR_SHIFT 0 /* SPKMIXR_TO_SPKOUTR */ +#define WM8993_SPKMIXR_TO_SPKOUTR_WIDTH 1 /* SPKMIXR_TO_SPKOUTR */ + +/* + * R37 (0x25) - SPKOUT Boost + */ +#define WM8993_SPKOUTL_BOOST_MASK 0x0038 /* SPKOUTL_BOOST - [5:3] */ +#define WM8993_SPKOUTL_BOOST_SHIFT 3 /* SPKOUTL_BOOST - [5:3] */ +#define WM8993_SPKOUTL_BOOST_WIDTH 3 /* SPKOUTL_BOOST - [5:3] */ +#define WM8993_SPKOUTR_BOOST_MASK 0x0007 /* SPKOUTR_BOOST - [2:0] */ +#define WM8993_SPKOUTR_BOOST_SHIFT 0 /* SPKOUTR_BOOST - [2:0] */ +#define WM8993_SPKOUTR_BOOST_WIDTH 3 /* SPKOUTR_BOOST - [2:0] */ + +/* + * R38 (0x26) - Speaker Volume Left + */ +#define WM8993_SPKOUT_VU 0x0100 /* SPKOUT_VU */ +#define WM8993_SPKOUT_VU_MASK 0x0100 /* SPKOUT_VU */ +#define WM8993_SPKOUT_VU_SHIFT 8 /* SPKOUT_VU */ +#define WM8993_SPKOUT_VU_WIDTH 1 /* SPKOUT_VU */ +#define WM8993_SPKOUTL_ZC 0x0080 /* SPKOUTL_ZC */ +#define WM8993_SPKOUTL_ZC_MASK 0x0080 /* SPKOUTL_ZC */ +#define WM8993_SPKOUTL_ZC_SHIFT 7 /* SPKOUTL_ZC */ +#define WM8993_SPKOUTL_ZC_WIDTH 1 /* SPKOUTL_ZC */ +#define WM8993_SPKOUTL_MUTE_N 0x0040 /* SPKOUTL_MUTE_N */ +#define WM8993_SPKOUTL_MUTE_N_MASK 0x0040 /* SPKOUTL_MUTE_N */ +#define WM8993_SPKOUTL_MUTE_N_SHIFT 6 /* SPKOUTL_MUTE_N */ +#define WM8993_SPKOUTL_MUTE_N_WIDTH 1 /* SPKOUTL_MUTE_N */ +#define WM8993_SPKOUTL_VOL_MASK 0x003F /* SPKOUTL_VOL - [5:0] */ +#define WM8993_SPKOUTL_VOL_SHIFT 0 /* SPKOUTL_VOL - [5:0] */ +#define WM8993_SPKOUTL_VOL_WIDTH 6 /* SPKOUTL_VOL - [5:0] */ + +/* + * R39 (0x27) - Speaker Volume Right + */ +#define WM8993_SPKOUT_VU 0x0100 /* SPKOUT_VU */ +#define WM8993_SPKOUT_VU_MASK 0x0100 /* SPKOUT_VU */ +#define WM8993_SPKOUT_VU_SHIFT 8 /* SPKOUT_VU */ +#define WM8993_SPKOUT_VU_WIDTH 1 /* SPKOUT_VU */ +#define WM8993_SPKOUTR_ZC 0x0080 /* SPKOUTR_ZC */ +#define WM8993_SPKOUTR_ZC_MASK 0x0080 /* SPKOUTR_ZC */ +#define WM8993_SPKOUTR_ZC_SHIFT 7 /* SPKOUTR_ZC */ +#define WM8993_SPKOUTR_ZC_WIDTH 1 /* SPKOUTR_ZC */ +#define WM8993_SPKOUTR_MUTE_N 0x0040 /* SPKOUTR_MUTE_N */ +#define WM8993_SPKOUTR_MUTE_N_MASK 0x0040 /* SPKOUTR_MUTE_N */ +#define WM8993_SPKOUTR_MUTE_N_SHIFT 6 /* SPKOUTR_MUTE_N */ +#define WM8993_SPKOUTR_MUTE_N_WIDTH 1 /* SPKOUTR_MUTE_N */ +#define WM8993_SPKOUTR_VOL_MASK 0x003F /* SPKOUTR_VOL - [5:0] */ +#define WM8993_SPKOUTR_VOL_SHIFT 0 /* SPKOUTR_VOL - [5:0] */ +#define WM8993_SPKOUTR_VOL_WIDTH 6 /* SPKOUTR_VOL - [5:0] */ + +/* + * R40 (0x28) - Input Mixer2 + */ +#define WM8993_IN2LP_TO_IN2L 0x0080 /* IN2LP_TO_IN2L */ +#define WM8993_IN2LP_TO_IN2L_MASK 0x0080 /* IN2LP_TO_IN2L */ +#define WM8993_IN2LP_TO_IN2L_SHIFT 7 /* IN2LP_TO_IN2L */ +#define WM8993_IN2LP_TO_IN2L_WIDTH 1 /* IN2LP_TO_IN2L */ +#define WM8993_IN2LN_TO_IN2L 0x0040 /* IN2LN_TO_IN2L */ +#define WM8993_IN2LN_TO_IN2L_MASK 0x0040 /* IN2LN_TO_IN2L */ +#define WM8993_IN2LN_TO_IN2L_SHIFT 6 /* IN2LN_TO_IN2L */ +#define WM8993_IN2LN_TO_IN2L_WIDTH 1 /* IN2LN_TO_IN2L */ +#define WM8993_IN1LP_TO_IN1L 0x0020 /* IN1LP_TO_IN1L */ +#define WM8993_IN1LP_TO_IN1L_MASK 0x0020 /* IN1LP_TO_IN1L */ +#define WM8993_IN1LP_TO_IN1L_SHIFT 5 /* IN1LP_TO_IN1L */ +#define WM8993_IN1LP_TO_IN1L_WIDTH 1 /* IN1LP_TO_IN1L */ +#define WM8993_IN1LN_TO_IN1L 0x0010 /* IN1LN_TO_IN1L */ +#define WM8993_IN1LN_TO_IN1L_MASK 0x0010 /* IN1LN_TO_IN1L */ +#define WM8993_IN1LN_TO_IN1L_SHIFT 4 /* IN1LN_TO_IN1L */ +#define WM8993_IN1LN_TO_IN1L_WIDTH 1 /* IN1LN_TO_IN1L */ +#define WM8993_IN2RP_TO_IN2R 0x0008 /* IN2RP_TO_IN2R */ +#define WM8993_IN2RP_TO_IN2R_MASK 0x0008 /* IN2RP_TO_IN2R */ +#define WM8993_IN2RP_TO_IN2R_SHIFT 3 /* IN2RP_TO_IN2R */ +#define WM8993_IN2RP_TO_IN2R_WIDTH 1 /* IN2RP_TO_IN2R */ +#define WM8993_IN2RN_TO_IN2R 0x0004 /* IN2RN_TO_IN2R */ +#define WM8993_IN2RN_TO_IN2R_MASK 0x0004 /* IN2RN_TO_IN2R */ +#define WM8993_IN2RN_TO_IN2R_SHIFT 2 /* IN2RN_TO_IN2R */ +#define WM8993_IN2RN_TO_IN2R_WIDTH 1 /* IN2RN_TO_IN2R */ +#define WM8993_IN1RP_TO_IN1R 0x0002 /* IN1RP_TO_IN1R */ +#define WM8993_IN1RP_TO_IN1R_MASK 0x0002 /* IN1RP_TO_IN1R */ +#define WM8993_IN1RP_TO_IN1R_SHIFT 1 /* IN1RP_TO_IN1R */ +#define WM8993_IN1RP_TO_IN1R_WIDTH 1 /* IN1RP_TO_IN1R */ +#define WM8993_IN1RN_TO_IN1R 0x0001 /* IN1RN_TO_IN1R */ +#define WM8993_IN1RN_TO_IN1R_MASK 0x0001 /* IN1RN_TO_IN1R */ +#define WM8993_IN1RN_TO_IN1R_SHIFT 0 /* IN1RN_TO_IN1R */ +#define WM8993_IN1RN_TO_IN1R_WIDTH 1 /* IN1RN_TO_IN1R */ + +/* + * R41 (0x29) - Input Mixer3 + */ +#define WM8993_IN2L_TO_MIXINL 0x0100 /* IN2L_TO_MIXINL */ +#define WM8993_IN2L_TO_MIXINL_MASK 0x0100 /* IN2L_TO_MIXINL */ +#define WM8993_IN2L_TO_MIXINL_SHIFT 8 /* IN2L_TO_MIXINL */ +#define WM8993_IN2L_TO_MIXINL_WIDTH 1 /* IN2L_TO_MIXINL */ +#define WM8993_IN2L_MIXINL_VOL 0x0080 /* IN2L_MIXINL_VOL */ +#define WM8993_IN2L_MIXINL_VOL_MASK 0x0080 /* IN2L_MIXINL_VOL */ +#define WM8993_IN2L_MIXINL_VOL_SHIFT 7 /* IN2L_MIXINL_VOL */ +#define WM8993_IN2L_MIXINL_VOL_WIDTH 1 /* IN2L_MIXINL_VOL */ +#define WM8993_IN1L_TO_MIXINL 0x0020 /* IN1L_TO_MIXINL */ +#define WM8993_IN1L_TO_MIXINL_MASK 0x0020 /* IN1L_TO_MIXINL */ +#define WM8993_IN1L_TO_MIXINL_SHIFT 5 /* IN1L_TO_MIXINL */ +#define WM8993_IN1L_TO_MIXINL_WIDTH 1 /* IN1L_TO_MIXINL */ +#define WM8993_IN1L_MIXINL_VOL 0x0010 /* IN1L_MIXINL_VOL */ +#define WM8993_IN1L_MIXINL_VOL_MASK 0x0010 /* IN1L_MIXINL_VOL */ +#define WM8993_IN1L_MIXINL_VOL_SHIFT 4 /* IN1L_MIXINL_VOL */ +#define WM8993_IN1L_MIXINL_VOL_WIDTH 1 /* IN1L_MIXINL_VOL */ +#define WM8993_MIXOUTL_MIXINL_VOL_MASK 0x0007 /* MIXOUTL_MIXINL_VOL - [2:0] */ +#define WM8993_MIXOUTL_MIXINL_VOL_SHIFT 0 /* MIXOUTL_MIXINL_VOL - [2:0] */ +#define WM8993_MIXOUTL_MIXINL_VOL_WIDTH 3 /* MIXOUTL_MIXINL_VOL - [2:0] */ + +/* + * R42 (0x2A) - Input Mixer4 + */ +#define WM8993_IN2R_TO_MIXINR 0x0100 /* IN2R_TO_MIXINR */ +#define WM8993_IN2R_TO_MIXINR_MASK 0x0100 /* IN2R_TO_MIXINR */ +#define WM8993_IN2R_TO_MIXINR_SHIFT 8 /* IN2R_TO_MIXINR */ +#define WM8993_IN2R_TO_MIXINR_WIDTH 1 /* IN2R_TO_MIXINR */ +#define WM8993_IN2R_MIXINR_VOL 0x0080 /* IN2R_MIXINR_VOL */ +#define WM8993_IN2R_MIXINR_VOL_MASK 0x0080 /* IN2R_MIXINR_VOL */ +#define WM8993_IN2R_MIXINR_VOL_SHIFT 7 /* IN2R_MIXINR_VOL */ +#define WM8993_IN2R_MIXINR_VOL_WIDTH 1 /* IN2R_MIXINR_VOL */ +#define WM8993_IN1R_TO_MIXINR 0x0020 /* IN1R_TO_MIXINR */ +#define WM8993_IN1R_TO_MIXINR_MASK 0x0020 /* IN1R_TO_MIXINR */ +#define WM8993_IN1R_TO_MIXINR_SHIFT 5 /* IN1R_TO_MIXINR */ +#define WM8993_IN1R_TO_MIXINR_WIDTH 1 /* IN1R_TO_MIXINR */ +#define WM8993_IN1R_MIXINR_VOL 0x0010 /* IN1R_MIXINR_VOL */ +#define WM8993_IN1R_MIXINR_VOL_MASK 0x0010 /* IN1R_MIXINR_VOL */ +#define WM8993_IN1R_MIXINR_VOL_SHIFT 4 /* IN1R_MIXINR_VOL */ +#define WM8993_IN1R_MIXINR_VOL_WIDTH 1 /* IN1R_MIXINR_VOL */ +#define WM8993_MIXOUTR_MIXINR_VOL_MASK 0x0007 /* MIXOUTR_MIXINR_VOL - [2:0] */ +#define WM8993_MIXOUTR_MIXINR_VOL_SHIFT 0 /* MIXOUTR_MIXINR_VOL - [2:0] */ +#define WM8993_MIXOUTR_MIXINR_VOL_WIDTH 3 /* MIXOUTR_MIXINR_VOL - [2:0] */ + +/* + * R43 (0x2B) - Input Mixer5 + */ +#define WM8993_IN1LP_MIXINL_VOL_MASK 0x01C0 /* IN1LP_MIXINL_VOL - [8:6] */ +#define WM8993_IN1LP_MIXINL_VOL_SHIFT 6 /* IN1LP_MIXINL_VOL - [8:6] */ +#define WM8993_IN1LP_MIXINL_VOL_WIDTH 3 /* IN1LP_MIXINL_VOL - [8:6] */ +#define WM8993_VRX_MIXINL_VOL_MASK 0x0007 /* VRX_MIXINL_VOL - [2:0] */ +#define WM8993_VRX_MIXINL_VOL_SHIFT 0 /* VRX_MIXINL_VOL - [2:0] */ +#define WM8993_VRX_MIXINL_VOL_WIDTH 3 /* VRX_MIXINL_VOL - [2:0] */ + +/* + * R44 (0x2C) - Input Mixer6 + */ +#define WM8993_IN1RP_MIXINR_VOL_MASK 0x01C0 /* IN1RP_MIXINR_VOL - [8:6] */ +#define WM8993_IN1RP_MIXINR_VOL_SHIFT 6 /* IN1RP_MIXINR_VOL - [8:6] */ +#define WM8993_IN1RP_MIXINR_VOL_WIDTH 3 /* IN1RP_MIXINR_VOL - [8:6] */ +#define WM8993_VRX_MIXINR_VOL_MASK 0x0007 /* VRX_MIXINR_VOL - [2:0] */ +#define WM8993_VRX_MIXINR_VOL_SHIFT 0 /* VRX_MIXINR_VOL - [2:0] */ +#define WM8993_VRX_MIXINR_VOL_WIDTH 3 /* VRX_MIXINR_VOL - [2:0] */ + +/* + * R45 (0x2D) - Output Mixer1 + */ +#define WM8993_DACL_TO_HPOUT1L 0x0100 /* DACL_TO_HPOUT1L */ +#define WM8993_DACL_TO_HPOUT1L_MASK 0x0100 /* DACL_TO_HPOUT1L */ +#define WM8993_DACL_TO_HPOUT1L_SHIFT 8 /* DACL_TO_HPOUT1L */ +#define WM8993_DACL_TO_HPOUT1L_WIDTH 1 /* DACL_TO_HPOUT1L */ +#define WM8993_MIXINR_TO_MIXOUTL 0x0080 /* MIXINR_TO_MIXOUTL */ +#define WM8993_MIXINR_TO_MIXOUTL_MASK 0x0080 /* MIXINR_TO_MIXOUTL */ +#define WM8993_MIXINR_TO_MIXOUTL_SHIFT 7 /* MIXINR_TO_MIXOUTL */ +#define WM8993_MIXINR_TO_MIXOUTL_WIDTH 1 /* MIXINR_TO_MIXOUTL */ +#define WM8993_MIXINL_TO_MIXOUTL 0x0040 /* MIXINL_TO_MIXOUTL */ +#define WM8993_MIXINL_TO_MIXOUTL_MASK 0x0040 /* MIXINL_TO_MIXOUTL */ +#define WM8993_MIXINL_TO_MIXOUTL_SHIFT 6 /* MIXINL_TO_MIXOUTL */ +#define WM8993_MIXINL_TO_MIXOUTL_WIDTH 1 /* MIXINL_TO_MIXOUTL */ +#define WM8993_IN2RN_TO_MIXOUTL 0x0020 /* IN2RN_TO_MIXOUTL */ +#define WM8993_IN2RN_TO_MIXOUTL_MASK 0x0020 /* IN2RN_TO_MIXOUTL */ +#define WM8993_IN2RN_TO_MIXOUTL_SHIFT 5 /* IN2RN_TO_MIXOUTL */ +#define WM8993_IN2RN_TO_MIXOUTL_WIDTH 1 /* IN2RN_TO_MIXOUTL */ +#define WM8993_IN2LN_TO_MIXOUTL 0x0010 /* IN2LN_TO_MIXOUTL */ +#define WM8993_IN2LN_TO_MIXOUTL_MASK 0x0010 /* IN2LN_TO_MIXOUTL */ +#define WM8993_IN2LN_TO_MIXOUTL_SHIFT 4 /* IN2LN_TO_MIXOUTL */ +#define WM8993_IN2LN_TO_MIXOUTL_WIDTH 1 /* IN2LN_TO_MIXOUTL */ +#define WM8993_IN1R_TO_MIXOUTL 0x0008 /* IN1R_TO_MIXOUTL */ +#define WM8993_IN1R_TO_MIXOUTL_MASK 0x0008 /* IN1R_TO_MIXOUTL */ +#define WM8993_IN1R_TO_MIXOUTL_SHIFT 3 /* IN1R_TO_MIXOUTL */ +#define WM8993_IN1R_TO_MIXOUTL_WIDTH 1 /* IN1R_TO_MIXOUTL */ +#define WM8993_IN1L_TO_MIXOUTL 0x0004 /* IN1L_TO_MIXOUTL */ +#define WM8993_IN1L_TO_MIXOUTL_MASK 0x0004 /* IN1L_TO_MIXOUTL */ +#define WM8993_IN1L_TO_MIXOUTL_SHIFT 2 /* IN1L_TO_MIXOUTL */ +#define WM8993_IN1L_TO_MIXOUTL_WIDTH 1 /* IN1L_TO_MIXOUTL */ +#define WM8993_IN2LP_TO_MIXOUTL 0x0002 /* IN2LP_TO_MIXOUTL */ +#define WM8993_IN2LP_TO_MIXOUTL_MASK 0x0002 /* IN2LP_TO_MIXOUTL */ +#define WM8993_IN2LP_TO_MIXOUTL_SHIFT 1 /* IN2LP_TO_MIXOUTL */ +#define WM8993_IN2LP_TO_MIXOUTL_WIDTH 1 /* IN2LP_TO_MIXOUTL */ +#define WM8993_DACL_TO_MIXOUTL 0x0001 /* DACL_TO_MIXOUTL */ +#define WM8993_DACL_TO_MIXOUTL_MASK 0x0001 /* DACL_TO_MIXOUTL */ +#define WM8993_DACL_TO_MIXOUTL_SHIFT 0 /* DACL_TO_MIXOUTL */ +#define WM8993_DACL_TO_MIXOUTL_WIDTH 1 /* DACL_TO_MIXOUTL */ + +/* + * R46 (0x2E) - Output Mixer2 + */ +#define WM8993_DACR_TO_HPOUT1R 0x0100 /* DACR_TO_HPOUT1R */ +#define WM8993_DACR_TO_HPOUT1R_MASK 0x0100 /* DACR_TO_HPOUT1R */ +#define WM8993_DACR_TO_HPOUT1R_SHIFT 8 /* DACR_TO_HPOUT1R */ +#define WM8993_DACR_TO_HPOUT1R_WIDTH 1 /* DACR_TO_HPOUT1R */ +#define WM8993_MIXINL_TO_MIXOUTR 0x0080 /* MIXINL_TO_MIXOUTR */ +#define WM8993_MIXINL_TO_MIXOUTR_MASK 0x0080 /* MIXINL_TO_MIXOUTR */ +#define WM8993_MIXINL_TO_MIXOUTR_SHIFT 7 /* MIXINL_TO_MIXOUTR */ +#define WM8993_MIXINL_TO_MIXOUTR_WIDTH 1 /* MIXINL_TO_MIXOUTR */ +#define WM8993_MIXINR_TO_MIXOUTR 0x0040 /* MIXINR_TO_MIXOUTR */ +#define WM8993_MIXINR_TO_MIXOUTR_MASK 0x0040 /* MIXINR_TO_MIXOUTR */ +#define WM8993_MIXINR_TO_MIXOUTR_SHIFT 6 /* MIXINR_TO_MIXOUTR */ +#define WM8993_MIXINR_TO_MIXOUTR_WIDTH 1 /* MIXINR_TO_MIXOUTR */ +#define WM8993_IN2LN_TO_MIXOUTR 0x0020 /* IN2LN_TO_MIXOUTR */ +#define WM8993_IN2LN_TO_MIXOUTR_MASK 0x0020 /* IN2LN_TO_MIXOUTR */ +#define WM8993_IN2LN_TO_MIXOUTR_SHIFT 5 /* IN2LN_TO_MIXOUTR */ +#define WM8993_IN2LN_TO_MIXOUTR_WIDTH 1 /* IN2LN_TO_MIXOUTR */ +#define WM8993_IN2RN_TO_MIXOUTR 0x0010 /* IN2RN_TO_MIXOUTR */ +#define WM8993_IN2RN_TO_MIXOUTR_MASK 0x0010 /* IN2RN_TO_MIXOUTR */ +#define WM8993_IN2RN_TO_MIXOUTR_SHIFT 4 /* IN2RN_TO_MIXOUTR */ +#define WM8993_IN2RN_TO_MIXOUTR_WIDTH 1 /* IN2RN_TO_MIXOUTR */ +#define WM8993_IN1L_TO_MIXOUTR 0x0008 /* IN1L_TO_MIXOUTR */ +#define WM8993_IN1L_TO_MIXOUTR_MASK 0x0008 /* IN1L_TO_MIXOUTR */ +#define WM8993_IN1L_TO_MIXOUTR_SHIFT 3 /* IN1L_TO_MIXOUTR */ +#define WM8993_IN1L_TO_MIXOUTR_WIDTH 1 /* IN1L_TO_MIXOUTR */ +#define WM8993_IN1R_TO_MIXOUTR 0x0004 /* IN1R_TO_MIXOUTR */ +#define WM8993_IN1R_TO_MIXOUTR_MASK 0x0004 /* IN1R_TO_MIXOUTR */ +#define WM8993_IN1R_TO_MIXOUTR_SHIFT 2 /* IN1R_TO_MIXOUTR */ +#define WM8993_IN1R_TO_MIXOUTR_WIDTH 1 /* IN1R_TO_MIXOUTR */ +#define WM8993_IN2RP_TO_MIXOUTR 0x0002 /* IN2RP_TO_MIXOUTR */ +#define WM8993_IN2RP_TO_MIXOUTR_MASK 0x0002 /* IN2RP_TO_MIXOUTR */ +#define WM8993_IN2RP_TO_MIXOUTR_SHIFT 1 /* IN2RP_TO_MIXOUTR */ +#define WM8993_IN2RP_TO_MIXOUTR_WIDTH 1 /* IN2RP_TO_MIXOUTR */ +#define WM8993_DACR_TO_MIXOUTR 0x0001 /* DACR_TO_MIXOUTR */ +#define WM8993_DACR_TO_MIXOUTR_MASK 0x0001 /* DACR_TO_MIXOUTR */ +#define WM8993_DACR_TO_MIXOUTR_SHIFT 0 /* DACR_TO_MIXOUTR */ +#define WM8993_DACR_TO_MIXOUTR_WIDTH 1 /* DACR_TO_MIXOUTR */ + +/* + * R47 (0x2F) - Output Mixer3 + */ +#define WM8993_IN2LP_MIXOUTL_VOL_MASK 0x0E00 /* IN2LP_MIXOUTL_VOL - [11:9] */ +#define WM8993_IN2LP_MIXOUTL_VOL_SHIFT 9 /* IN2LP_MIXOUTL_VOL - [11:9] */ +#define WM8993_IN2LP_MIXOUTL_VOL_WIDTH 3 /* IN2LP_MIXOUTL_VOL - [11:9] */ +#define WM8993_IN2LN_MIXOUTL_VOL_MASK 0x01C0 /* IN2LN_MIXOUTL_VOL - [8:6] */ +#define WM8993_IN2LN_MIXOUTL_VOL_SHIFT 6 /* IN2LN_MIXOUTL_VOL - [8:6] */ +#define WM8993_IN2LN_MIXOUTL_VOL_WIDTH 3 /* IN2LN_MIXOUTL_VOL - [8:6] */ +#define WM8993_IN1R_MIXOUTL_VOL_MASK 0x0038 /* IN1R_MIXOUTL_VOL - [5:3] */ +#define WM8993_IN1R_MIXOUTL_VOL_SHIFT 3 /* IN1R_MIXOUTL_VOL - [5:3] */ +#define WM8993_IN1R_MIXOUTL_VOL_WIDTH 3 /* IN1R_MIXOUTL_VOL - [5:3] */ +#define WM8993_IN1L_MIXOUTL_VOL_MASK 0x0007 /* IN1L_MIXOUTL_VOL - [2:0] */ +#define WM8993_IN1L_MIXOUTL_VOL_SHIFT 0 /* IN1L_MIXOUTL_VOL - [2:0] */ +#define WM8993_IN1L_MIXOUTL_VOL_WIDTH 3 /* IN1L_MIXOUTL_VOL - [2:0] */ + +/* + * R48 (0x30) - Output Mixer4 + */ +#define WM8993_IN2RP_MIXOUTR_VOL_MASK 0x0E00 /* IN2RP_MIXOUTR_VOL - [11:9] */ +#define WM8993_IN2RP_MIXOUTR_VOL_SHIFT 9 /* IN2RP_MIXOUTR_VOL - [11:9] */ +#define WM8993_IN2RP_MIXOUTR_VOL_WIDTH 3 /* IN2RP_MIXOUTR_VOL - [11:9] */ +#define WM8993_IN2RN_MIXOUTR_VOL_MASK 0x01C0 /* IN2RN_MIXOUTR_VOL - [8:6] */ +#define WM8993_IN2RN_MIXOUTR_VOL_SHIFT 6 /* IN2RN_MIXOUTR_VOL - [8:6] */ +#define WM8993_IN2RN_MIXOUTR_VOL_WIDTH 3 /* IN2RN_MIXOUTR_VOL - [8:6] */ +#define WM8993_IN1L_MIXOUTR_VOL_MASK 0x0038 /* IN1L_MIXOUTR_VOL - [5:3] */ +#define WM8993_IN1L_MIXOUTR_VOL_SHIFT 3 /* IN1L_MIXOUTR_VOL - [5:3] */ +#define WM8993_IN1L_MIXOUTR_VOL_WIDTH 3 /* IN1L_MIXOUTR_VOL - [5:3] */ +#define WM8993_IN1R_MIXOUTR_VOL_MASK 0x0007 /* IN1R_MIXOUTR_VOL - [2:0] */ +#define WM8993_IN1R_MIXOUTR_VOL_SHIFT 0 /* IN1R_MIXOUTR_VOL - [2:0] */ +#define WM8993_IN1R_MIXOUTR_VOL_WIDTH 3 /* IN1R_MIXOUTR_VOL - [2:0] */ + +/* + * R49 (0x31) - Output Mixer5 + */ +#define WM8993_DACL_MIXOUTL_VOL_MASK 0x0E00 /* DACL_MIXOUTL_VOL - [11:9] */ +#define WM8993_DACL_MIXOUTL_VOL_SHIFT 9 /* DACL_MIXOUTL_VOL - [11:9] */ +#define WM8993_DACL_MIXOUTL_VOL_WIDTH 3 /* DACL_MIXOUTL_VOL - [11:9] */ +#define WM8993_IN2RN_MIXOUTL_VOL_MASK 0x01C0 /* IN2RN_MIXOUTL_VOL - [8:6] */ +#define WM8993_IN2RN_MIXOUTL_VOL_SHIFT 6 /* IN2RN_MIXOUTL_VOL - [8:6] */ +#define WM8993_IN2RN_MIXOUTL_VOL_WIDTH 3 /* IN2RN_MIXOUTL_VOL - [8:6] */ +#define WM8993_MIXINR_MIXOUTL_VOL_MASK 0x0038 /* MIXINR_MIXOUTL_VOL - [5:3] */ +#define WM8993_MIXINR_MIXOUTL_VOL_SHIFT 3 /* MIXINR_MIXOUTL_VOL - [5:3] */ +#define WM8993_MIXINR_MIXOUTL_VOL_WIDTH 3 /* MIXINR_MIXOUTL_VOL - [5:3] */ +#define WM8993_MIXINL_MIXOUTL_VOL_MASK 0x0007 /* MIXINL_MIXOUTL_VOL - [2:0] */ +#define WM8993_MIXINL_MIXOUTL_VOL_SHIFT 0 /* MIXINL_MIXOUTL_VOL - [2:0] */ +#define WM8993_MIXINL_MIXOUTL_VOL_WIDTH 3 /* MIXINL_MIXOUTL_VOL - [2:0] */ + +/* + * R50 (0x32) - Output Mixer6 + */ +#define WM8993_DACR_MIXOUTR_VOL_MASK 0x0E00 /* DACR_MIXOUTR_VOL - [11:9] */ +#define WM8993_DACR_MIXOUTR_VOL_SHIFT 9 /* DACR_MIXOUTR_VOL - [11:9] */ +#define WM8993_DACR_MIXOUTR_VOL_WIDTH 3 /* DACR_MIXOUTR_VOL - [11:9] */ +#define WM8993_IN2LN_MIXOUTR_VOL_MASK 0x01C0 /* IN2LN_MIXOUTR_VOL - [8:6] */ +#define WM8993_IN2LN_MIXOUTR_VOL_SHIFT 6 /* IN2LN_MIXOUTR_VOL - [8:6] */ +#define WM8993_IN2LN_MIXOUTR_VOL_WIDTH 3 /* IN2LN_MIXOUTR_VOL - [8:6] */ +#define WM8993_MIXINL_MIXOUTR_VOL_MASK 0x0038 /* MIXINL_MIXOUTR_VOL - [5:3] */ +#define WM8993_MIXINL_MIXOUTR_VOL_SHIFT 3 /* MIXINL_MIXOUTR_VOL - [5:3] */ +#define WM8993_MIXINL_MIXOUTR_VOL_WIDTH 3 /* MIXINL_MIXOUTR_VOL - [5:3] */ +#define WM8993_MIXINR_MIXOUTR_VOL_MASK 0x0007 /* MIXINR_MIXOUTR_VOL - [2:0] */ +#define WM8993_MIXINR_MIXOUTR_VOL_SHIFT 0 /* MIXINR_MIXOUTR_VOL - [2:0] */ +#define WM8993_MIXINR_MIXOUTR_VOL_WIDTH 3 /* MIXINR_MIXOUTR_VOL - [2:0] */ + +/* + * R51 (0x33) - HPOUT2 Mixer + */ +#define WM8993_VRX_TO_HPOUT2 0x0020 /* VRX_TO_HPOUT2 */ +#define WM8993_VRX_TO_HPOUT2_MASK 0x0020 /* VRX_TO_HPOUT2 */ +#define WM8993_VRX_TO_HPOUT2_SHIFT 5 /* VRX_TO_HPOUT2 */ +#define WM8993_VRX_TO_HPOUT2_WIDTH 1 /* VRX_TO_HPOUT2 */ +#define WM8993_MIXOUTLVOL_TO_HPOUT2 0x0010 /* MIXOUTLVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTLVOL_TO_HPOUT2_MASK 0x0010 /* MIXOUTLVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTLVOL_TO_HPOUT2_SHIFT 4 /* MIXOUTLVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTLVOL_TO_HPOUT2_WIDTH 1 /* MIXOUTLVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTRVOL_TO_HPOUT2 0x0008 /* MIXOUTRVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTRVOL_TO_HPOUT2_MASK 0x0008 /* MIXOUTRVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTRVOL_TO_HPOUT2_SHIFT 3 /* MIXOUTRVOL_TO_HPOUT2 */ +#define WM8993_MIXOUTRVOL_TO_HPOUT2_WIDTH 1 /* MIXOUTRVOL_TO_HPOUT2 */ + +/* + * R52 (0x34) - Line Mixer1 + */ +#define WM8993_MIXOUTL_TO_LINEOUT1N 0x0040 /* MIXOUTL_TO_LINEOUT1N */ +#define WM8993_MIXOUTL_TO_LINEOUT1N_MASK 0x0040 /* MIXOUTL_TO_LINEOUT1N */ +#define WM8993_MIXOUTL_TO_LINEOUT1N_SHIFT 6 /* MIXOUTL_TO_LINEOUT1N */ +#define WM8993_MIXOUTL_TO_LINEOUT1N_WIDTH 1 /* MIXOUTL_TO_LINEOUT1N */ +#define WM8993_MIXOUTR_TO_LINEOUT1N 0x0020 /* MIXOUTR_TO_LINEOUT1N */ +#define WM8993_MIXOUTR_TO_LINEOUT1N_MASK 0x0020 /* MIXOUTR_TO_LINEOUT1N */ +#define WM8993_MIXOUTR_TO_LINEOUT1N_SHIFT 5 /* MIXOUTR_TO_LINEOUT1N */ +#define WM8993_MIXOUTR_TO_LINEOUT1N_WIDTH 1 /* MIXOUTR_TO_LINEOUT1N */ +#define WM8993_LINEOUT1_MODE 0x0010 /* LINEOUT1_MODE */ +#define WM8993_LINEOUT1_MODE_MASK 0x0010 /* LINEOUT1_MODE */ +#define WM8993_LINEOUT1_MODE_SHIFT 4 /* LINEOUT1_MODE */ +#define WM8993_LINEOUT1_MODE_WIDTH 1 /* LINEOUT1_MODE */ +#define WM8993_IN1R_TO_LINEOUT1P 0x0004 /* IN1R_TO_LINEOUT1P */ +#define WM8993_IN1R_TO_LINEOUT1P_MASK 0x0004 /* IN1R_TO_LINEOUT1P */ +#define WM8993_IN1R_TO_LINEOUT1P_SHIFT 2 /* IN1R_TO_LINEOUT1P */ +#define WM8993_IN1R_TO_LINEOUT1P_WIDTH 1 /* IN1R_TO_LINEOUT1P */ +#define WM8993_IN1L_TO_LINEOUT1P 0x0002 /* IN1L_TO_LINEOUT1P */ +#define WM8993_IN1L_TO_LINEOUT1P_MASK 0x0002 /* IN1L_TO_LINEOUT1P */ +#define WM8993_IN1L_TO_LINEOUT1P_SHIFT 1 /* IN1L_TO_LINEOUT1P */ +#define WM8993_IN1L_TO_LINEOUT1P_WIDTH 1 /* IN1L_TO_LINEOUT1P */ +#define WM8993_MIXOUTL_TO_LINEOUT1P 0x0001 /* MIXOUTL_TO_LINEOUT1P */ +#define WM8993_MIXOUTL_TO_LINEOUT1P_MASK 0x0001 /* MIXOUTL_TO_LINEOUT1P */ +#define WM8993_MIXOUTL_TO_LINEOUT1P_SHIFT 0 /* MIXOUTL_TO_LINEOUT1P */ +#define WM8993_MIXOUTL_TO_LINEOUT1P_WIDTH 1 /* MIXOUTL_TO_LINEOUT1P */ + +/* + * R53 (0x35) - Line Mixer2 + */ +#define WM8993_MIXOUTR_TO_LINEOUT2N 0x0040 /* MIXOUTR_TO_LINEOUT2N */ +#define WM8993_MIXOUTR_TO_LINEOUT2N_MASK 0x0040 /* MIXOUTR_TO_LINEOUT2N */ +#define WM8993_MIXOUTR_TO_LINEOUT2N_SHIFT 6 /* MIXOUTR_TO_LINEOUT2N */ +#define WM8993_MIXOUTR_TO_LINEOUT2N_WIDTH 1 /* MIXOUTR_TO_LINEOUT2N */ +#define WM8993_MIXOUTL_TO_LINEOUT2N 0x0020 /* MIXOUTL_TO_LINEOUT2N */ +#define WM8993_MIXOUTL_TO_LINEOUT2N_MASK 0x0020 /* MIXOUTL_TO_LINEOUT2N */ +#define WM8993_MIXOUTL_TO_LINEOUT2N_SHIFT 5 /* MIXOUTL_TO_LINEOUT2N */ +#define WM8993_MIXOUTL_TO_LINEOUT2N_WIDTH 1 /* MIXOUTL_TO_LINEOUT2N */ +#define WM8993_LINEOUT2_MODE 0x0010 /* LINEOUT2_MODE */ +#define WM8993_LINEOUT2_MODE_MASK 0x0010 /* LINEOUT2_MODE */ +#define WM8993_LINEOUT2_MODE_SHIFT 4 /* LINEOUT2_MODE */ +#define WM8993_LINEOUT2_MODE_WIDTH 1 /* LINEOUT2_MODE */ +#define WM8993_IN1L_TO_LINEOUT2P 0x0004 /* IN1L_TO_LINEOUT2P */ +#define WM8993_IN1L_TO_LINEOUT2P_MASK 0x0004 /* IN1L_TO_LINEOUT2P */ +#define WM8993_IN1L_TO_LINEOUT2P_SHIFT 2 /* IN1L_TO_LINEOUT2P */ +#define WM8993_IN1L_TO_LINEOUT2P_WIDTH 1 /* IN1L_TO_LINEOUT2P */ +#define WM8993_IN1R_TO_LINEOUT2P 0x0002 /* IN1R_TO_LINEOUT2P */ +#define WM8993_IN1R_TO_LINEOUT2P_MASK 0x0002 /* IN1R_TO_LINEOUT2P */ +#define WM8993_IN1R_TO_LINEOUT2P_SHIFT 1 /* IN1R_TO_LINEOUT2P */ +#define WM8993_IN1R_TO_LINEOUT2P_WIDTH 1 /* IN1R_TO_LINEOUT2P */ +#define WM8993_MIXOUTR_TO_LINEOUT2P 0x0001 /* MIXOUTR_TO_LINEOUT2P */ +#define WM8993_MIXOUTR_TO_LINEOUT2P_MASK 0x0001 /* MIXOUTR_TO_LINEOUT2P */ +#define WM8993_MIXOUTR_TO_LINEOUT2P_SHIFT 0 /* MIXOUTR_TO_LINEOUT2P */ +#define WM8993_MIXOUTR_TO_LINEOUT2P_WIDTH 1 /* MIXOUTR_TO_LINEOUT2P */ + +/* + * R54 (0x36) - Speaker Mixer + */ +#define WM8993_SPKAB_REF_SEL 0x0100 /* SPKAB_REF_SEL */ +#define WM8993_SPKAB_REF_SEL_MASK 0x0100 /* SPKAB_REF_SEL */ +#define WM8993_SPKAB_REF_SEL_SHIFT 8 /* SPKAB_REF_SEL */ +#define WM8993_SPKAB_REF_SEL_WIDTH 1 /* SPKAB_REF_SEL */ +#define WM8993_MIXINL_TO_SPKMIXL 0x0080 /* MIXINL_TO_SPKMIXL */ +#define WM8993_MIXINL_TO_SPKMIXL_MASK 0x0080 /* MIXINL_TO_SPKMIXL */ +#define WM8993_MIXINL_TO_SPKMIXL_SHIFT 7 /* MIXINL_TO_SPKMIXL */ +#define WM8993_MIXINL_TO_SPKMIXL_WIDTH 1 /* MIXINL_TO_SPKMIXL */ +#define WM8993_MIXINR_TO_SPKMIXR 0x0040 /* MIXINR_TO_SPKMIXR */ +#define WM8993_MIXINR_TO_SPKMIXR_MASK 0x0040 /* MIXINR_TO_SPKMIXR */ +#define WM8993_MIXINR_TO_SPKMIXR_SHIFT 6 /* MIXINR_TO_SPKMIXR */ +#define WM8993_MIXINR_TO_SPKMIXR_WIDTH 1 /* MIXINR_TO_SPKMIXR */ +#define WM8993_IN1LP_TO_SPKMIXL 0x0020 /* IN1LP_TO_SPKMIXL */ +#define WM8993_IN1LP_TO_SPKMIXL_MASK 0x0020 /* IN1LP_TO_SPKMIXL */ +#define WM8993_IN1LP_TO_SPKMIXL_SHIFT 5 /* IN1LP_TO_SPKMIXL */ +#define WM8993_IN1LP_TO_SPKMIXL_WIDTH 1 /* IN1LP_TO_SPKMIXL */ +#define WM8993_IN1RP_TO_SPKMIXR 0x0010 /* IN1RP_TO_SPKMIXR */ +#define WM8993_IN1RP_TO_SPKMIXR_MASK 0x0010 /* IN1RP_TO_SPKMIXR */ +#define WM8993_IN1RP_TO_SPKMIXR_SHIFT 4 /* IN1RP_TO_SPKMIXR */ +#define WM8993_IN1RP_TO_SPKMIXR_WIDTH 1 /* IN1RP_TO_SPKMIXR */ +#define WM8993_MIXOUTL_TO_SPKMIXL 0x0008 /* MIXOUTL_TO_SPKMIXL */ +#define WM8993_MIXOUTL_TO_SPKMIXL_MASK 0x0008 /* MIXOUTL_TO_SPKMIXL */ +#define WM8993_MIXOUTL_TO_SPKMIXL_SHIFT 3 /* MIXOUTL_TO_SPKMIXL */ +#define WM8993_MIXOUTL_TO_SPKMIXL_WIDTH 1 /* MIXOUTL_TO_SPKMIXL */ +#define WM8993_MIXOUTR_TO_SPKMIXR 0x0004 /* MIXOUTR_TO_SPKMIXR */ +#define WM8993_MIXOUTR_TO_SPKMIXR_MASK 0x0004 /* MIXOUTR_TO_SPKMIXR */ +#define WM8993_MIXOUTR_TO_SPKMIXR_SHIFT 2 /* MIXOUTR_TO_SPKMIXR */ +#define WM8993_MIXOUTR_TO_SPKMIXR_WIDTH 1 /* MIXOUTR_TO_SPKMIXR */ +#define WM8993_DACL_TO_SPKMIXL 0x0002 /* DACL_TO_SPKMIXL */ +#define WM8993_DACL_TO_SPKMIXL_MASK 0x0002 /* DACL_TO_SPKMIXL */ +#define WM8993_DACL_TO_SPKMIXL_SHIFT 1 /* DACL_TO_SPKMIXL */ +#define WM8993_DACL_TO_SPKMIXL_WIDTH 1 /* DACL_TO_SPKMIXL */ +#define WM8993_DACR_TO_SPKMIXR 0x0001 /* DACR_TO_SPKMIXR */ +#define WM8993_DACR_TO_SPKMIXR_MASK 0x0001 /* DACR_TO_SPKMIXR */ +#define WM8993_DACR_TO_SPKMIXR_SHIFT 0 /* DACR_TO_SPKMIXR */ +#define WM8993_DACR_TO_SPKMIXR_WIDTH 1 /* DACR_TO_SPKMIXR */ + +/* + * R55 (0x37) - Additional Control + */ +#define WM8993_LINEOUT1_FB 0x0080 /* LINEOUT1_FB */ +#define WM8993_LINEOUT1_FB_MASK 0x0080 /* LINEOUT1_FB */ +#define WM8993_LINEOUT1_FB_SHIFT 7 /* LINEOUT1_FB */ +#define WM8993_LINEOUT1_FB_WIDTH 1 /* LINEOUT1_FB */ +#define WM8993_LINEOUT2_FB 0x0040 /* LINEOUT2_FB */ +#define WM8993_LINEOUT2_FB_MASK 0x0040 /* LINEOUT2_FB */ +#define WM8993_LINEOUT2_FB_SHIFT 6 /* LINEOUT2_FB */ +#define WM8993_LINEOUT2_FB_WIDTH 1 /* LINEOUT2_FB */ +#define WM8993_VROI 0x0001 /* VROI */ +#define WM8993_VROI_MASK 0x0001 /* VROI */ +#define WM8993_VROI_SHIFT 0 /* VROI */ +#define WM8993_VROI_WIDTH 1 /* VROI */ + +/* + * R56 (0x38) - AntiPOP1 + */ +#define WM8993_LINEOUT_VMID_BUF_ENA 0x0080 /* LINEOUT_VMID_BUF_ENA */ +#define WM8993_LINEOUT_VMID_BUF_ENA_MASK 0x0080 /* LINEOUT_VMID_BUF_ENA */ +#define WM8993_LINEOUT_VMID_BUF_ENA_SHIFT 7 /* LINEOUT_VMID_BUF_ENA */ +#define WM8993_LINEOUT_VMID_BUF_ENA_WIDTH 1 /* LINEOUT_VMID_BUF_ENA */ +#define WM8993_HPOUT2_IN_ENA 0x0040 /* HPOUT2_IN_ENA */ +#define WM8993_HPOUT2_IN_ENA_MASK 0x0040 /* HPOUT2_IN_ENA */ +#define WM8993_HPOUT2_IN_ENA_SHIFT 6 /* HPOUT2_IN_ENA */ +#define WM8993_HPOUT2_IN_ENA_WIDTH 1 /* HPOUT2_IN_ENA */ +#define WM8993_LINEOUT1_DISCH 0x0020 /* LINEOUT1_DISCH */ +#define WM8993_LINEOUT1_DISCH_MASK 0x0020 /* LINEOUT1_DISCH */ +#define WM8993_LINEOUT1_DISCH_SHIFT 5 /* LINEOUT1_DISCH */ +#define WM8993_LINEOUT1_DISCH_WIDTH 1 /* LINEOUT1_DISCH */ +#define WM8993_LINEOUT2_DISCH 0x0010 /* LINEOUT2_DISCH */ +#define WM8993_LINEOUT2_DISCH_MASK 0x0010 /* LINEOUT2_DISCH */ +#define WM8993_LINEOUT2_DISCH_SHIFT 4 /* LINEOUT2_DISCH */ +#define WM8993_LINEOUT2_DISCH_WIDTH 1 /* LINEOUT2_DISCH */ + +/* + * R57 (0x39) - AntiPOP2 + */ +#define WM8993_VMID_RAMP_MASK 0x0060 /* VMID_RAMP - [6:5] */ +#define WM8993_VMID_RAMP_SHIFT 5 /* VMID_RAMP - [6:5] */ +#define WM8993_VMID_RAMP_WIDTH 2 /* VMID_RAMP - [6:5] */ +#define WM8993_VMID_BUF_ENA 0x0008 /* VMID_BUF_ENA */ +#define WM8993_VMID_BUF_ENA_MASK 0x0008 /* VMID_BUF_ENA */ +#define WM8993_VMID_BUF_ENA_SHIFT 3 /* VMID_BUF_ENA */ +#define WM8993_VMID_BUF_ENA_WIDTH 1 /* VMID_BUF_ENA */ +#define WM8993_STARTUP_BIAS_ENA 0x0004 /* STARTUP_BIAS_ENA */ +#define WM8993_STARTUP_BIAS_ENA_MASK 0x0004 /* STARTUP_BIAS_ENA */ +#define WM8993_STARTUP_BIAS_ENA_SHIFT 2 /* STARTUP_BIAS_ENA */ +#define WM8993_STARTUP_BIAS_ENA_WIDTH 1 /* STARTUP_BIAS_ENA */ +#define WM8993_BIAS_SRC 0x0002 /* BIAS_SRC */ +#define WM8993_BIAS_SRC_MASK 0x0002 /* BIAS_SRC */ +#define WM8993_BIAS_SRC_SHIFT 1 /* BIAS_SRC */ +#define WM8993_BIAS_SRC_WIDTH 1 /* BIAS_SRC */ +#define WM8993_VMID_DISCH 0x0001 /* VMID_DISCH */ +#define WM8993_VMID_DISCH_MASK 0x0001 /* VMID_DISCH */ +#define WM8993_VMID_DISCH_SHIFT 0 /* VMID_DISCH */ +#define WM8993_VMID_DISCH_WIDTH 1 /* VMID_DISCH */ + +/* + * R58 (0x3A) - MICBIAS + */ +#define WM8993_JD_SCTHR_MASK 0x00C0 /* JD_SCTHR - [7:6] */ +#define WM8993_JD_SCTHR_SHIFT 6 /* JD_SCTHR - [7:6] */ +#define WM8993_JD_SCTHR_WIDTH 2 /* JD_SCTHR - [7:6] */ +#define WM8993_JD_THR_MASK 0x0030 /* JD_THR - [5:4] */ +#define WM8993_JD_THR_SHIFT 4 /* JD_THR - [5:4] */ +#define WM8993_JD_THR_WIDTH 2 /* JD_THR - [5:4] */ +#define WM8993_JD_ENA 0x0004 /* JD_ENA */ +#define WM8993_JD_ENA_MASK 0x0004 /* JD_ENA */ +#define WM8993_JD_ENA_SHIFT 2 /* JD_ENA */ +#define WM8993_JD_ENA_WIDTH 1 /* JD_ENA */ +#define WM8993_MICB2_LVL 0x0002 /* MICB2_LVL */ +#define WM8993_MICB2_LVL_MASK 0x0002 /* MICB2_LVL */ +#define WM8993_MICB2_LVL_SHIFT 1 /* MICB2_LVL */ +#define WM8993_MICB2_LVL_WIDTH 1 /* MICB2_LVL */ +#define WM8993_MICB1_LVL 0x0001 /* MICB1_LVL */ +#define WM8993_MICB1_LVL_MASK 0x0001 /* MICB1_LVL */ +#define WM8993_MICB1_LVL_SHIFT 0 /* MICB1_LVL */ +#define WM8993_MICB1_LVL_WIDTH 1 /* MICB1_LVL */ + +/* + * R60 (0x3C) - FLL Control 1 + */ +#define WM8993_FLL_FRAC 0x0004 /* FLL_FRAC */ +#define WM8993_FLL_FRAC_MASK 0x0004 /* FLL_FRAC */ +#define WM8993_FLL_FRAC_SHIFT 2 /* FLL_FRAC */ +#define WM8993_FLL_FRAC_WIDTH 1 /* FLL_FRAC */ +#define WM8993_FLL_OSC_ENA 0x0002 /* FLL_OSC_ENA */ +#define WM8993_FLL_OSC_ENA_MASK 0x0002 /* FLL_OSC_ENA */ +#define WM8993_FLL_OSC_ENA_SHIFT 1 /* FLL_OSC_ENA */ +#define WM8993_FLL_OSC_ENA_WIDTH 1 /* FLL_OSC_ENA */ +#define WM8993_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM8993_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM8993_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM8993_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R61 (0x3D) - FLL Control 2 + */ +#define WM8993_FLL_OUTDIV_MASK 0x0700 /* FLL_OUTDIV - [10:8] */ +#define WM8993_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [10:8] */ +#define WM8993_FLL_OUTDIV_WIDTH 3 /* FLL_OUTDIV - [10:8] */ +#define WM8993_FLL_CTRL_RATE_MASK 0x0070 /* FLL_CTRL_RATE - [6:4] */ +#define WM8993_FLL_CTRL_RATE_SHIFT 4 /* FLL_CTRL_RATE - [6:4] */ +#define WM8993_FLL_CTRL_RATE_WIDTH 3 /* FLL_CTRL_RATE - [6:4] */ +#define WM8993_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM8993_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM8993_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R62 (0x3E) - FLL Control 3 + */ +#define WM8993_FLL_K_MASK 0xFFFF /* FLL_K - [15:0] */ +#define WM8993_FLL_K_SHIFT 0 /* FLL_K - [15:0] */ +#define WM8993_FLL_K_WIDTH 16 /* FLL_K - [15:0] */ + +/* + * R63 (0x3F) - FLL Control 4 + */ +#define WM8993_FLL_N_MASK 0x7FE0 /* FLL_N - [14:5] */ +#define WM8993_FLL_N_SHIFT 5 /* FLL_N - [14:5] */ +#define WM8993_FLL_N_WIDTH 10 /* FLL_N - [14:5] */ +#define WM8993_FLL_GAIN_MASK 0x000F /* FLL_GAIN - [3:0] */ +#define WM8993_FLL_GAIN_SHIFT 0 /* FLL_GAIN - [3:0] */ +#define WM8993_FLL_GAIN_WIDTH 4 /* FLL_GAIN - [3:0] */ + +/* + * R64 (0x40) - FLL Control 5 + */ +#define WM8993_FLL_FRC_NCO_VAL_MASK 0x1F80 /* FLL_FRC_NCO_VAL - [12:7] */ +#define WM8993_FLL_FRC_NCO_VAL_SHIFT 7 /* FLL_FRC_NCO_VAL - [12:7] */ +#define WM8993_FLL_FRC_NCO_VAL_WIDTH 6 /* FLL_FRC_NCO_VAL - [12:7] */ +#define WM8993_FLL_FRC_NCO 0x0040 /* FLL_FRC_NCO */ +#define WM8993_FLL_FRC_NCO_MASK 0x0040 /* FLL_FRC_NCO */ +#define WM8993_FLL_FRC_NCO_SHIFT 6 /* FLL_FRC_NCO */ +#define WM8993_FLL_FRC_NCO_WIDTH 1 /* FLL_FRC_NCO */ +#define WM8993_FLL_CLK_REF_DIV_MASK 0x0018 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8993_FLL_CLK_REF_DIV_SHIFT 3 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8993_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [4:3] */ +#define WM8993_FLL_CLK_SRC_MASK 0x0003 /* FLL_CLK_SRC - [1:0] */ +#define WM8993_FLL_CLK_SRC_SHIFT 0 /* FLL_CLK_SRC - [1:0] */ +#define WM8993_FLL_CLK_SRC_WIDTH 2 /* FLL_CLK_SRC - [1:0] */ + +/* + * R65 (0x41) - Clocking 3 + */ +#define WM8993_CLK_DCS_DIV_MASK 0x3C00 /* CLK_DCS_DIV - [13:10] */ +#define WM8993_CLK_DCS_DIV_SHIFT 10 /* CLK_DCS_DIV - [13:10] */ +#define WM8993_CLK_DCS_DIV_WIDTH 4 /* CLK_DCS_DIV - [13:10] */ +#define WM8993_SAMPLE_RATE_MASK 0x0380 /* SAMPLE_RATE - [9:7] */ +#define WM8993_SAMPLE_RATE_SHIFT 7 /* SAMPLE_RATE - [9:7] */ +#define WM8993_SAMPLE_RATE_WIDTH 3 /* SAMPLE_RATE - [9:7] */ +#define WM8993_CLK_SYS_RATE_MASK 0x001E /* CLK_SYS_RATE - [4:1] */ +#define WM8993_CLK_SYS_RATE_SHIFT 1 /* CLK_SYS_RATE - [4:1] */ +#define WM8993_CLK_SYS_RATE_WIDTH 4 /* CLK_SYS_RATE - [4:1] */ +#define WM8993_CLK_DSP_ENA 0x0001 /* CLK_DSP_ENA */ +#define WM8993_CLK_DSP_ENA_MASK 0x0001 /* CLK_DSP_ENA */ +#define WM8993_CLK_DSP_ENA_SHIFT 0 /* CLK_DSP_ENA */ +#define WM8993_CLK_DSP_ENA_WIDTH 1 /* CLK_DSP_ENA */ + +/* + * R66 (0x42) - Clocking 4 + */ +#define WM8993_DAC_DIV4 0x0200 /* DAC_DIV4 */ +#define WM8993_DAC_DIV4_MASK 0x0200 /* DAC_DIV4 */ +#define WM8993_DAC_DIV4_SHIFT 9 /* DAC_DIV4 */ +#define WM8993_DAC_DIV4_WIDTH 1 /* DAC_DIV4 */ +#define WM8993_CLK_256K_DIV_MASK 0x007E /* CLK_256K_DIV - [6:1] */ +#define WM8993_CLK_256K_DIV_SHIFT 1 /* CLK_256K_DIV - [6:1] */ +#define WM8993_CLK_256K_DIV_WIDTH 6 /* CLK_256K_DIV - [6:1] */ +#define WM8993_SR_MODE 0x0001 /* SR_MODE */ +#define WM8993_SR_MODE_MASK 0x0001 /* SR_MODE */ +#define WM8993_SR_MODE_SHIFT 0 /* SR_MODE */ +#define WM8993_SR_MODE_WIDTH 1 /* SR_MODE */ + +/* + * R67 (0x43) - MW Slave Control + */ +#define WM8993_MASK_WRITE_ENA 0x0001 /* MASK_WRITE_ENA */ +#define WM8993_MASK_WRITE_ENA_MASK 0x0001 /* MASK_WRITE_ENA */ +#define WM8993_MASK_WRITE_ENA_SHIFT 0 /* MASK_WRITE_ENA */ +#define WM8993_MASK_WRITE_ENA_WIDTH 1 /* MASK_WRITE_ENA */ + +/* + * R69 (0x45) - Bus Control 1 + */ +#define WM8993_CLK_SYS_ENA 0x0002 /* CLK_SYS_ENA */ +#define WM8993_CLK_SYS_ENA_MASK 0x0002 /* CLK_SYS_ENA */ +#define WM8993_CLK_SYS_ENA_SHIFT 1 /* CLK_SYS_ENA */ +#define WM8993_CLK_SYS_ENA_WIDTH 1 /* CLK_SYS_ENA */ + +/* + * R70 (0x46) - Write Sequencer 0 + */ +#define WM8993_WSEQ_ENA 0x0100 /* WSEQ_ENA */ +#define WM8993_WSEQ_ENA_MASK 0x0100 /* WSEQ_ENA */ +#define WM8993_WSEQ_ENA_SHIFT 8 /* WSEQ_ENA */ +#define WM8993_WSEQ_ENA_WIDTH 1 /* WSEQ_ENA */ +#define WM8993_WSEQ_WRITE_INDEX_MASK 0x001F /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8993_WSEQ_WRITE_INDEX_SHIFT 0 /* WSEQ_WRITE_INDEX - [4:0] */ +#define WM8993_WSEQ_WRITE_INDEX_WIDTH 5 /* WSEQ_WRITE_INDEX - [4:0] */ + +/* + * R71 (0x47) - Write Sequencer 1 + */ +#define WM8993_WSEQ_DATA_WIDTH_MASK 0x7000 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8993_WSEQ_DATA_WIDTH_SHIFT 12 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8993_WSEQ_DATA_WIDTH_WIDTH 3 /* WSEQ_DATA_WIDTH - [14:12] */ +#define WM8993_WSEQ_DATA_START_MASK 0x0F00 /* WSEQ_DATA_START - [11:8] */ +#define WM8993_WSEQ_DATA_START_SHIFT 8 /* WSEQ_DATA_START - [11:8] */ +#define WM8993_WSEQ_DATA_START_WIDTH 4 /* WSEQ_DATA_START - [11:8] */ +#define WM8993_WSEQ_ADDR_MASK 0x00FF /* WSEQ_ADDR - [7:0] */ +#define WM8993_WSEQ_ADDR_SHIFT 0 /* WSEQ_ADDR - [7:0] */ +#define WM8993_WSEQ_ADDR_WIDTH 8 /* WSEQ_ADDR - [7:0] */ + +/* + * R72 (0x48) - Write Sequencer 2 + */ +#define WM8993_WSEQ_EOS 0x4000 /* WSEQ_EOS */ +#define WM8993_WSEQ_EOS_MASK 0x4000 /* WSEQ_EOS */ +#define WM8993_WSEQ_EOS_SHIFT 14 /* WSEQ_EOS */ +#define WM8993_WSEQ_EOS_WIDTH 1 /* WSEQ_EOS */ +#define WM8993_WSEQ_DELAY_MASK 0x0F00 /* WSEQ_DELAY - [11:8] */ +#define WM8993_WSEQ_DELAY_SHIFT 8 /* WSEQ_DELAY - [11:8] */ +#define WM8993_WSEQ_DELAY_WIDTH 4 /* WSEQ_DELAY - [11:8] */ +#define WM8993_WSEQ_DATA_MASK 0x00FF /* WSEQ_DATA - [7:0] */ +#define WM8993_WSEQ_DATA_SHIFT 0 /* WSEQ_DATA - [7:0] */ +#define WM8993_WSEQ_DATA_WIDTH 8 /* WSEQ_DATA - [7:0] */ + +/* + * R73 (0x49) - Write Sequencer 3 + */ +#define WM8993_WSEQ_ABORT 0x0200 /* WSEQ_ABORT */ +#define WM8993_WSEQ_ABORT_MASK 0x0200 /* WSEQ_ABORT */ +#define WM8993_WSEQ_ABORT_SHIFT 9 /* WSEQ_ABORT */ +#define WM8993_WSEQ_ABORT_WIDTH 1 /* WSEQ_ABORT */ +#define WM8993_WSEQ_START 0x0100 /* WSEQ_START */ +#define WM8993_WSEQ_START_MASK 0x0100 /* WSEQ_START */ +#define WM8993_WSEQ_START_SHIFT 8 /* WSEQ_START */ +#define WM8993_WSEQ_START_WIDTH 1 /* WSEQ_START */ +#define WM8993_WSEQ_START_INDEX_MASK 0x003F /* WSEQ_START_INDEX - [5:0] */ +#define WM8993_WSEQ_START_INDEX_SHIFT 0 /* WSEQ_START_INDEX - [5:0] */ +#define WM8993_WSEQ_START_INDEX_WIDTH 6 /* WSEQ_START_INDEX - [5:0] */ + +/* + * R74 (0x4A) - Write Sequencer 4 + */ +#define WM8993_WSEQ_BUSY 0x0001 /* WSEQ_BUSY */ +#define WM8993_WSEQ_BUSY_MASK 0x0001 /* WSEQ_BUSY */ +#define WM8993_WSEQ_BUSY_SHIFT 0 /* WSEQ_BUSY */ +#define WM8993_WSEQ_BUSY_WIDTH 1 /* WSEQ_BUSY */ + +/* + * R75 (0x4B) - Write Sequencer 5 + */ +#define WM8993_WSEQ_CURRENT_INDEX_MASK 0x003F /* WSEQ_CURRENT_INDEX - [5:0] */ +#define WM8993_WSEQ_CURRENT_INDEX_SHIFT 0 /* WSEQ_CURRENT_INDEX - [5:0] */ +#define WM8993_WSEQ_CURRENT_INDEX_WIDTH 6 /* WSEQ_CURRENT_INDEX - [5:0] */ + +/* + * R76 (0x4C) - Charge Pump 1 + */ +#define WM8993_CP_ENA 0x8000 /* CP_ENA */ +#define WM8993_CP_ENA_MASK 0x8000 /* CP_ENA */ +#define WM8993_CP_ENA_SHIFT 15 /* CP_ENA */ +#define WM8993_CP_ENA_WIDTH 1 /* CP_ENA */ + +/* + * R81 (0x51) - Class W 0 + */ +#define WM8993_CP_DYN_FREQ 0x0002 /* CP_DYN_FREQ */ +#define WM8993_CP_DYN_FREQ_MASK 0x0002 /* CP_DYN_FREQ */ +#define WM8993_CP_DYN_FREQ_SHIFT 1 /* CP_DYN_FREQ */ +#define WM8993_CP_DYN_FREQ_WIDTH 1 /* CP_DYN_FREQ */ +#define WM8993_CP_DYN_V 0x0001 /* CP_DYN_V */ +#define WM8993_CP_DYN_V_MASK 0x0001 /* CP_DYN_V */ +#define WM8993_CP_DYN_V_SHIFT 0 /* CP_DYN_V */ +#define WM8993_CP_DYN_V_WIDTH 1 /* CP_DYN_V */ + +/* + * R84 (0x54) - DC Servo 0 + */ +#define WM8993_DCS_TRIG_SINGLE_1 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8993_DCS_TRIG_SINGLE_1_MASK 0x2000 /* DCS_TRIG_SINGLE_1 */ +#define WM8993_DCS_TRIG_SINGLE_1_SHIFT 13 /* DCS_TRIG_SINGLE_1 */ +#define WM8993_DCS_TRIG_SINGLE_1_WIDTH 1 /* DCS_TRIG_SINGLE_1 */ +#define WM8993_DCS_TRIG_SINGLE_0 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8993_DCS_TRIG_SINGLE_0_MASK 0x1000 /* DCS_TRIG_SINGLE_0 */ +#define WM8993_DCS_TRIG_SINGLE_0_SHIFT 12 /* DCS_TRIG_SINGLE_0 */ +#define WM8993_DCS_TRIG_SINGLE_0_WIDTH 1 /* DCS_TRIG_SINGLE_0 */ +#define WM8993_DCS_TRIG_SERIES_1 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8993_DCS_TRIG_SERIES_1_MASK 0x0200 /* DCS_TRIG_SERIES_1 */ +#define WM8993_DCS_TRIG_SERIES_1_SHIFT 9 /* DCS_TRIG_SERIES_1 */ +#define WM8993_DCS_TRIG_SERIES_1_WIDTH 1 /* DCS_TRIG_SERIES_1 */ +#define WM8993_DCS_TRIG_SERIES_0 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8993_DCS_TRIG_SERIES_0_MASK 0x0100 /* DCS_TRIG_SERIES_0 */ +#define WM8993_DCS_TRIG_SERIES_0_SHIFT 8 /* DCS_TRIG_SERIES_0 */ +#define WM8993_DCS_TRIG_SERIES_0_WIDTH 1 /* DCS_TRIG_SERIES_0 */ +#define WM8993_DCS_TRIG_STARTUP_1 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8993_DCS_TRIG_STARTUP_1_MASK 0x0020 /* DCS_TRIG_STARTUP_1 */ +#define WM8993_DCS_TRIG_STARTUP_1_SHIFT 5 /* DCS_TRIG_STARTUP_1 */ +#define WM8993_DCS_TRIG_STARTUP_1_WIDTH 1 /* DCS_TRIG_STARTUP_1 */ +#define WM8993_DCS_TRIG_STARTUP_0 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8993_DCS_TRIG_STARTUP_0_MASK 0x0010 /* DCS_TRIG_STARTUP_0 */ +#define WM8993_DCS_TRIG_STARTUP_0_SHIFT 4 /* DCS_TRIG_STARTUP_0 */ +#define WM8993_DCS_TRIG_STARTUP_0_WIDTH 1 /* DCS_TRIG_STARTUP_0 */ +#define WM8993_DCS_TRIG_DAC_WR_1 0x0008 /* DCS_TRIG_DAC_WR_1 */ +#define WM8993_DCS_TRIG_DAC_WR_1_MASK 0x0008 /* DCS_TRIG_DAC_WR_1 */ +#define WM8993_DCS_TRIG_DAC_WR_1_SHIFT 3 /* DCS_TRIG_DAC_WR_1 */ +#define WM8993_DCS_TRIG_DAC_WR_1_WIDTH 1 /* DCS_TRIG_DAC_WR_1 */ +#define WM8993_DCS_TRIG_DAC_WR_0 0x0004 /* DCS_TRIG_DAC_WR_0 */ +#define WM8993_DCS_TRIG_DAC_WR_0_MASK 0x0004 /* DCS_TRIG_DAC_WR_0 */ +#define WM8993_DCS_TRIG_DAC_WR_0_SHIFT 2 /* DCS_TRIG_DAC_WR_0 */ +#define WM8993_DCS_TRIG_DAC_WR_0_WIDTH 1 /* DCS_TRIG_DAC_WR_0 */ +#define WM8993_DCS_ENA_CHAN_1 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8993_DCS_ENA_CHAN_1_MASK 0x0002 /* DCS_ENA_CHAN_1 */ +#define WM8993_DCS_ENA_CHAN_1_SHIFT 1 /* DCS_ENA_CHAN_1 */ +#define WM8993_DCS_ENA_CHAN_1_WIDTH 1 /* DCS_ENA_CHAN_1 */ +#define WM8993_DCS_ENA_CHAN_0 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8993_DCS_ENA_CHAN_0_MASK 0x0001 /* DCS_ENA_CHAN_0 */ +#define WM8993_DCS_ENA_CHAN_0_SHIFT 0 /* DCS_ENA_CHAN_0 */ +#define WM8993_DCS_ENA_CHAN_0_WIDTH 1 /* DCS_ENA_CHAN_0 */ + +/* + * R85 (0x55) - DC Servo 1 + */ +#define WM8993_DCS_SERIES_NO_01_MASK 0x0FE0 /* DCS_SERIES_NO_01 - [11:5] */ +#define WM8993_DCS_SERIES_NO_01_SHIFT 5 /* DCS_SERIES_NO_01 - [11:5] */ +#define WM8993_DCS_SERIES_NO_01_WIDTH 7 /* DCS_SERIES_NO_01 - [11:5] */ +#define WM8993_DCS_TIMER_PERIOD_01_MASK 0x000F /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8993_DCS_TIMER_PERIOD_01_SHIFT 0 /* DCS_TIMER_PERIOD_01 - [3:0] */ +#define WM8993_DCS_TIMER_PERIOD_01_WIDTH 4 /* DCS_TIMER_PERIOD_01 - [3:0] */ + +/* + * R87 (0x57) - DC Servo 3 + */ +#define WM8993_DCS_DAC_WR_VAL_1_MASK 0xFF00 /* DCS_DAC_WR_VAL_1 - [15:8] */ +#define WM8993_DCS_DAC_WR_VAL_1_SHIFT 8 /* DCS_DAC_WR_VAL_1 - [15:8] */ +#define WM8993_DCS_DAC_WR_VAL_1_WIDTH 8 /* DCS_DAC_WR_VAL_1 - [15:8] */ +#define WM8993_DCS_DAC_WR_VAL_0_MASK 0x00FF /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8993_DCS_DAC_WR_VAL_0_SHIFT 0 /* DCS_DAC_WR_VAL_0 - [7:0] */ +#define WM8993_DCS_DAC_WR_VAL_0_WIDTH 8 /* DCS_DAC_WR_VAL_0 - [7:0] */ + +/* + * R88 (0x58) - DC Servo Readback 0 + */ +#define WM8993_DCS_DATAPATH_BUSY 0x4000 /* DCS_DATAPATH_BUSY */ +#define WM8993_DCS_DATAPATH_BUSY_MASK 0x4000 /* DCS_DATAPATH_BUSY */ +#define WM8993_DCS_DATAPATH_BUSY_SHIFT 14 /* DCS_DATAPATH_BUSY */ +#define WM8993_DCS_DATAPATH_BUSY_WIDTH 1 /* DCS_DATAPATH_BUSY */ +#define WM8993_DCS_CHANNEL_MASK 0x3000 /* DCS_CHANNEL - [13:12] */ +#define WM8993_DCS_CHANNEL_SHIFT 12 /* DCS_CHANNEL - [13:12] */ +#define WM8993_DCS_CHANNEL_WIDTH 2 /* DCS_CHANNEL - [13:12] */ +#define WM8993_DCS_CAL_COMPLETE_MASK 0x0300 /* DCS_CAL_COMPLETE - [9:8] */ +#define WM8993_DCS_CAL_COMPLETE_SHIFT 8 /* DCS_CAL_COMPLETE - [9:8] */ +#define WM8993_DCS_CAL_COMPLETE_WIDTH 2 /* DCS_CAL_COMPLETE - [9:8] */ +#define WM8993_DCS_DAC_WR_COMPLETE_MASK 0x0030 /* DCS_DAC_WR_COMPLETE - [5:4] */ +#define WM8993_DCS_DAC_WR_COMPLETE_SHIFT 4 /* DCS_DAC_WR_COMPLETE - [5:4] */ +#define WM8993_DCS_DAC_WR_COMPLETE_WIDTH 2 /* DCS_DAC_WR_COMPLETE - [5:4] */ +#define WM8993_DCS_STARTUP_COMPLETE_MASK 0x0003 /* DCS_STARTUP_COMPLETE - [1:0] */ +#define WM8993_DCS_STARTUP_COMPLETE_SHIFT 0 /* DCS_STARTUP_COMPLETE - [1:0] */ +#define WM8993_DCS_STARTUP_COMPLETE_WIDTH 2 /* DCS_STARTUP_COMPLETE - [1:0] */ + +/* + * R89 (0x59) - DC Servo Readback 1 + */ +#define WM8993_DCS_INTEG_CHAN_1_MASK 0x00FF /* DCS_INTEG_CHAN_1 - [7:0] */ +#define WM8993_DCS_INTEG_CHAN_1_SHIFT 0 /* DCS_INTEG_CHAN_1 - [7:0] */ +#define WM8993_DCS_INTEG_CHAN_1_WIDTH 8 /* DCS_INTEG_CHAN_1 - [7:0] */ + +/* + * R90 (0x5A) - DC Servo Readback 2 + */ +#define WM8993_DCS_INTEG_CHAN_0_MASK 0x00FF /* DCS_INTEG_CHAN_0 - [7:0] */ +#define WM8993_DCS_INTEG_CHAN_0_SHIFT 0 /* DCS_INTEG_CHAN_0 - [7:0] */ +#define WM8993_DCS_INTEG_CHAN_0_WIDTH 8 /* DCS_INTEG_CHAN_0 - [7:0] */ + +/* + * R96 (0x60) - Analogue HP 0 + */ +#define WM8993_HPOUT1_AUTO_PU 0x0100 /* HPOUT1_AUTO_PU */ +#define WM8993_HPOUT1_AUTO_PU_MASK 0x0100 /* HPOUT1_AUTO_PU */ +#define WM8993_HPOUT1_AUTO_PU_SHIFT 8 /* HPOUT1_AUTO_PU */ +#define WM8993_HPOUT1_AUTO_PU_WIDTH 1 /* HPOUT1_AUTO_PU */ +#define WM8993_HPOUT1L_RMV_SHORT 0x0080 /* HPOUT1L_RMV_SHORT */ +#define WM8993_HPOUT1L_RMV_SHORT_MASK 0x0080 /* HPOUT1L_RMV_SHORT */ +#define WM8993_HPOUT1L_RMV_SHORT_SHIFT 7 /* HPOUT1L_RMV_SHORT */ +#define WM8993_HPOUT1L_RMV_SHORT_WIDTH 1 /* HPOUT1L_RMV_SHORT */ +#define WM8993_HPOUT1L_OUTP 0x0040 /* HPOUT1L_OUTP */ +#define WM8993_HPOUT1L_OUTP_MASK 0x0040 /* HPOUT1L_OUTP */ +#define WM8993_HPOUT1L_OUTP_SHIFT 6 /* HPOUT1L_OUTP */ +#define WM8993_HPOUT1L_OUTP_WIDTH 1 /* HPOUT1L_OUTP */ +#define WM8993_HPOUT1L_DLY 0x0020 /* HPOUT1L_DLY */ +#define WM8993_HPOUT1L_DLY_MASK 0x0020 /* HPOUT1L_DLY */ +#define WM8993_HPOUT1L_DLY_SHIFT 5 /* HPOUT1L_DLY */ +#define WM8993_HPOUT1L_DLY_WIDTH 1 /* HPOUT1L_DLY */ +#define WM8993_HPOUT1R_RMV_SHORT 0x0008 /* HPOUT1R_RMV_SHORT */ +#define WM8993_HPOUT1R_RMV_SHORT_MASK 0x0008 /* HPOUT1R_RMV_SHORT */ +#define WM8993_HPOUT1R_RMV_SHORT_SHIFT 3 /* HPOUT1R_RMV_SHORT */ +#define WM8993_HPOUT1R_RMV_SHORT_WIDTH 1 /* HPOUT1R_RMV_SHORT */ +#define WM8993_HPOUT1R_OUTP 0x0004 /* HPOUT1R_OUTP */ +#define WM8993_HPOUT1R_OUTP_MASK 0x0004 /* HPOUT1R_OUTP */ +#define WM8993_HPOUT1R_OUTP_SHIFT 2 /* HPOUT1R_OUTP */ +#define WM8993_HPOUT1R_OUTP_WIDTH 1 /* HPOUT1R_OUTP */ +#define WM8993_HPOUT1R_DLY 0x0002 /* HPOUT1R_DLY */ +#define WM8993_HPOUT1R_DLY_MASK 0x0002 /* HPOUT1R_DLY */ +#define WM8993_HPOUT1R_DLY_SHIFT 1 /* HPOUT1R_DLY */ +#define WM8993_HPOUT1R_DLY_WIDTH 1 /* HPOUT1R_DLY */ + +/* + * R98 (0x62) - EQ1 + */ +#define WM8993_EQ_ENA 0x0001 /* EQ_ENA */ +#define WM8993_EQ_ENA_MASK 0x0001 /* EQ_ENA */ +#define WM8993_EQ_ENA_SHIFT 0 /* EQ_ENA */ +#define WM8993_EQ_ENA_WIDTH 1 /* EQ_ENA */ + +/* + * R99 (0x63) - EQ2 + */ +#define WM8993_EQ_B1_GAIN_MASK 0x001F /* EQ_B1_GAIN - [4:0] */ +#define WM8993_EQ_B1_GAIN_SHIFT 0 /* EQ_B1_GAIN - [4:0] */ +#define WM8993_EQ_B1_GAIN_WIDTH 5 /* EQ_B1_GAIN - [4:0] */ + +/* + * R100 (0x64) - EQ3 + */ +#define WM8993_EQ_B2_GAIN_MASK 0x001F /* EQ_B2_GAIN - [4:0] */ +#define WM8993_EQ_B2_GAIN_SHIFT 0 /* EQ_B2_GAIN - [4:0] */ +#define WM8993_EQ_B2_GAIN_WIDTH 5 /* EQ_B2_GAIN - [4:0] */ + +/* + * R101 (0x65) - EQ4 + */ +#define WM8993_EQ_B3_GAIN_MASK 0x001F /* EQ_B3_GAIN - [4:0] */ +#define WM8993_EQ_B3_GAIN_SHIFT 0 /* EQ_B3_GAIN - [4:0] */ +#define WM8993_EQ_B3_GAIN_WIDTH 5 /* EQ_B3_GAIN - [4:0] */ + +/* + * R102 (0x66) - EQ5 + */ +#define WM8993_EQ_B4_GAIN_MASK 0x001F /* EQ_B4_GAIN - [4:0] */ +#define WM8993_EQ_B4_GAIN_SHIFT 0 /* EQ_B4_GAIN - [4:0] */ +#define WM8993_EQ_B4_GAIN_WIDTH 5 /* EQ_B4_GAIN - [4:0] */ + +/* + * R103 (0x67) - EQ6 + */ +#define WM8993_EQ_B5_GAIN_MASK 0x001F /* EQ_B5_GAIN - [4:0] */ +#define WM8993_EQ_B5_GAIN_SHIFT 0 /* EQ_B5_GAIN - [4:0] */ +#define WM8993_EQ_B5_GAIN_WIDTH 5 /* EQ_B5_GAIN - [4:0] */ + +/* + * R104 (0x68) - EQ7 + */ +#define WM8993_EQ_B1_A_MASK 0xFFFF /* EQ_B1_A - [15:0] */ +#define WM8993_EQ_B1_A_SHIFT 0 /* EQ_B1_A - [15:0] */ +#define WM8993_EQ_B1_A_WIDTH 16 /* EQ_B1_A - [15:0] */ + +/* + * R105 (0x69) - EQ8 + */ +#define WM8993_EQ_B1_B_MASK 0xFFFF /* EQ_B1_B - [15:0] */ +#define WM8993_EQ_B1_B_SHIFT 0 /* EQ_B1_B - [15:0] */ +#define WM8993_EQ_B1_B_WIDTH 16 /* EQ_B1_B - [15:0] */ + +/* + * R106 (0x6A) - EQ9 + */ +#define WM8993_EQ_B1_PG_MASK 0xFFFF /* EQ_B1_PG - [15:0] */ +#define WM8993_EQ_B1_PG_SHIFT 0 /* EQ_B1_PG - [15:0] */ +#define WM8993_EQ_B1_PG_WIDTH 16 /* EQ_B1_PG - [15:0] */ + +/* + * R107 (0x6B) - EQ10 + */ +#define WM8993_EQ_B2_A_MASK 0xFFFF /* EQ_B2_A - [15:0] */ +#define WM8993_EQ_B2_A_SHIFT 0 /* EQ_B2_A - [15:0] */ +#define WM8993_EQ_B2_A_WIDTH 16 /* EQ_B2_A - [15:0] */ + +/* + * R108 (0x6C) - EQ11 + */ +#define WM8993_EQ_B2_B_MASK 0xFFFF /* EQ_B2_B - [15:0] */ +#define WM8993_EQ_B2_B_SHIFT 0 /* EQ_B2_B - [15:0] */ +#define WM8993_EQ_B2_B_WIDTH 16 /* EQ_B2_B - [15:0] */ + +/* + * R109 (0x6D) - EQ12 + */ +#define WM8993_EQ_B2_C_MASK 0xFFFF /* EQ_B2_C - [15:0] */ +#define WM8993_EQ_B2_C_SHIFT 0 /* EQ_B2_C - [15:0] */ +#define WM8993_EQ_B2_C_WIDTH 16 /* EQ_B2_C - [15:0] */ + +/* + * R110 (0x6E) - EQ13 + */ +#define WM8993_EQ_B2_PG_MASK 0xFFFF /* EQ_B2_PG - [15:0] */ +#define WM8993_EQ_B2_PG_SHIFT 0 /* EQ_B2_PG - [15:0] */ +#define WM8993_EQ_B2_PG_WIDTH 16 /* EQ_B2_PG - [15:0] */ + +/* + * R111 (0x6F) - EQ14 + */ +#define WM8993_EQ_B3_A_MASK 0xFFFF /* EQ_B3_A - [15:0] */ +#define WM8993_EQ_B3_A_SHIFT 0 /* EQ_B3_A - [15:0] */ +#define WM8993_EQ_B3_A_WIDTH 16 /* EQ_B3_A - [15:0] */ + +/* + * R112 (0x70) - EQ15 + */ +#define WM8993_EQ_B3_B_MASK 0xFFFF /* EQ_B3_B - [15:0] */ +#define WM8993_EQ_B3_B_SHIFT 0 /* EQ_B3_B - [15:0] */ +#define WM8993_EQ_B3_B_WIDTH 16 /* EQ_B3_B - [15:0] */ + +/* + * R113 (0x71) - EQ16 + */ +#define WM8993_EQ_B3_C_MASK 0xFFFF /* EQ_B3_C - [15:0] */ +#define WM8993_EQ_B3_C_SHIFT 0 /* EQ_B3_C - [15:0] */ +#define WM8993_EQ_B3_C_WIDTH 16 /* EQ_B3_C - [15:0] */ + +/* + * R114 (0x72) - EQ17 + */ +#define WM8993_EQ_B3_PG_MASK 0xFFFF /* EQ_B3_PG - [15:0] */ +#define WM8993_EQ_B3_PG_SHIFT 0 /* EQ_B3_PG - [15:0] */ +#define WM8993_EQ_B3_PG_WIDTH 16 /* EQ_B3_PG - [15:0] */ + +/* + * R115 (0x73) - EQ18 + */ +#define WM8993_EQ_B4_A_MASK 0xFFFF /* EQ_B4_A - [15:0] */ +#define WM8993_EQ_B4_A_SHIFT 0 /* EQ_B4_A - [15:0] */ +#define WM8993_EQ_B4_A_WIDTH 16 /* EQ_B4_A - [15:0] */ + +/* + * R116 (0x74) - EQ19 + */ +#define WM8993_EQ_B4_B_MASK 0xFFFF /* EQ_B4_B - [15:0] */ +#define WM8993_EQ_B4_B_SHIFT 0 /* EQ_B4_B - [15:0] */ +#define WM8993_EQ_B4_B_WIDTH 16 /* EQ_B4_B - [15:0] */ + +/* + * R117 (0x75) - EQ20 + */ +#define WM8993_EQ_B4_C_MASK 0xFFFF /* EQ_B4_C - [15:0] */ +#define WM8993_EQ_B4_C_SHIFT 0 /* EQ_B4_C - [15:0] */ +#define WM8993_EQ_B4_C_WIDTH 16 /* EQ_B4_C - [15:0] */ + +/* + * R118 (0x76) - EQ21 + */ +#define WM8993_EQ_B4_PG_MASK 0xFFFF /* EQ_B4_PG - [15:0] */ +#define WM8993_EQ_B4_PG_SHIFT 0 /* EQ_B4_PG - [15:0] */ +#define WM8993_EQ_B4_PG_WIDTH 16 /* EQ_B4_PG - [15:0] */ + +/* + * R119 (0x77) - EQ22 + */ +#define WM8993_EQ_B5_A_MASK 0xFFFF /* EQ_B5_A - [15:0] */ +#define WM8993_EQ_B5_A_SHIFT 0 /* EQ_B5_A - [15:0] */ +#define WM8993_EQ_B5_A_WIDTH 16 /* EQ_B5_A - [15:0] */ + +/* + * R120 (0x78) - EQ23 + */ +#define WM8993_EQ_B5_B_MASK 0xFFFF /* EQ_B5_B - [15:0] */ +#define WM8993_EQ_B5_B_SHIFT 0 /* EQ_B5_B - [15:0] */ +#define WM8993_EQ_B5_B_WIDTH 16 /* EQ_B5_B - [15:0] */ + +/* + * R121 (0x79) - EQ24 + */ +#define WM8993_EQ_B5_PG_MASK 0xFFFF /* EQ_B5_PG - [15:0] */ +#define WM8993_EQ_B5_PG_SHIFT 0 /* EQ_B5_PG - [15:0] */ +#define WM8993_EQ_B5_PG_WIDTH 16 /* EQ_B5_PG - [15:0] */ + +/* + * R122 (0x7A) - Digital Pulls + */ +#define WM8993_MCLK_PU 0x0080 /* MCLK_PU */ +#define WM8993_MCLK_PU_MASK 0x0080 /* MCLK_PU */ +#define WM8993_MCLK_PU_SHIFT 7 /* MCLK_PU */ +#define WM8993_MCLK_PU_WIDTH 1 /* MCLK_PU */ +#define WM8993_MCLK_PD 0x0040 /* MCLK_PD */ +#define WM8993_MCLK_PD_MASK 0x0040 /* MCLK_PD */ +#define WM8993_MCLK_PD_SHIFT 6 /* MCLK_PD */ +#define WM8993_MCLK_PD_WIDTH 1 /* MCLK_PD */ +#define WM8993_DACDAT_PU 0x0020 /* DACDAT_PU */ +#define WM8993_DACDAT_PU_MASK 0x0020 /* DACDAT_PU */ +#define WM8993_DACDAT_PU_SHIFT 5 /* DACDAT_PU */ +#define WM8993_DACDAT_PU_WIDTH 1 /* DACDAT_PU */ +#define WM8993_DACDAT_PD 0x0010 /* DACDAT_PD */ +#define WM8993_DACDAT_PD_MASK 0x0010 /* DACDAT_PD */ +#define WM8993_DACDAT_PD_SHIFT 4 /* DACDAT_PD */ +#define WM8993_DACDAT_PD_WIDTH 1 /* DACDAT_PD */ +#define WM8993_LRCLK_PU 0x0008 /* LRCLK_PU */ +#define WM8993_LRCLK_PU_MASK 0x0008 /* LRCLK_PU */ +#define WM8993_LRCLK_PU_SHIFT 3 /* LRCLK_PU */ +#define WM8993_LRCLK_PU_WIDTH 1 /* LRCLK_PU */ +#define WM8993_LRCLK_PD 0x0004 /* LRCLK_PD */ +#define WM8993_LRCLK_PD_MASK 0x0004 /* LRCLK_PD */ +#define WM8993_LRCLK_PD_SHIFT 2 /* LRCLK_PD */ +#define WM8993_LRCLK_PD_WIDTH 1 /* LRCLK_PD */ +#define WM8993_BCLK_PU 0x0002 /* BCLK_PU */ +#define WM8993_BCLK_PU_MASK 0x0002 /* BCLK_PU */ +#define WM8993_BCLK_PU_SHIFT 1 /* BCLK_PU */ +#define WM8993_BCLK_PU_WIDTH 1 /* BCLK_PU */ +#define WM8993_BCLK_PD 0x0001 /* BCLK_PD */ +#define WM8993_BCLK_PD_MASK 0x0001 /* BCLK_PD */ +#define WM8993_BCLK_PD_SHIFT 0 /* BCLK_PD */ +#define WM8993_BCLK_PD_WIDTH 1 /* BCLK_PD */ + +/* + * R123 (0x7B) - DRC Control 1 + */ +#define WM8993_DRC_ENA 0x8000 /* DRC_ENA */ +#define WM8993_DRC_ENA_MASK 0x8000 /* DRC_ENA */ +#define WM8993_DRC_ENA_SHIFT 15 /* DRC_ENA */ +#define WM8993_DRC_ENA_WIDTH 1 /* DRC_ENA */ +#define WM8993_DRC_DAC_PATH 0x4000 /* DRC_DAC_PATH */ +#define WM8993_DRC_DAC_PATH_MASK 0x4000 /* DRC_DAC_PATH */ +#define WM8993_DRC_DAC_PATH_SHIFT 14 /* DRC_DAC_PATH */ +#define WM8993_DRC_DAC_PATH_WIDTH 1 /* DRC_DAC_PATH */ +#define WM8993_DRC_SMOOTH_ENA 0x0800 /* DRC_SMOOTH_ENA */ +#define WM8993_DRC_SMOOTH_ENA_MASK 0x0800 /* DRC_SMOOTH_ENA */ +#define WM8993_DRC_SMOOTH_ENA_SHIFT 11 /* DRC_SMOOTH_ENA */ +#define WM8993_DRC_SMOOTH_ENA_WIDTH 1 /* DRC_SMOOTH_ENA */ +#define WM8993_DRC_QR_ENA 0x0400 /* DRC_QR_ENA */ +#define WM8993_DRC_QR_ENA_MASK 0x0400 /* DRC_QR_ENA */ +#define WM8993_DRC_QR_ENA_SHIFT 10 /* DRC_QR_ENA */ +#define WM8993_DRC_QR_ENA_WIDTH 1 /* DRC_QR_ENA */ +#define WM8993_DRC_ANTICLIP_ENA 0x0200 /* DRC_ANTICLIP_ENA */ +#define WM8993_DRC_ANTICLIP_ENA_MASK 0x0200 /* DRC_ANTICLIP_ENA */ +#define WM8993_DRC_ANTICLIP_ENA_SHIFT 9 /* DRC_ANTICLIP_ENA */ +#define WM8993_DRC_ANTICLIP_ENA_WIDTH 1 /* DRC_ANTICLIP_ENA */ +#define WM8993_DRC_HYST_ENA 0x0100 /* DRC_HYST_ENA */ +#define WM8993_DRC_HYST_ENA_MASK 0x0100 /* DRC_HYST_ENA */ +#define WM8993_DRC_HYST_ENA_SHIFT 8 /* DRC_HYST_ENA */ +#define WM8993_DRC_HYST_ENA_WIDTH 1 /* DRC_HYST_ENA */ +#define WM8993_DRC_THRESH_HYST_MASK 0x0030 /* DRC_THRESH_HYST - [5:4] */ +#define WM8993_DRC_THRESH_HYST_SHIFT 4 /* DRC_THRESH_HYST - [5:4] */ +#define WM8993_DRC_THRESH_HYST_WIDTH 2 /* DRC_THRESH_HYST - [5:4] */ +#define WM8993_DRC_MINGAIN_MASK 0x000C /* DRC_MINGAIN - [3:2] */ +#define WM8993_DRC_MINGAIN_SHIFT 2 /* DRC_MINGAIN - [3:2] */ +#define WM8993_DRC_MINGAIN_WIDTH 2 /* DRC_MINGAIN - [3:2] */ +#define WM8993_DRC_MAXGAIN_MASK 0x0003 /* DRC_MAXGAIN - [1:0] */ +#define WM8993_DRC_MAXGAIN_SHIFT 0 /* DRC_MAXGAIN - [1:0] */ +#define WM8993_DRC_MAXGAIN_WIDTH 2 /* DRC_MAXGAIN - [1:0] */ + +/* + * R124 (0x7C) - DRC Control 2 + */ +#define WM8993_DRC_ATTACK_RATE_MASK 0xF000 /* DRC_ATTACK_RATE - [15:12] */ +#define WM8993_DRC_ATTACK_RATE_SHIFT 12 /* DRC_ATTACK_RATE - [15:12] */ +#define WM8993_DRC_ATTACK_RATE_WIDTH 4 /* DRC_ATTACK_RATE - [15:12] */ +#define WM8993_DRC_DECAY_RATE_MASK 0x0F00 /* DRC_DECAY_RATE - [11:8] */ +#define WM8993_DRC_DECAY_RATE_SHIFT 8 /* DRC_DECAY_RATE - [11:8] */ +#define WM8993_DRC_DECAY_RATE_WIDTH 4 /* DRC_DECAY_RATE - [11:8] */ +#define WM8993_DRC_THRESH_COMP_MASK 0x00FC /* DRC_THRESH_COMP - [7:2] */ +#define WM8993_DRC_THRESH_COMP_SHIFT 2 /* DRC_THRESH_COMP - [7:2] */ +#define WM8993_DRC_THRESH_COMP_WIDTH 6 /* DRC_THRESH_COMP - [7:2] */ + +/* + * R125 (0x7D) - DRC Control 3 + */ +#define WM8993_DRC_AMP_COMP_MASK 0xF800 /* DRC_AMP_COMP - [15:11] */ +#define WM8993_DRC_AMP_COMP_SHIFT 11 /* DRC_AMP_COMP - [15:11] */ +#define WM8993_DRC_AMP_COMP_WIDTH 5 /* DRC_AMP_COMP - [15:11] */ +#define WM8993_DRC_R0_SLOPE_COMP_MASK 0x0700 /* DRC_R0_SLOPE_COMP - [10:8] */ +#define WM8993_DRC_R0_SLOPE_COMP_SHIFT 8 /* DRC_R0_SLOPE_COMP - [10:8] */ +#define WM8993_DRC_R0_SLOPE_COMP_WIDTH 3 /* DRC_R0_SLOPE_COMP - [10:8] */ +#define WM8993_DRC_FF_DELAY 0x0080 /* DRC_FF_DELAY */ +#define WM8993_DRC_FF_DELAY_MASK 0x0080 /* DRC_FF_DELAY */ +#define WM8993_DRC_FF_DELAY_SHIFT 7 /* DRC_FF_DELAY */ +#define WM8993_DRC_FF_DELAY_WIDTH 1 /* DRC_FF_DELAY */ +#define WM8993_DRC_THRESH_QR_MASK 0x000C /* DRC_THRESH_QR - [3:2] */ +#define WM8993_DRC_THRESH_QR_SHIFT 2 /* DRC_THRESH_QR - [3:2] */ +#define WM8993_DRC_THRESH_QR_WIDTH 2 /* DRC_THRESH_QR - [3:2] */ +#define WM8993_DRC_RATE_QR_MASK 0x0003 /* DRC_RATE_QR - [1:0] */ +#define WM8993_DRC_RATE_QR_SHIFT 0 /* DRC_RATE_QR - [1:0] */ +#define WM8993_DRC_RATE_QR_WIDTH 2 /* DRC_RATE_QR - [1:0] */ + +/* + * R126 (0x7E) - DRC Control 4 + */ +#define WM8993_DRC_R1_SLOPE_COMP_MASK 0xE000 /* DRC_R1_SLOPE_COMP - [15:13] */ +#define WM8993_DRC_R1_SLOPE_COMP_SHIFT 13 /* DRC_R1_SLOPE_COMP - [15:13] */ +#define WM8993_DRC_R1_SLOPE_COMP_WIDTH 3 /* DRC_R1_SLOPE_COMP - [15:13] */ +#define WM8993_DRC_STARTUP_GAIN_MASK 0x1F00 /* DRC_STARTUP_GAIN - [12:8] */ +#define WM8993_DRC_STARTUP_GAIN_SHIFT 8 /* DRC_STARTUP_GAIN - [12:8] */ +#define WM8993_DRC_STARTUP_GAIN_WIDTH 5 /* DRC_STARTUP_GAIN - [12:8] */ + +#endif diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e25f97..c64e55aa63b6 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -165,87 +165,23 @@ struct wm9081_priv { int master; int fll_fref; int fll_fout; + int tdm_width; struct wm9081_retune_mobile_config *retune; }; -static int wm9081_reg_is_volatile(int reg) +static int wm9081_volatile_register(unsigned int reg) { switch (reg) { + case WM9081_SOFTWARE_RESET: + return 1; default: return 0; } } -static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - BUG_ON(reg > WM9081_MAX_REGISTER); - return cache[reg]; -} - -static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg) -{ - struct i2c_msg xfer[2]; - u16 data; - int ret; - struct i2c_client *client = codec->control_data; - - BUG_ON(reg > WM9081_MAX_REGISTER); - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = 1; - xfer[0].buf = ® - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = 2; - xfer[1].buf = (u8 *)&data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret != 2) { - dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); - return 0; - } - - return (data >> 8) | ((data & 0xff) << 8); -} - -static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg) -{ - if (wm9081_reg_is_volatile(reg)) - return wm9081_read_hw(codec, reg); - else - return wm9081_read_reg_cache(codec, reg); -} - -static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 *cache = codec->reg_cache; - u8 data[3]; - - BUG_ON(reg > WM9081_MAX_REGISTER); - - if (!wm9081_reg_is_volatile(reg)) - cache[reg] = value; - - data[0] = reg; - data[1] = value >> 8; - data[2] = value & 0x00ff; - - if (codec->hw_write(codec->control_data, data, 3) == 3) - return 0; - else - return -EIO; -} - static int wm9081_reset(struct snd_soc_codec *codec) { - return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0); + return snd_soc_write(codec, WM9081_SOFTWARE_RESET, 0); } static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0); @@ -356,7 +292,7 @@ static int speaker_mode_get(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg; - reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2); + reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_2); if (reg & WM9081_SPK_MODE) ucontrol->value.integer.value[0] = 1; else @@ -375,8 +311,8 @@ static int speaker_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT); - unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2); + unsigned int reg_pwr = snd_soc_read(codec, WM9081_POWER_MANAGEMENT); + unsigned int reg2 = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_2); /* Are we changing anything? */ if (ucontrol->value.integer.value[0] == @@ -397,7 +333,7 @@ static int speaker_mode_put(struct snd_kcontrol *kcontrol, reg2 &= ~WM9081_SPK_MODE; } - wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2); + snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2); return 0; } @@ -456,7 +392,7 @@ static int speaker_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT); + unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -468,7 +404,7 @@ static int speaker_event(struct snd_soc_dapm_widget *w, break; } - wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg); + snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg); return 0; } @@ -607,7 +543,7 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id, if (ret != 0) return ret; - reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5); + reg5 = snd_soc_read(codec, WM9081_FLL_CONTROL_5); reg5 &= ~WM9081_FLL_CLK_SRC_MASK; switch (fll_id) { @@ -621,44 +557,44 @@ static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id, } /* Disable CLK_SYS while we reconfigure */ - clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3); + clk_sys_reg = snd_soc_read(codec, WM9081_CLOCK_CONTROL_3); if (clk_sys_reg & WM9081_CLK_SYS_ENA) - wm9081_write(codec, WM9081_CLOCK_CONTROL_3, + snd_soc_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg & ~WM9081_CLK_SYS_ENA); /* Any FLL configuration change requires that the FLL be * disabled first. */ - reg1 = wm9081_read(codec, WM9081_FLL_CONTROL_1); + reg1 = snd_soc_read(codec, WM9081_FLL_CONTROL_1); reg1 &= ~WM9081_FLL_ENA; - wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1); /* Apply the configuration */ if (fll_div.k) reg1 |= WM9081_FLL_FRAC_MASK; else reg1 &= ~WM9081_FLL_FRAC_MASK; - wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1); + snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1); - wm9081_write(codec, WM9081_FLL_CONTROL_2, + snd_soc_write(codec, WM9081_FLL_CONTROL_2, (fll_div.fll_outdiv << WM9081_FLL_OUTDIV_SHIFT) | (fll_div.fll_fratio << WM9081_FLL_FRATIO_SHIFT)); - wm9081_write(codec, WM9081_FLL_CONTROL_3, fll_div.k); + snd_soc_write(codec, WM9081_FLL_CONTROL_3, fll_div.k); - reg4 = wm9081_read(codec, WM9081_FLL_CONTROL_4); + reg4 = snd_soc_read(codec, WM9081_FLL_CONTROL_4); reg4 &= ~WM9081_FLL_N_MASK; reg4 |= fll_div.n << WM9081_FLL_N_SHIFT; - wm9081_write(codec, WM9081_FLL_CONTROL_4, reg4); + snd_soc_write(codec, WM9081_FLL_CONTROL_4, reg4); reg5 &= ~WM9081_FLL_CLK_REF_DIV_MASK; reg5 |= fll_div.fll_clk_ref_div << WM9081_FLL_CLK_REF_DIV_SHIFT; - wm9081_write(codec, WM9081_FLL_CONTROL_5, reg5); + snd_soc_write(codec, WM9081_FLL_CONTROL_5, reg5); /* Enable the FLL */ - wm9081_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA); + snd_soc_write(codec, WM9081_FLL_CONTROL_1, reg1 | WM9081_FLL_ENA); /* Then bring CLK_SYS up again if it was disabled */ if (clk_sys_reg & WM9081_CLK_SYS_ENA) - wm9081_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg); + snd_soc_write(codec, WM9081_CLOCK_CONTROL_3, clk_sys_reg); dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); @@ -707,6 +643,10 @@ static int configure_clock(struct snd_soc_codec *codec) target > 3000000) break; } + + if (i == ARRAY_SIZE(clk_sys_rates)) + return -EINVAL; + } else if (wm9081->fs) { for (i = 0; i < ARRAY_SIZE(clk_sys_rates); i++) { new_sysclk = clk_sys_rates[i].ratio @@ -714,6 +654,10 @@ static int configure_clock(struct snd_soc_codec *codec) if (new_sysclk > 3000000) break; } + + if (i == ARRAY_SIZE(clk_sys_rates)) + return -EINVAL; + } else { new_sysclk = 12288000; } @@ -734,19 +678,19 @@ static int configure_clock(struct snd_soc_codec *codec) return -EINVAL; } - reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_1); + reg = snd_soc_read(codec, WM9081_CLOCK_CONTROL_1); if (mclkdiv) reg |= WM9081_MCLKDIV2; else reg &= ~WM9081_MCLKDIV2; - wm9081_write(codec, WM9081_CLOCK_CONTROL_1, reg); + snd_soc_write(codec, WM9081_CLOCK_CONTROL_1, reg); - reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3); + reg = snd_soc_read(codec, WM9081_CLOCK_CONTROL_3); if (fll) reg |= WM9081_CLK_SRC_SEL; else reg &= ~WM9081_CLK_SRC_SEL; - wm9081_write(codec, WM9081_CLOCK_CONTROL_3, reg); + snd_soc_write(codec, WM9081_CLOCK_CONTROL_3, reg); dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm9081->sysclk_rate); @@ -846,76 +790,76 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x2; - wm9081_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Normal bias current */ - reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_STBY_BIAS_ENA; - wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL); + reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); reg &= ~WM9081_LINEOUT_DISCH; - wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); /* Select startup bias source */ - reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); /* VMID 2*4k; Soft VMID ramp enable */ - reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg |= WM9081_VMID_RAMP | 0x6; - wm9081_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_write(codec, WM9081_VMID_CONTROL, reg); mdelay(100); /* Normal bias enable & soft start off */ reg |= WM9081_BIAS_ENA; reg &= ~WM9081_VMID_RAMP; - wm9081_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Standard bias source */ - reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_BIAS_SRC; - wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); } /* VMID 2*240k */ - reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg &= ~WM9081_VMID_SEL_MASK; reg |= 0x40; - wm9081_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Standby bias current on */ - reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_STBY_BIAS_ENA; - wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); break; case SND_SOC_BIAS_OFF: /* Startup bias source */ - reg = wm9081_read(codec, WM9081_BIAS_CONTROL_1); + reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); reg |= WM9081_BIAS_SRC; - wm9081_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); /* Disable VMID and biases with soft ramping */ - reg = wm9081_read(codec, WM9081_VMID_CONTROL); + reg = snd_soc_read(codec, WM9081_VMID_CONTROL); reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); reg |= WM9081_VMID_RAMP; - wm9081_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_write(codec, WM9081_VMID_CONTROL, reg); /* Actively discharge LINEOUT */ - reg = wm9081_read(codec, WM9081_ANTI_POP_CONTROL); + reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); reg |= WM9081_LINEOUT_DISCH; - wm9081_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); break; } @@ -929,7 +873,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, { struct snd_soc_codec *codec = dai->codec; struct wm9081_priv *wm9081 = codec->private_data; - unsigned int aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2); + unsigned int aif2 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_2); aif2 &= ~(WM9081_AIF_BCLK_INV | WM9081_AIF_LRCLK_INV | WM9081_BCLK_DIR | WM9081_LRCLK_DIR | WM9081_AIF_FMT_MASK); @@ -1010,7 +954,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, return -EINVAL; } - wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); + snd_soc_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); return 0; } @@ -1024,47 +968,51 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, int ret, i, best, best_val, cur_val; unsigned int clk_ctrl2, aif1, aif2, aif3, aif4; - clk_ctrl2 = wm9081_read(codec, WM9081_CLOCK_CONTROL_2); + clk_ctrl2 = snd_soc_read(codec, WM9081_CLOCK_CONTROL_2); clk_ctrl2 &= ~(WM9081_CLK_SYS_RATE_MASK | WM9081_SAMPLE_RATE_MASK); - aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); + aif1 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_1); - aif2 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_2); + aif2 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_2); aif2 &= ~WM9081_AIF_WL_MASK; - aif3 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_3); + aif3 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_3); aif3 &= ~WM9081_BCLK_DIV_MASK; - aif4 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_4); + aif4 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_4); aif4 &= ~WM9081_LRCLK_RATE_MASK; - /* What BCLK do we need? */ wm9081->fs = params_rate(params); - wm9081->bclk = 2 * wm9081->fs; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - wm9081->bclk *= 16; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - wm9081->bclk *= 20; - aif2 |= 0x4; - break; - case SNDRV_PCM_FORMAT_S24_LE: - wm9081->bclk *= 24; - aif2 |= 0x8; - break; - case SNDRV_PCM_FORMAT_S32_LE: - wm9081->bclk *= 32; - aif2 |= 0xc; - break; - default: - return -EINVAL; - } - if (aif1 & WM9081_AIFDAC_TDM_MODE_MASK) { + if (wm9081->tdm_width) { + /* If TDM is set up then that fixes our BCLK. */ int slots = ((aif1 & WM9081_AIFDAC_TDM_MODE_MASK) >> WM9081_AIFDAC_TDM_MODE_SHIFT) + 1; - wm9081->bclk *= slots; + + wm9081->bclk = wm9081->fs * wm9081->tdm_width * slots; + } else { + /* Otherwise work out a BCLK from the sample size */ + wm9081->bclk = 2 * wm9081->fs; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + wm9081->bclk *= 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + wm9081->bclk *= 20; + aif2 |= 0x4; + break; + case SNDRV_PCM_FORMAT_S24_LE: + wm9081->bclk *= 24; + aif2 |= 0x8; + break; + case SNDRV_PCM_FORMAT_S32_LE: + wm9081->bclk *= 32; + aif2 |= 0xc; + break; + default: + return -EINVAL; + } } dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm9081->bclk); @@ -1149,22 +1097,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, s->name, s->rate); /* If the EQ is enabled then disable it while we write out */ - eq1 = wm9081_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA; + eq1 = snd_soc_read(codec, WM9081_EQ_1) & WM9081_EQ_ENA; if (eq1 & WM9081_EQ_ENA) - wm9081_write(codec, WM9081_EQ_1, 0); + snd_soc_write(codec, WM9081_EQ_1, 0); /* Write out the other values */ for (i = 1; i < ARRAY_SIZE(s->config); i++) - wm9081_write(codec, WM9081_EQ_1 + i, s->config[i]); + snd_soc_write(codec, WM9081_EQ_1 + i, s->config[i]); eq1 |= (s->config[0] & ~WM9081_EQ_ENA); - wm9081_write(codec, WM9081_EQ_1, eq1); + snd_soc_write(codec, WM9081_EQ_1, eq1); } - wm9081_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2); - wm9081_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); - wm9081_write(codec, WM9081_AUDIO_INTERFACE_3, aif3); - wm9081_write(codec, WM9081_AUDIO_INTERFACE_4, aif4); + snd_soc_write(codec, WM9081_CLOCK_CONTROL_2, clk_ctrl2); + snd_soc_write(codec, WM9081_AUDIO_INTERFACE_2, aif2); + snd_soc_write(codec, WM9081_AUDIO_INTERFACE_3, aif3); + snd_soc_write(codec, WM9081_AUDIO_INTERFACE_4, aif4); return 0; } @@ -1174,14 +1122,14 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) struct snd_soc_codec *codec = codec_dai->codec; unsigned int reg; - reg = wm9081_read(codec, WM9081_DAC_DIGITAL_2); + reg = snd_soc_read(codec, WM9081_DAC_DIGITAL_2); if (mute) reg |= WM9081_DAC_MUTE; else reg &= ~WM9081_DAC_MUTE; - wm9081_write(codec, WM9081_DAC_DIGITAL_2, reg); + snd_soc_write(codec, WM9081_DAC_DIGITAL_2, reg); return 0; } @@ -1207,19 +1155,25 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, } static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); + struct wm9081_priv *wm9081 = codec->private_data; + unsigned int aif1 = snd_soc_read(codec, WM9081_AUDIO_INTERFACE_1); aif1 &= ~(WM9081_AIFDAC_TDM_SLOT_MASK | WM9081_AIFDAC_TDM_MODE_MASK); - if (slots < 1 || slots > 4) + if (slots < 0 || slots > 4) return -EINVAL; + wm9081->tdm_width = slot_width; + + if (slots == 0) + slots = 1; + aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: @@ -1235,7 +1189,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, return -EINVAL; } - wm9081_write(codec, WM9081_AUDIO_INTERFACE_1, aif1); + snd_soc_write(codec, WM9081_AUDIO_INTERFACE_1, aif1); return 0; } @@ -1357,7 +1311,7 @@ static int wm9081_resume(struct platform_device *pdev) if (i == WM9081_SOFTWARE_RESET) continue; - wm9081_write(codec, i, reg_cache[i]); + snd_soc_write(codec, i, reg_cache[i]); } wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1377,7 +1331,8 @@ struct snd_soc_codec_device soc_codec_dev_wm9081 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm9081); -static int wm9081_register(struct wm9081_priv *wm9081) +static int wm9081_register(struct wm9081_priv *wm9081, + enum snd_soc_control_type control) { struct snd_soc_codec *codec = &wm9081->codec; int ret; @@ -1396,19 +1351,24 @@ static int wm9081_register(struct wm9081_priv *wm9081) codec->private_data = wm9081; codec->name = "WM9081"; codec->owner = THIS_MODULE; - codec->read = wm9081_read; - codec->write = wm9081_write; codec->dai = &wm9081_dai; codec->num_dai = 1; codec->reg_cache_size = ARRAY_SIZE(wm9081->reg_cache); codec->reg_cache = &wm9081->reg_cache; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm9081_set_bias_level; + codec->volatile_register = wm9081_volatile_register; memcpy(codec->reg_cache, wm9081_reg_defaults, sizeof(wm9081_reg_defaults)); - reg = wm9081_read_hw(codec, WM9081_SOFTWARE_RESET); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET); if (reg != 0x9081) { dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg); ret = -EINVAL; @@ -1424,10 +1384,10 @@ static int wm9081_register(struct wm9081_priv *wm9081) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = wm9081_read(codec, WM9081_ANALOGUE_LINEOUT); - wm9081_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, + reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); + snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); + reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); + snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, reg | WM9081_SPKPGAZC); wm9081_dai.dev = codec->dev; @@ -1482,7 +1442,7 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, codec->dev = &i2c->dev; - return wm9081_register(wm9081); + return wm9081_register(wm9081, SND_SOC_I2C); } static __devexit int wm9081_i2c_remove(struct i2c_client *client) @@ -1492,6 +1452,21 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM +static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg) +{ + return snd_soc_suspend_device(&client->dev); +} + +static int wm9081_i2c_resume(struct i2c_client *client) +{ + return snd_soc_resume_device(&client->dev); +} +#else +#define wm9081_i2c_suspend NULL +#define wm9081_i2c_resume NULL +#endif + static const struct i2c_device_id wm9081_i2c_id[] = { { "wm9081", 0 }, { } @@ -1505,6 +1480,8 @@ static struct i2c_driver wm9081_i2c_driver = { }, .probe = wm9081_i2c_probe, .remove = __devexit_p(wm9081_i2c_remove), + .suspend = wm9081_i2c_suspend, + .resume = wm9081_i2c_resume, .id_table = wm9081_i2c_id, }; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index fa88b463e71f..e7d2840d9e59 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -406,7 +406,7 @@ static int wm9705_soc_probe(struct platform_device *pdev) ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm9705: failed to register card\n"); - goto pcm_err; + goto reset_err; } return 0; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c new file mode 100644 index 000000000000..e542027eea89 --- /dev/null +++ b/sound/soc/codecs/wm_hubs.c @@ -0,0 +1,743 @@ +/* + * wm_hubs.c -- WM8993/4 common code + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8993.h" +#include "wm_hubs.h" + +const DECLARE_TLV_DB_SCALE(wm_hubs_spkmix_tlv, -300, 300, 0); +EXPORT_SYMBOL_GPL(wm_hubs_spkmix_tlv); + +static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1650, 150, 0); +static const DECLARE_TLV_DB_SCALE(inmix_sw_tlv, 0, 3000, 0); +static const DECLARE_TLV_DB_SCALE(inmix_tlv, -1500, 300, 1); +static const DECLARE_TLV_DB_SCALE(earpiece_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0); +static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1); +static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0); +static const unsigned int spkboost_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 6, TLV_DB_SCALE_ITEM(0, 150, 0), + 7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(line_tlv, -600, 600, 0); + +static const char *speaker_ref_text[] = { + "SPKVDD/2", + "VMID", +}; + +static const struct soc_enum speaker_ref = + SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text); + +static const char *speaker_mode_text[] = { + "Class D", + "Class AB", +}; + +static const struct soc_enum speaker_mode = + SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text); + +static void wait_for_dc_servo(struct snd_soc_codec *codec) +{ + unsigned int reg; + int count = 0; + + dev_dbg(codec->dev, "Waiting for DC servo...\n"); + do { + count++; + msleep(1); + reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); + dev_dbg(codec->dev, "DC servo status: %x\n", reg); + } while ((reg & WM8993_DCS_CAL_COMPLETE_MASK) + != WM8993_DCS_CAL_COMPLETE_MASK && count < 1000); + + if ((reg & WM8993_DCS_CAL_COMPLETE_MASK) + != WM8993_DCS_CAL_COMPLETE_MASK) + dev_err(codec->dev, "Timed out waiting for DC Servo\n"); +} + +/* + * Update the DC servo calibration on gain changes + */ +static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int ret; + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + + /* Only need to do this if the outputs are active */ + if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1) + & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA)) + snd_soc_update_bits(codec, + WM8993_DC_SERVO_0, + WM8993_DCS_TRIG_SINGLE_0 | + WM8993_DCS_TRIG_SINGLE_1, + WM8993_DCS_TRIG_SINGLE_0 | + WM8993_DCS_TRIG_SINGLE_1); + + return ret; +} + +static const struct snd_kcontrol_new analogue_snd_controls[] = { +SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, + inpga_tlv), +SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), +SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), + +SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0, + inpga_tlv), +SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1), +SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0), + + +SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, + inpga_tlv), +SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), +SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), + +SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0, + inpga_tlv), +SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1), +SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0), + +SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0, + inmix_sw_tlv), +SOC_SINGLE_TLV("MIXINL IN1L Volume", WM8993_INPUT_MIXER3, 4, 1, 0, + inmix_sw_tlv), +SOC_SINGLE_TLV("MIXINL Output Record Volume", WM8993_INPUT_MIXER3, 0, 7, 0, + inmix_tlv), +SOC_SINGLE_TLV("MIXINL IN1LP Volume", WM8993_INPUT_MIXER5, 6, 7, 0, inmix_tlv), +SOC_SINGLE_TLV("MIXINL Direct Voice Volume", WM8993_INPUT_MIXER5, 0, 6, 0, + inmix_tlv), + +SOC_SINGLE_TLV("MIXINR IN2R Volume", WM8993_INPUT_MIXER4, 7, 1, 0, + inmix_sw_tlv), +SOC_SINGLE_TLV("MIXINR IN1R Volume", WM8993_INPUT_MIXER4, 4, 1, 0, + inmix_sw_tlv), +SOC_SINGLE_TLV("MIXINR Output Record Volume", WM8993_INPUT_MIXER4, 0, 7, 0, + inmix_tlv), +SOC_SINGLE_TLV("MIXINR IN1RP Volume", WM8993_INPUT_MIXER6, 6, 7, 0, inmix_tlv), +SOC_SINGLE_TLV("MIXINR Direct Voice Volume", WM8993_INPUT_MIXER6, 0, 6, 0, + inmix_tlv), + +SOC_SINGLE_TLV("Left Output Mixer IN2RN Volume", WM8993_OUTPUT_MIXER5, 6, 7, 1, + outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer IN2LN Volume", WM8993_OUTPUT_MIXER3, 6, 7, 1, + outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer IN2LP Volume", WM8993_OUTPUT_MIXER3, 9, 7, 1, + outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer IN1L Volume", WM8993_OUTPUT_MIXER3, 0, 7, 1, + outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer IN1R Volume", WM8993_OUTPUT_MIXER3, 3, 7, 1, + outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer Right Input Volume", + WM8993_OUTPUT_MIXER5, 3, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer Left Input Volume", + WM8993_OUTPUT_MIXER5, 0, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Left Output Mixer DAC Volume", WM8993_OUTPUT_MIXER5, 9, 7, 1, + outmix_tlv), + +SOC_SINGLE_TLV("Right Output Mixer IN2LN Volume", + WM8993_OUTPUT_MIXER6, 6, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer IN2RN Volume", + WM8993_OUTPUT_MIXER4, 6, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer IN1L Volume", + WM8993_OUTPUT_MIXER4, 3, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer IN1R Volume", + WM8993_OUTPUT_MIXER4, 0, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer IN2RP Volume", + WM8993_OUTPUT_MIXER4, 9, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer Left Input Volume", + WM8993_OUTPUT_MIXER6, 3, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer Right Input Volume", + WM8993_OUTPUT_MIXER6, 6, 7, 1, outmix_tlv), +SOC_SINGLE_TLV("Right Output Mixer DAC Volume", + WM8993_OUTPUT_MIXER6, 9, 7, 1, outmix_tlv), + +SOC_DOUBLE_R_TLV("Output Volume", WM8993_LEFT_OPGA_VOLUME, + WM8993_RIGHT_OPGA_VOLUME, 0, 63, 0, outpga_tlv), +SOC_DOUBLE_R("Output Switch", WM8993_LEFT_OPGA_VOLUME, + WM8993_RIGHT_OPGA_VOLUME, 6, 1, 0), +SOC_DOUBLE_R("Output ZC Switch", WM8993_LEFT_OPGA_VOLUME, + WM8993_RIGHT_OPGA_VOLUME, 7, 1, 0), + +SOC_SINGLE("Earpiece Switch", WM8993_HPOUT2_VOLUME, 5, 1, 1), +SOC_SINGLE_TLV("Earpiece Volume", WM8993_HPOUT2_VOLUME, 4, 1, 1, earpiece_tlv), + +SOC_SINGLE_TLV("SPKL Input Volume", WM8993_SPKMIXL_ATTENUATION, + 5, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKL IN1LP Volume", WM8993_SPKMIXL_ATTENUATION, + 4, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKL Output Volume", WM8993_SPKMIXL_ATTENUATION, + 3, 1, 1, wm_hubs_spkmix_tlv), + +SOC_SINGLE_TLV("SPKR Input Volume", WM8993_SPKMIXR_ATTENUATION, + 5, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKR IN1RP Volume", WM8993_SPKMIXR_ATTENUATION, + 4, 1, 1, wm_hubs_spkmix_tlv), +SOC_SINGLE_TLV("SPKR Output Volume", WM8993_SPKMIXR_ATTENUATION, + 3, 1, 1, wm_hubs_spkmix_tlv), + +SOC_DOUBLE_R_TLV("Speaker Mixer Volume", + WM8993_SPKMIXL_ATTENUATION, WM8993_SPKMIXR_ATTENUATION, + 0, 3, 1, spkmixout_tlv), +SOC_DOUBLE_R_TLV("Speaker Volume", + WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT, + 0, 63, 0, outpga_tlv), +SOC_DOUBLE_R("Speaker Switch", + WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT, + 6, 1, 0), +SOC_DOUBLE_R("Speaker ZC Switch", + WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT, + 7, 1, 0), +SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 0, 3, 7, 0, + spkboost_tlv), +SOC_ENUM("Speaker Reference", speaker_ref), +SOC_ENUM("Speaker Mode", speaker_mode), + +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .tlv.p = outpga_tlv, + .info = snd_soc_info_volsw_2r, + .get = snd_soc_get_volsw_2r, .put = wm8993_put_dc_servo, + .private_value = (unsigned long)&(struct soc_mixer_control) { + .reg = WM8993_LEFT_OUTPUT_VOLUME, + .rreg = WM8993_RIGHT_OUTPUT_VOLUME, + .shift = 0, .max = 63 + }, +}, +SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME, + WM8993_RIGHT_OUTPUT_VOLUME, 6, 1, 0), +SOC_DOUBLE_R("Headphone ZC Switch", WM8993_LEFT_OUTPUT_VOLUME, + WM8993_RIGHT_OUTPUT_VOLUME, 7, 1, 0), + +SOC_SINGLE("LINEOUT1N Switch", WM8993_LINE_OUTPUTS_VOLUME, 6, 1, 1), +SOC_SINGLE("LINEOUT1P Switch", WM8993_LINE_OUTPUTS_VOLUME, 5, 1, 1), +SOC_SINGLE_TLV("LINEOUT1 Volume", WM8993_LINE_OUTPUTS_VOLUME, 4, 1, 1, + line_tlv), + +SOC_SINGLE("LINEOUT2N Switch", WM8993_LINE_OUTPUTS_VOLUME, 2, 1, 1), +SOC_SINGLE("LINEOUT2P Switch", WM8993_LINE_OUTPUTS_VOLUME, 1, 1, 1), +SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1, + line_tlv), +}; + +static int hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int reg = snd_soc_read(codec, WM8993_ANALOGUE_HP_0); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, + WM8993_CP_ENA, WM8993_CP_ENA); + + msleep(5); + + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA, + WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA); + + reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY; + snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); + + /* Start the DC servo */ + snd_soc_update_bits(codec, WM8993_DC_SERVO_0, + 0xFFFF, + WM8993_DCS_ENA_CHAN_0 | + WM8993_DCS_ENA_CHAN_1 | + WM8993_DCS_TRIG_STARTUP_1 | + WM8993_DCS_TRIG_STARTUP_0); + wait_for_dc_servo(codec); + + reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT | + WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT; + snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); + break; + + case SND_SOC_DAPM_PRE_PMD: + reg &= ~(WM8993_HPOUT1L_RMV_SHORT | + WM8993_HPOUT1L_DLY | + WM8993_HPOUT1L_OUTP | + WM8993_HPOUT1R_RMV_SHORT | + WM8993_HPOUT1R_DLY | + WM8993_HPOUT1R_OUTP); + + snd_soc_update_bits(codec, WM8993_DC_SERVO_0, + 0xffff, 0); + + snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1, + WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA, + 0); + + snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, + WM8993_CP_ENA, 0); + break; + } + + return 0; +} + +static int earpiece_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 reg = snd_soc_read(codec, WM8993_ANTIPOP1) & ~WM8993_HPOUT2_IN_ENA; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + reg |= WM8993_HPOUT2_IN_ENA; + snd_soc_write(codec, WM8993_ANTIPOP1, reg); + udelay(50); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(codec, WM8993_ANTIPOP1, reg); + break; + + default: + BUG(); + break; + } + + return 0; +} + +static const struct snd_kcontrol_new in1l_pga[] = { +SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), +SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), +}; + +static const struct snd_kcontrol_new in1r_pga[] = { +SOC_DAPM_SINGLE("IN1RP Switch", WM8993_INPUT_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1RN Switch", WM8993_INPUT_MIXER2, 0, 1, 0), +}; + +static const struct snd_kcontrol_new in2l_pga[] = { +SOC_DAPM_SINGLE("IN2LP Switch", WM8993_INPUT_MIXER2, 7, 1, 0), +SOC_DAPM_SINGLE("IN2LN Switch", WM8993_INPUT_MIXER2, 6, 1, 0), +}; + +static const struct snd_kcontrol_new in2r_pga[] = { +SOC_DAPM_SINGLE("IN2RP Switch", WM8993_INPUT_MIXER2, 3, 1, 0), +SOC_DAPM_SINGLE("IN2RN Switch", WM8993_INPUT_MIXER2, 2, 1, 0), +}; + +static const struct snd_kcontrol_new mixinl[] = { +SOC_DAPM_SINGLE("IN2L Switch", WM8993_INPUT_MIXER3, 8, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_INPUT_MIXER3, 5, 1, 0), +}; + +static const struct snd_kcontrol_new mixinr[] = { +SOC_DAPM_SINGLE("IN2R Switch", WM8993_INPUT_MIXER4, 8, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0), +}; + +static const struct snd_kcontrol_new left_output_mixer[] = { +SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0), +SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0), +SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0), +SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0), +SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_mixer[] = { +SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0), +SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0), +SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0), +SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0), +}; + +static const struct snd_kcontrol_new earpiece_mixer[] = { +SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_HPOUT2_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Left Output Switch", WM8993_HPOUT2_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Right Output Switch", WM8993_HPOUT2_MIXER, 3, 1, 0), +}; + +static const struct snd_kcontrol_new left_speaker_boost[] = { +SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_SPKOUT_MIXERS, 5, 1, 0), +SOC_DAPM_SINGLE("SPKL Switch", WM8993_SPKOUT_MIXERS, 4, 1, 0), +SOC_DAPM_SINGLE("SPKR Switch", WM8993_SPKOUT_MIXERS, 3, 1, 0), +}; + +static const struct snd_kcontrol_new right_speaker_boost[] = { +SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_SPKOUT_MIXERS, 2, 1, 0), +SOC_DAPM_SINGLE("SPKL Switch", WM8993_SPKOUT_MIXERS, 1, 1, 0), +SOC_DAPM_SINGLE("SPKR Switch", WM8993_SPKOUT_MIXERS, 0, 1, 0), +}; + +static const struct snd_kcontrol_new line1_mix[] = { +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER1, 2, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER1, 1, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), +}; + +static const struct snd_kcontrol_new line1n_mix[] = { +SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 6, 1, 0), +SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER1, 5, 1, 0), +}; + +static const struct snd_kcontrol_new line1p_mix[] = { +SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), +}; + +static const struct snd_kcontrol_new line2_mix[] = { +SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), +}; + +static const struct snd_kcontrol_new line2n_mix[] = { +SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0), +SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0), +}; + +static const struct snd_kcontrol_new line2p_mix[] = { +SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), +}; + +static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN1LN"), +SND_SOC_DAPM_INPUT("IN1LP"), +SND_SOC_DAPM_INPUT("IN2LN"), +SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN1RN"), +SND_SOC_DAPM_INPUT("IN1RP"), +SND_SOC_DAPM_INPUT("IN2RN"), +SND_SOC_DAPM_INPUT("IN2RP/VXRP"), + +SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), +SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), + +SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, + in1l_pga, ARRAY_SIZE(in1l_pga)), +SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, + in1r_pga, ARRAY_SIZE(in1r_pga)), + +SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0, + in2l_pga, ARRAY_SIZE(in2l_pga)), +SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0, + in2r_pga, ARRAY_SIZE(in2r_pga)), + +/* Dummy widgets to represent differential paths */ +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), + +SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0, + mixinl, ARRAY_SIZE(mixinl)), +SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0, + mixinr, ARRAY_SIZE(mixinr)), + +SND_SOC_DAPM_MIXER("Left Output Mixer", WM8993_POWER_MANAGEMENT_3, 5, 0, + left_output_mixer, ARRAY_SIZE(left_output_mixer)), +SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0, + right_output_mixer, ARRAY_SIZE(right_output_mixer)), + +SND_SOC_DAPM_PGA("Left Output PGA", WM8993_POWER_MANAGEMENT_3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0), + +SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0, + NULL, 0, + hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + +SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + earpiece_mixer, ARRAY_SIZE(earpiece_mixer)), +SND_SOC_DAPM_PGA_E("Earpiece Driver", WM8993_POWER_MANAGEMENT_1, 11, 0, + NULL, 0, earpiece_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + +SND_SOC_DAPM_MIXER("SPKL Boost", SND_SOC_NOPM, 0, 0, + left_speaker_boost, ARRAY_SIZE(left_speaker_boost)), +SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, + right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), + +SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, + NULL, 0), +SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, + NULL, 0), + +SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, + line1_mix, ARRAY_SIZE(line1_mix)), +SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0, + line2_mix, ARRAY_SIZE(line2_mix)), + +SND_SOC_DAPM_MIXER("LINEOUT1N Mixer", SND_SOC_NOPM, 0, 0, + line1n_mix, ARRAY_SIZE(line1n_mix)), +SND_SOC_DAPM_MIXER("LINEOUT1P Mixer", SND_SOC_NOPM, 0, 0, + line1p_mix, ARRAY_SIZE(line1p_mix)), +SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, + line2n_mix, ARRAY_SIZE(line2n_mix)), +SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, + line2p_mix, ARRAY_SIZE(line2p_mix)), + +SND_SOC_DAPM_PGA("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0), + +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2P"), +SND_SOC_DAPM_OUTPUT("HPOUT2N"), +SND_SOC_DAPM_OUTPUT("LINEOUT1P"), +SND_SOC_DAPM_OUTPUT("LINEOUT1N"), +SND_SOC_DAPM_OUTPUT("LINEOUT2P"), +SND_SOC_DAPM_OUTPUT("LINEOUT2N"), +}; + +static const struct snd_soc_dapm_route analogue_routes[] = { + { "IN1L PGA", "IN1LP Switch", "IN1LP" }, + { "IN1L PGA", "IN1LN Switch", "IN1LN" }, + + { "IN1R PGA", "IN1RP Switch", "IN1RP" }, + { "IN1R PGA", "IN1RN Switch", "IN1RN" }, + + { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LN Switch", "IN2LN" }, + + { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RN Switch", "IN2RN" }, + + { "Direct Voice", NULL, "IN2LP/VXRN" }, + { "Direct Voice", NULL, "IN2RP/VXRP" }, + + { "MIXINL", "IN1L Switch", "IN1L PGA" }, + { "MIXINL", "IN2L Switch", "IN2L PGA" }, + { "MIXINL", NULL, "Direct Voice" }, + { "MIXINL", NULL, "IN1LP" }, + { "MIXINL", NULL, "Left Output Mixer" }, + + { "MIXINR", "IN1R Switch", "IN1R PGA" }, + { "MIXINR", "IN2R Switch", "IN2R PGA" }, + { "MIXINR", NULL, "Direct Voice" }, + { "MIXINR", NULL, "IN1RP" }, + { "MIXINR", NULL, "Right Output Mixer" }, + + { "ADCL", NULL, "MIXINL" }, + { "ADCR", NULL, "MIXINR" }, + + { "Left Output Mixer", "Left Input Switch", "MIXINL" }, + { "Left Output Mixer", "Right Input Switch", "MIXINR" }, + { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, + { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, + { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, + + { "Right Output Mixer", "Left Input Switch", "MIXINL" }, + { "Right Output Mixer", "Right Input Switch", "MIXINR" }, + { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, + { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, + { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, + + { "Left Output PGA", NULL, "Left Output Mixer" }, + { "Left Output PGA", NULL, "TOCLK" }, + + { "Right Output PGA", NULL, "Right Output Mixer" }, + { "Right Output PGA", NULL, "TOCLK" }, + + { "Earpiece Mixer", "Direct Voice Switch", "Direct Voice" }, + { "Earpiece Mixer", "Left Output Switch", "Left Output PGA" }, + { "Earpiece Mixer", "Right Output Switch", "Right Output PGA" }, + + { "Earpiece Driver", NULL, "Earpiece Mixer" }, + { "HPOUT2N", NULL, "Earpiece Driver" }, + { "HPOUT2P", NULL, "Earpiece Driver" }, + + { "SPKL", "Input Switch", "MIXINL" }, + { "SPKL", "IN1LP Switch", "IN1LP" }, + { "SPKL", "Output Switch", "Left Output Mixer" }, + { "SPKL", NULL, "TOCLK" }, + + { "SPKR", "Input Switch", "MIXINR" }, + { "SPKR", "IN1RP Switch", "IN1RP" }, + { "SPKR", "Output Switch", "Right Output Mixer" }, + { "SPKR", NULL, "TOCLK" }, + + { "SPKL Boost", "Direct Voice Switch", "Direct Voice" }, + { "SPKL Boost", "SPKL Switch", "SPKL" }, + { "SPKL Boost", "SPKR Switch", "SPKR" }, + + { "SPKR Boost", "Direct Voice Switch", "Direct Voice" }, + { "SPKR Boost", "SPKR Switch", "SPKR" }, + { "SPKR Boost", "SPKL Switch", "SPKL" }, + + { "SPKL Driver", NULL, "SPKL Boost" }, + { "SPKL Driver", NULL, "CLK_SYS" }, + + { "SPKR Driver", NULL, "SPKR Boost" }, + { "SPKR Driver", NULL, "CLK_SYS" }, + + { "SPKOUTLP", NULL, "SPKL Driver" }, + { "SPKOUTLN", NULL, "SPKL Driver" }, + { "SPKOUTRP", NULL, "SPKR Driver" }, + { "SPKOUTRN", NULL, "SPKR Driver" }, + + { "Left Headphone Mux", "Mixer", "Left Output Mixer" }, + { "Right Headphone Mux", "Mixer", "Right Output Mixer" }, + + { "Headphone PGA", NULL, "Left Headphone Mux" }, + { "Headphone PGA", NULL, "Right Headphone Mux" }, + { "Headphone PGA", NULL, "CLK_SYS" }, + + { "HPOUT1L", NULL, "Headphone PGA" }, + { "HPOUT1R", NULL, "Headphone PGA" }, + + { "LINEOUT1N", NULL, "LINEOUT1N Driver" }, + { "LINEOUT1P", NULL, "LINEOUT1P Driver" }, + { "LINEOUT2N", NULL, "LINEOUT2N Driver" }, + { "LINEOUT2P", NULL, "LINEOUT2P Driver" }, +}; + +static const struct snd_soc_dapm_route lineout1_diff_routes[] = { + { "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" }, + { "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" }, + + { "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" }, + { "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" }, +}; + +static const struct snd_soc_dapm_route lineout1_se_routes[] = { + { "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" }, + { "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" }, + + { "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" }, + + { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, + { "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" }, +}; + +static const struct snd_soc_dapm_route lineout2_diff_routes[] = { + { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, + { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" }, + + { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, + { "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" }, +}; + +static const struct snd_soc_dapm_route lineout2_se_routes[] = { + { "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" }, + { "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" }, + + { "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" }, + + { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, + { "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" }, +}; + +int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) +{ + /* Latch volume update bits & default ZC on */ + snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME, + WM8993_IN1_VU, WM8993_IN1_VU); + snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8993_IN1_VU, WM8993_IN1_VU); + snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_3_4_VOLUME, + WM8993_IN2_VU, WM8993_IN2_VU); + snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8993_IN2_VU, WM8993_IN2_VU); + + snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT, + WM8993_SPKOUT_VU, WM8993_SPKOUT_VU); + + snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME, + WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC); + snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME, + WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC, + WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC); + + snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME, + WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC); + snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME, + WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, + WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); + + snd_soc_add_controls(codec, analogue_snd_controls, + ARRAY_SIZE(analogue_snd_controls)); + + snd_soc_dapm_new_controls(codec, analogue_dapm_widgets, + ARRAY_SIZE(analogue_dapm_widgets)); + return 0; +} +EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls); + +int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, + int lineout1_diff, int lineout2_diff) +{ + snd_soc_dapm_add_routes(codec, analogue_routes, + ARRAY_SIZE(analogue_routes)); + + if (lineout1_diff) + snd_soc_dapm_add_routes(codec, + lineout1_diff_routes, + ARRAY_SIZE(lineout1_diff_routes)); + else + snd_soc_dapm_add_routes(codec, + lineout1_se_routes, + ARRAY_SIZE(lineout1_se_routes)); + + if (lineout2_diff) + snd_soc_dapm_add_routes(codec, + lineout2_diff_routes, + ARRAY_SIZE(lineout2_diff_routes)); + else + snd_soc_dapm_add_routes(codec, + lineout2_se_routes, + ARRAY_SIZE(lineout2_se_routes)); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); + +MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h new file mode 100644 index 000000000000..ec09cb6a2939 --- /dev/null +++ b/sound/soc/codecs/wm_hubs.h @@ -0,0 +1,24 @@ +/* + * wm_hubs.h -- WM899x common code + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM_HUBS_H +#define _WM_HUBS_H + +struct snd_soc_codec; + +extern const unsigned int wm_hubs_spkmix_tlv[]; + +extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); +extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int); + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 411a710be660..4dfd4ad9d90e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -9,6 +9,9 @@ config SND_DAVINCI_SOC config SND_DAVINCI_SOC_I2S tristate +config SND_DAVINCI_SOC_MCASP + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" depends on SND_DAVINCI_SOC @@ -19,6 +22,16 @@ config SND_DAVINCI_SOC_EVM Say Y if you want to add support for SoC audio on TI DaVinci DM6446 or DM355 EVM platforms. +config SND_DM6467_SOC_EVM + tristate "SoC Audio support for DaVinci DM6467 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + select SND_SOC_SPDIF + + help + Say Y if you want to add support for SoC audio on TI + config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" depends on SND_DAVINCI_SOC && MACH_SFFSDR @@ -28,3 +41,23 @@ config SND_DAVINCI_SOC_SFFSDR help Say Y if you want to add support for SoC audio on Lyrtech SFFSDR board. + +config SND_DA830_SOC_EVM + tristate "SoC Audio support for DA830/OMAP-L137 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + + help + Say Y if you want to add support for SoC audio on TI + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM + diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index ca8bae1fc3f6..a6939d71b988 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,13 +1,18 @@ # DAVINCI Platform Support snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o +snd-soc-davinci-mcasp-objs:= davinci-mcasp.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o +obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 58fd1cbedd88..67414f659405 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -14,6 +14,7 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -27,9 +28,10 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" - +#include "davinci-mcasp.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -43,7 +45,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -53,6 +55,10 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) + sysclk = 24576000; + else return -EINVAL; @@ -144,6 +150,32 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm6467_evm_dai[] = { + { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, + }, + { + .name = "McASP", + .stream_name = "spdif", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI], + .codec_dai = &dit_stub_dai, + .ops = &evm_ops, + }, +}; +static struct snd_soc_dai_link da8xx_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", @@ -152,73 +184,80 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; -/* evm audio private data */ -static struct aic3x_setup_data evm_aic3x_setup = { - .i2c_bus = 1, - .i2c_address = 0x1b, +/* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card dm6467_snd_soc_card_evm = { + .name = "DaVinci DM6467 EVM", + .platform = &davinci_soc_platform, + .dai_link = dm6467_evm_dai, + .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +static struct snd_soc_card da830_snd_soc_card = { + .name = "DA830/OMAP-L137 EVM", + .dai_link = &da8xx_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + +static struct snd_soc_card da850_snd_soc_card = { + .name = "DA850/OMAP-L138 EVM", + .dai_link = &da8xx_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + +static struct aic3x_setup_data aic3x_setup; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &evm_aic3x_setup, -}; - -/* DM6446 EVM uses ASP0; line-out is a pair of RCA jacks */ -static struct resource evm_snd_resources[] = { - { - .start = DAVINCI_ASP0_BASE, - .end = DAVINCI_ASP0_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, + .codec_data = &aic3x_setup, }; -static struct evm_snd_platform_data evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP0_TX, - .rx_dma_ch = DAVINCI_DMA_ASP0_RX, +/* evm audio subsystem */ +static struct snd_soc_device dm6467_evm_snd_devdata = { + .card = &dm6467_snd_soc_card_evm, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, }; -/* DM335 EVM uses ASP1; line-out is a stereo mini-jack */ -static struct resource dm335evm_snd_resources[] = { - { - .start = DAVINCI_ASP1_BASE, - .end = DAVINCI_ASP1_BASE + SZ_8K - 1, - .flags = IORESOURCE_MEM, - }, +/* evm audio subsystem */ +static struct snd_soc_device da830_evm_snd_devdata = { + .card = &da830_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, }; -static struct evm_snd_platform_data dm335evm_snd_data = { - .tx_dma_ch = DAVINCI_DMA_ASP1_TX, - .rx_dma_ch = DAVINCI_DMA_ASP1_RX, +static struct snd_soc_device da850_evm_snd_devdata = { + .card = &da850_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, }; static struct platform_device *evm_snd_device; static int __init evm_init(void) { - struct resource *resources; - unsigned num_resources; - struct evm_snd_platform_data *data; + struct snd_soc_device *evm_snd_dev_data; int index; int ret; if (machine_is_davinci_evm()) { - davinci_cfg_reg(DM644X_MCBSP); - - resources = evm_snd_resources; - num_resources = ARRAY_SIZE(evm_snd_resources); - data = &evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { - /* we don't use ASP1 IRQs, or we'd need to mux them ... */ - davinci_cfg_reg(DM355_EVT8_ASP1_TX); - davinci_cfg_reg(DM355_EVT9_ASP1_RX); - - resources = dm335evm_snd_resources; - num_resources = ARRAY_SIZE(dm335evm_snd_resources); - data = &dm335evm_snd_data; + evm_snd_dev_data = &evm_snd_devdata; + index = 1; + } else if (machine_is_davinci_dm6467_evm()) { + evm_snd_dev_data = &dm6467_evm_snd_devdata; + index = 0; + } else if (machine_is_davinci_da830_evm()) { + evm_snd_dev_data = &da830_evm_snd_devdata; index = 1; + } else if (machine_is_davinci_da850_evm()) { + evm_snd_dev_data = &da850_evm_snd_devdata; + index = 0; } else return -EINVAL; @@ -226,17 +265,8 @@ static int __init evm_init(void) if (!evm_snd_device) return -ENOMEM; - platform_set_drvdata(evm_snd_device, &evm_snd_devdata); - evm_snd_devdata.dev = &evm_snd_device->dev; - platform_device_add_data(evm_snd_device, data, sizeof(*data)); - - ret = platform_device_add_resources(evm_snd_device, resources, - num_resources); - if (ret) { - platform_device_put(evm_snd_device); - return ret; - } - + platform_set_drvdata(evm_snd_device, evm_snd_dev_data); + evm_snd_dev_data->dev = &evm_snd_device->dev; ret = platform_device_add(evm_snd_device); if (ret) platform_device_put(evm_snd_device); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index b1ea52fc83c7..12a6c549ee6e 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -22,6 +22,8 @@ #include <sound/initval.h> #include <sound/soc.h> +#include <mach/asp.h> + #include "davinci-pcm.h" @@ -63,6 +65,7 @@ #define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) @@ -85,14 +88,6 @@ #define DAVINCI_MCBSP_PCR_FSRM (1 << 10) #define DAVINCI_MCBSP_PCR_FSXM (1 << 11) -#define MOD_REG_BIT(val, mask, set) do { \ - if (set) { \ - val |= mask; \ - } else { \ - val &= ~mask; \ - } \ -} while (0) - enum { DAVINCI_MCBSP_WORD_8 = 0, DAVINCI_MCBSP_WORD_12, @@ -112,6 +107,10 @@ static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { struct davinci_mcbsp_dev { void __iomem *base; +#define MOD_DSP_A 0 +#define MOD_DSP_B 1 + int mode; + u32 pcr; struct clk *clk; struct davinci_pcm_dma_params *dma_params[2]; }; @@ -127,96 +126,100 @@ static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg) return __raw_readl(dev->base + reg); } -static void davinci_mcbsp_start(struct snd_pcm_substream *substream) +static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback) +{ + u32 m = playback ? DAVINCI_MCBSP_PCR_CLKXP : DAVINCI_MCBSP_PCR_CLKRP; + /* The clock needs to toggle to complete reset. + * So, fake it by toggling the clk polarity. + */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr ^ m); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr); +} + +static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev, + struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_platform *platform = socdev->card->platform; - u32 w; - int ret; - - /* Start the sample generator and enable transmitter/receiver */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + u32 spcr; + u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST; + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + if (spcr & mask) { + /* start off disabled */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + spcr & ~mask); + toggle_clock(dev, playback); + } + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) { + /* Start the sample generator */ + spcr |= DAVINCI_MCBSP_SPCR_GRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (playback) { /* Stop the DMA to avoid data loss */ /* while the transmitter is out of reset to handle XSYNCERR */ if (platform->pcm_ops->trigger) { - ret = platform->pcm_ops->trigger(substream, + int ret = platform->pcm_ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP); if (ret < 0) printk(KERN_DEBUG "Playback DMA stop failed\n"); } /* Enable the transmitter */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); /* wait for any unexpected frame sync error to occur */ udelay(100); /* Disable the transmitter to clear any outstanding XSYNCERR */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~DAVINCI_MCBSP_SPCR_XRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); /* Restart the DMA */ if (platform->pcm_ops->trigger) { - ret = platform->pcm_ops->trigger(substream, + int ret = platform->pcm_ops->trigger(substream, SNDRV_PCM_TRIGGER_START); if (ret < 0) printk(KERN_DEBUG "Playback DMA start failed\n"); } - /* Enable the transmitter */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); - - } else { - - /* Enable the reciever */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); } + /* Enable transmitter or receiver */ + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr |= mask; - /* Start frame sync */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM)) { + /* Start frame sync */ + spcr |= DAVINCI_MCBSP_SPCR_FRST; + } + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } -static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) +static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; - u32 w; + u32 spcr; /* Reset transmitter/receiver and sample rate/frame sync generators */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST | - DAVINCI_MCBSP_SPCR_FRST, 0); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); - else - MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr &= ~(DAVINCI_MCBSP_SPCR_GRST | DAVINCI_MCBSP_SPCR_FRST); + spcr &= playback ? ~DAVINCI_MCBSP_SPCR_XRST : ~DAVINCI_MCBSP_SPCR_RRST; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); + toggle_clock(dev, playback); } static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) + struct snd_soc_dai *cpu_dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; - + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; - return 0; } @@ -228,12 +231,11 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; - unsigned int rcr; - unsigned int xcr; srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* cpu is master */ @@ -258,11 +260,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - rcr = DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr = DAVINCI_MCBSP_XCR_XFIG | DAVINCI_MCBSP_XCR_XFRLEN1(1); + /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_B: - break; case SND_SOC_DAIFMT_I2S: /* Davinci doesn't support TRUE I2S, but some codecs will have * the left and right channels contiguous. This allows @@ -282,8 +281,10 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, */ fmt ^= SND_SOC_DAIFMT_NB_IF; case SND_SOC_DAIFMT_DSP_A: - rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1); - xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); + dev->mode = MOD_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + dev->mode = MOD_DSP_B; break; default: printk(KERN_ERR "%s:bad format\n", __func__); @@ -343,9 +344,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); + dev->pcr = pcr; davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); return 0; } @@ -353,31 +353,40 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_interval *i = NULL; int mcbsp_word_length; - u32 w; + unsigned int rcr, xcr, srgr; + u32 spcr; /* general line settings */ - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - w |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + spcr |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } else { - w |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + spcr |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE; + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - w = DAVINCI_MCBSP_SRGR_FSGM; - MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); + rcr = DAVINCI_MCBSP_RCR_RFIG; + xcr = DAVINCI_MCBSP_XCR_XFIG; + if (dev->mode == MOD_DSP_B) { + rcr |= DAVINCI_MCBSP_RCR_RDATDLY(0); + xcr |= DAVINCI_MCBSP_XCR_XDATDLY(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1); + xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); + } /* Determine xfer data type */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -397,18 +406,31 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + dma_params->acnt = dma_params->data_type; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); - } else { - w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); - MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); + xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); + else + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + return 0; +} + +static int davinci_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = dai->private_data; + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + davinci_mcbsp_stop(dev, playback); + if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) { + /* codec is master */ + davinci_mcbsp_start(dev, substream); } return 0; } @@ -416,35 +438,72 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { + struct davinci_mcbsp_dev *dev = dai->private_data; int ret = 0; + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) + return 0; /* return if codec is master */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - davinci_mcbsp_start(substream); + davinci_mcbsp_start(dev, substream); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - davinci_mcbsp_stop(substream); + davinci_mcbsp_stop(dev, playback); break; default: ret = -EINVAL; } - return ret; } -static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) +static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = dai->private_data; + int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + davinci_mcbsp_stop(dev, playback); +} + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +static struct snd_soc_dai_ops davinci_i2s_dai_ops = { + .startup = davinci_i2s_startup, + .shutdown = davinci_i2s_shutdown, + .prepare = davinci_i2s_prepare, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params, + .set_fmt = davinci_i2s_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_i2s_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; - struct resource *mem, *ioarea; - struct evm_snd_platform_data *pdata; + struct resource *mem, *ioarea, *res; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -466,8 +525,6 @@ static int davinci_i2s_probe(struct platform_device *pdev, goto err_release_region; } - cpu_dai->private_data = dev; - dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; @@ -476,18 +533,37 @@ static int davinci_i2s_probe(struct platform_device *pdev, clk_enable(dev->clk); dev->base = (void __iomem *)IO_ADDRESS(mem->start); - pdata = pdev->dev.platform_data; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + ret = -ENXIO; + goto err_free_mem; + } + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = res->start; + + davinci_i2s_dai.private_data = dev; + ret = snd_soc_register_dai(&davinci_i2s_dai); + if (ret != 0) + goto err_free_mem; + return 0; err_free_mem: @@ -498,62 +574,40 @@ err_release_region: return ret; } -static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int davinci_i2s_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link->cpu_dai; - struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = davinci_i2s_dai.private_data; struct resource *mem; + snd_soc_unregister_dai(&davinci_i2s_dai); clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; - kfree(dev); - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); release_mem_region(mem->start, (mem->end - mem->start) + 1); -} -#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 - -static struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, -}; + return 0; +} -struct snd_soc_dai davinci_i2s_dai = { - .name = "davinci-i2s", - .id = 0, - .probe = davinci_i2s_probe, - .remove = davinci_i2s_remove, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = DAVINCI_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &davinci_i2s_dai_ops, +static struct platform_driver davinci_mcbsp_driver = { + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .driver = { + .name = "davinci-asp", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(davinci_i2s_dai); static int __init davinci_i2s_init(void) { - return snd_soc_register_dai(&davinci_i2s_dai); + return platform_driver_register(&davinci_mcbsp_driver); } module_init(davinci_i2s_init); static void __exit davinci_i2s_exit(void) { - snd_soc_unregister_dai(&davinci_i2s_dai); + platform_driver_unregister(&davinci_mcbsp_driver); } module_exit(davinci_i2s_exit); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c new file mode 100644 index 000000000000..eca22d7829d2 --- /dev/null +++ b/sound/soc/davinci/davinci-mcasp.c @@ -0,0 +1,973 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * Multi-channel Audio Serial Port Driver + * + * Author: Nirmal Pandey <n-pandey@ti.com>, + * Suresh Rajashekara <suresh.r@ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-mcasp.h" + +/* + * McASP register definitions + */ +#define DAVINCI_MCASP_PID_REG 0x00 +#define DAVINCI_MCASP_PWREMUMGT_REG 0x04 + +#define DAVINCI_MCASP_PFUNC_REG 0x10 +#define DAVINCI_MCASP_PDIR_REG 0x14 +#define DAVINCI_MCASP_PDOUT_REG 0x18 +#define DAVINCI_MCASP_PDSET_REG 0x1c + +#define DAVINCI_MCASP_PDCLR_REG 0x20 + +#define DAVINCI_MCASP_TLGC_REG 0x30 +#define DAVINCI_MCASP_TLMR_REG 0x34 + +#define DAVINCI_MCASP_GBLCTL_REG 0x44 +#define DAVINCI_MCASP_AMUTE_REG 0x48 +#define DAVINCI_MCASP_LBCTL_REG 0x4c + +#define DAVINCI_MCASP_TXDITCTL_REG 0x50 + +#define DAVINCI_MCASP_GBLCTLR_REG 0x60 +#define DAVINCI_MCASP_RXMASK_REG 0x64 +#define DAVINCI_MCASP_RXFMT_REG 0x68 +#define DAVINCI_MCASP_RXFMCTL_REG 0x6c + +#define DAVINCI_MCASP_ACLKRCTL_REG 0x70 +#define DAVINCI_MCASP_AHCLKRCTL_REG 0x74 +#define DAVINCI_MCASP_RXTDM_REG 0x78 +#define DAVINCI_MCASP_EVTCTLR_REG 0x7c + +#define DAVINCI_MCASP_RXSTAT_REG 0x80 +#define DAVINCI_MCASP_RXTDMSLOT_REG 0x84 +#define DAVINCI_MCASP_RXCLKCHK_REG 0x88 +#define DAVINCI_MCASP_REVTCTL_REG 0x8c + +#define DAVINCI_MCASP_GBLCTLX_REG 0xa0 +#define DAVINCI_MCASP_TXMASK_REG 0xa4 +#define DAVINCI_MCASP_TXFMT_REG 0xa8 +#define DAVINCI_MCASP_TXFMCTL_REG 0xac + +#define DAVINCI_MCASP_ACLKXCTL_REG 0xb0 +#define DAVINCI_MCASP_AHCLKXCTL_REG 0xb4 +#define DAVINCI_MCASP_TXTDM_REG 0xb8 +#define DAVINCI_MCASP_EVTCTLX_REG 0xbc + +#define DAVINCI_MCASP_TXSTAT_REG 0xc0 +#define DAVINCI_MCASP_TXTDMSLOT_REG 0xc4 +#define DAVINCI_MCASP_TXCLKCHK_REG 0xc8 +#define DAVINCI_MCASP_XEVTCTL_REG 0xcc + +/* Left(even TDM Slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRA_REG 0x100 +/* Right(odd TDM slot) Channel Status Register File */ +#define DAVINCI_MCASP_DITCSRB_REG 0x118 +/* Left(even TDM slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRA_REG 0x130 +/* Right(odd TDM Slot) User Data Register File */ +#define DAVINCI_MCASP_DITUDRB_REG 0x148 + +/* Serializer n Control Register */ +#define DAVINCI_MCASP_XRSRCTL_BASE_REG 0x180 +#define DAVINCI_MCASP_XRSRCTL_REG(n) (DAVINCI_MCASP_XRSRCTL_BASE_REG + \ + (n << 2)) + +/* Transmit Buffer for Serializer n */ +#define DAVINCI_MCASP_TXBUF_REG 0x200 +/* Receive Buffer for Serializer n */ +#define DAVINCI_MCASP_RXBUF_REG 0x280 + +/* McASP FIFO Registers */ +#define DAVINCI_MCASP_WFIFOCTL (0x1010) +#define DAVINCI_MCASP_WFIFOSTS (0x1014) +#define DAVINCI_MCASP_RFIFOCTL (0x1018) +#define DAVINCI_MCASP_RFIFOSTS (0x101C) + +/* + * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management + * Register Bits + */ +#define MCASP_FREE BIT(0) +#define MCASP_SOFT BIT(1) + +/* + * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits + */ +#define AXR(n) (1<<n) +#define PFUNC_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits + */ +#define AXR(n) (1<<n) +#define PDIR_AMUTE BIT(25) +#define ACLKX BIT(26) +#define AHCLKX BIT(27) +#define AFSX BIT(28) +#define ACLKR BIT(29) +#define AHCLKR BIT(30) +#define AFSR BIT(31) + +/* + * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits + */ +#define DITEN BIT(0) /* Transmit DIT mode enable/disable */ +#define VA BIT(2) +#define VB BIT(3) + +/* + * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits + */ +#define TXROT(val) (val) +#define TXSEL BIT(3) +#define TXSSZ(val) (val<<4) +#define TXPBIT(val) (val<<8) +#define TXPAD(val) (val<<13) +#define TXORD BIT(15) +#define FSXDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits + */ +#define RXROT(val) (val) +#define RXSEL BIT(3) +#define RXSSZ(val) (val<<4) +#define RXPBIT(val) (val<<8) +#define RXPAD(val) (val<<13) +#define RXORD BIT(15) +#define FSRDLY(val) (val<<16) + +/* + * DAVINCI_MCASP_TXFMCTL_REG - Transmit Frame Control Register Bits + */ +#define FSXPOL BIT(0) +#define AFSXE BIT(1) +#define FSXDUR BIT(4) +#define FSXMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits + */ +#define FSRPOL BIT(0) +#define AFSRE BIT(1) +#define FSRDUR BIT(4) +#define FSRMOD(val) (val<<7) + +/* + * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits + */ +#define ACLKXDIV(val) (val) +#define ACLKXE BIT(5) +#define TX_ASYNC BIT(6) +#define ACLKXPOL BIT(7) + +/* + * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits + */ +#define ACLKRDIV(val) (val) +#define ACLKRE BIT(5) +#define RX_ASYNC BIT(6) +#define ACLKRPOL BIT(7) + +/* + * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control + * Register Bits + */ +#define AHCLKXDIV(val) (val) +#define AHCLKXPOL BIT(14) +#define AHCLKXE BIT(15) + +/* + * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control + * Register Bits + */ +#define AHCLKRDIV(val) (val) +#define AHCLKRPOL BIT(14) +#define AHCLKRE BIT(15) + +/* + * DAVINCI_MCASP_XRSRCTL_BASE_REG - Serializer Control Register Bits + */ +#define MODE(val) (val) +#define DISMOD (val)(val<<2) +#define TXSTATE BIT(4) +#define RXSTATE BIT(5) + +/* + * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits + */ +#define LBEN BIT(0) +#define LBORD BIT(1) +#define LBGENMODE(val) (val<<2) + +/* + * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration + */ +#define TXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration + */ +#define RXTDMS(n) (1<<n) + +/* + * DAVINCI_MCASP_GBLCTL_REG - Global Control Register Bits + */ +#define RXCLKRST BIT(0) /* Receiver Clock Divider Reset */ +#define RXHCLKRST BIT(1) /* Receiver High Frequency Clock Divider */ +#define RXSERCLR BIT(2) /* Receiver Serializer Clear */ +#define RXSMRST BIT(3) /* Receiver State Machine Reset */ +#define RXFSRST BIT(4) /* Frame Sync Generator Reset */ +#define TXCLKRST BIT(8) /* Transmitter Clock Divider Reset */ +#define TXHCLKRST BIT(9) /* Transmitter High Frequency Clock Divider*/ +#define TXSERCLR BIT(10) /* Transmit Serializer Clear */ +#define TXSMRST BIT(11) /* Transmitter State Machine Reset */ +#define TXFSRST BIT(12) /* Frame Sync Generator Reset */ + +/* + * DAVINCI_MCASP_AMUTE_REG - Mute Control Register Bits + */ +#define MUTENA(val) (val) +#define MUTEINPOL BIT(2) +#define MUTEINENA BIT(3) +#define MUTEIN BIT(4) +#define MUTER BIT(5) +#define MUTEX BIT(6) +#define MUTEFSR BIT(7) +#define MUTEFSX BIT(8) +#define MUTEBADCLKR BIT(9) +#define MUTEBADCLKX BIT(10) +#define MUTERXDMAERR BIT(11) +#define MUTETXDMAERR BIT(12) + +/* + * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits + */ +#define RXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits + */ +#define TXDATADMADIS BIT(0) + +/* + * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits + */ +#define FIFO_ENABLE BIT(16) +#define NUMEVT_MASK (0xFF << 8) +#define NUMDMA_MASK (0xFF) + +#define DAVINCI_MCASP_NUM_SERIALIZER 16 + +static inline void mcasp_set_bits(void __iomem *reg, u32 val) +{ + __raw_writel(__raw_readl(reg) | val, reg); +} + +static inline void mcasp_clr_bits(void __iomem *reg, u32 val) +{ + __raw_writel((__raw_readl(reg) & ~(val)), reg); +} + +static inline void mcasp_mod_bits(void __iomem *reg, u32 val, u32 mask) +{ + __raw_writel((__raw_readl(reg) & ~mask) | val, reg); +} + +static inline void mcasp_set_reg(void __iomem *reg, u32 val) +{ + __raw_writel(val, reg); +} + +static inline u32 mcasp_get_reg(void __iomem *reg) +{ + return (unsigned int)__raw_readl(reg); +} + +static inline void mcasp_set_ctl_reg(void __iomem *regs, u32 val) +{ + int i = 0; + + mcasp_set_bits(regs, val); + + /* programming GBLCTL needs to read back from GBLCTL and verfiy */ + /* loop count is to avoid the lock-up */ + for (i = 0; i < 1000; i++) { + if ((mcasp_get_reg(regs) & val) == val) + break; + } + + if (i == 1000 && ((mcasp_get_reg(regs) & val) != val)) + printk(KERN_ERR "GBLCTL write error\n"); +} + +static int davinci_mcasp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + cpu_dai->dma_data = dev->dma_params[substream->stream]; + return 0; +} + +static void mcasp_start_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, RXFSRST); +} + +static void mcasp_start_tx(struct davinci_audio_dev *dev) +{ + u8 offset = 0, i; + u32 cnt; + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) { + offset = i; + break; + } + } + + /* wait for TX ready */ + cnt = 0; + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + TXSTATE) && (cnt < 100000)) + cnt++; + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); +} + +static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_start_tx(dev); + else + mcasp_start_rx(dev); + + /* enable FIFO */ + if (dev->txnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); +} + +static void mcasp_stop_rx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLR_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); +} + +static void mcasp_stop_tx(struct davinci_audio_dev *dev) +{ + mcasp_set_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, 0); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); +} + +static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + mcasp_stop_tx(dev); + else + mcasp_stop_rx(dev); + + /* disable FIFO */ + if (dev->txnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); +} + +static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + void __iomem *base = dev->base; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* codec is clock and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* codec is clock and frame master */ + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_clr_bits(base + DAVINCI_MCASP_PDIR_REG, (0x3f << 26)); + break; + + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_IF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_IB_IF: + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + case SND_SOC_DAIFMT_NB_NF: + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); + + mcasp_clr_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL); + mcasp_clr_bits(base + DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_config_channel_size(struct davinci_audio_dev *dev, + int channel_size) +{ + u32 fmt = 0; + + switch (channel_size) { + case DAVINCI_AUDIO_WORD_8: + fmt = 0x03; + break; + + case DAVINCI_AUDIO_WORD_12: + fmt = 0x05; + break; + + case DAVINCI_AUDIO_WORD_16: + fmt = 0x07; + break; + + case DAVINCI_AUDIO_WORD_20: + fmt = 0x09; + break; + + case DAVINCI_AUDIO_WORD_24: + fmt = 0x0B; + break; + + case DAVINCI_AUDIO_WORD_28: + fmt = 0x0D; + break; + + case DAVINCI_AUDIO_WORD_32: + fmt = 0x0F; + break; + + default: + return -EINVAL; + } + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, + RXSSZ(fmt), RXSSZ(0x0F)); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXSSZ(fmt), TXSSZ(0x0F)); + return 0; +} + +static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +{ + int i; + u8 tx_ser = 0; + u8 rx_ser = 0; + + /* Default configuration */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); + + /* All PINS as McASP */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_PFUNC_REG, 0x00000000); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, + TXDATADMADIS); + } else { + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_REVTCTL_REG, + RXDATADMADIS); + } + + for (i = 0; i < dev->num_serializer; i++) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + dev->serial_dir[i]); + if (dev->serial_dir[i] == TX_MODE) { + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + tx_ser++; + } else if (dev->serial_dir[i] == RX_MODE) { + mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, + AXR(i)); + rx_ser++; + } + } + + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt * tx_ser > 64) + dev->txnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + + if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (dev->rxnumevt * rx_ser > 64) + dev->rxnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); + } +} + +static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) +{ + int i, active_slots; + u32 mask = 0; + + active_slots = (dev->tdm_slots > 31) ? 32 : dev->tdm_slots; + for (i = 0; i < active_slots; i++) + mask |= (1 << i); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, + AHCLKXE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXORD); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(dev->tdm_slots), FSXMOD(0x1FF)); + else + printk(KERN_ERR "playback tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_TXFMCTL_REG, FSXDUR); + } else { + /* bit stream is MSB first with no delay */ + /* DSP_B mode */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, RXORD); + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKRCTL_REG, + AHCLKRE); + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXTDM_REG, mask); + + if ((dev->tdm_slots >= 2) || (dev->tdm_slots <= 32)) + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(dev->tdm_slots), FSRMOD(0x1FF)); + else + printk(KERN_ERR "capture tdm slot %d not supported\n", + dev->tdm_slots); + + mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, 0xFFFFFFFF); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RXFMCTL_REG, FSRDUR); + } +} + +/* S/PDIF */ +static void davinci_hw_dit_param(struct davinci_audio_dev *dev) +{ + /* Set the PDIR for Serialiser as output */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); + + /* TXMASK for 24 bits */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); + + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 + and LSB first */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, + TXROT(6) | TXSSZ(15)); + + /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXFMCTL_REG, + AFSXE | FSXMOD(0x180)); + + /* Set the TX tdm : for all the slots */ + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); + + /* Set the TX clock controls : div = 1 and internal */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, + ACLKXE | TX_ASYNC); + + mcasp_clr_bits(dev->base + DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); + + /* Only 44100 and 48000 are valid, both have the same setting */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); + + /* Enable the DIT */ + mcasp_set_bits(dev->base + DAVINCI_MCASP_TXDITCTL_REG, DITEN); +} + +static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct davinci_audio_dev *dev = cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = + dev->dma_params[substream->stream]; + int word_length; + u8 numevt; + + davinci_hw_common_param(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + numevt = dev->txnumevt; + else + numevt = dev->rxnumevt; + + if (!numevt) + numevt = 1; + + if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) + davinci_hw_dit_param(dev); + else + davinci_hw_param(dev, substream->stream); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + word_length = DAVINCI_AUDIO_WORD_8; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + word_length = DAVINCI_AUDIO_WORD_16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + word_length = DAVINCI_AUDIO_WORD_32; + break; + + default: + printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); + return -EINVAL; + } + + if (dev->version == MCASP_VERSION_2) { + dma_params->data_type *= numevt; + dma_params->acnt = 4 * numevt; + } else + dma_params->acnt = dma_params->data_type; + + davinci_config_channel_size(dev, word_length); + + return 0; +} + +static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_audio_dev *dev = rtd->dai->cpu_dai->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcasp_start(dev, substream->stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcasp_stop(dev, substream->stream); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static struct snd_soc_dai_ops davinci_mcasp_dai_ops = { + .startup = davinci_mcasp_startup, + .trigger = davinci_mcasp_trigger, + .hw_params = davinci_mcasp_hw_params, + .set_fmt = davinci_mcasp_set_dai_fmt, + +}; + +struct snd_soc_dai davinci_mcasp_dai[] = { + { + .name = "davinci-i2s", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &davinci_mcasp_dai_ops, + + }, + { + .name = "davinci-dit", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 384, + .rates = DAVINCI_MCASP_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &davinci_mcasp_dai_ops, + }, + +}; +EXPORT_SYMBOL_GPL(davinci_mcasp_dai); + +static int davinci_mcasp_probe(struct platform_device *pdev) +{ + struct davinci_pcm_dma_params *dma_data; + struct resource *mem, *ioarea, *res; + struct snd_platform_data *pdata; + struct davinci_audio_dev *dev; + int count = 0; + int ret = 0; + + dev = kzalloc(sizeof(struct davinci_audio_dev), GFP_KERNEL); + if (!dev) + return -ENOMEM; + + dma_data = kzalloc(sizeof(struct davinci_pcm_dma_params) * 2, + GFP_KERNEL); + if (!dma_data) { + ret = -ENOMEM; + goto err_release_dev; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_data; + } + + ioarea = request_mem_region(mem->start, + (mem->end - mem->start) + 1, pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "Audio region already claimed\n"); + ret = -EBUSY; + goto err_release_data; + } + + pdata = pdev->dev.platform_data; + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_release_region; + } + + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + dev->op_mode = pdata->op_mode; + dev->tdm_slots = pdata->tdm_slots; + dev->num_serializer = pdata->num_serializer; + dev->serial_dir = pdata->serial_dir; + dev->codec_fmt = pdata->codec_fmt; + dev->version = pdata->version; + dev->txnumevt = pdata->txnumevt; + dev->rxnumevt = pdata->rxnumevt; + + dma_data[count].name = "I2S PCM Stereo out"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t) (pdata->tx_dma_offset + + io_v2p(dev->base)); + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &dma_data[count]; + + /* first TX, then RX */ + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + count++; + dma_data[count].name = "I2S PCM Stereo in"; + dma_data[count].eventq_no = pdata->eventq_no; + dma_data[count].dma_addr = (dma_addr_t)(pdata->rx_dma_offset + + io_v2p(dev->base)); + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &dma_data[count]; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + goto err_release_region; + } + + dma_data[count].channel = res->start; + davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); + + if (ret != 0) + goto err_release_region; + return 0; + +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); +err_release_data: + kfree(dma_data); +err_release_dev: + kfree(dev); + + return ret; +} + +static int davinci_mcasp_remove(struct platform_device *pdev) +{ + struct snd_platform_data *pdata = pdev->dev.platform_data; + struct davinci_pcm_dma_params *dma_data; + struct davinci_audio_dev *dev; + struct resource *mem; + + snd_soc_unregister_dai(&davinci_mcasp_dai[pdata->op_mode]); + dev = davinci_mcasp_dai[pdata->op_mode].private_data; + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + dma_data = dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + kfree(dma_data); + kfree(dev); + + return 0; +} + +static struct platform_driver davinci_mcasp_driver = { + .probe = davinci_mcasp_probe, + .remove = davinci_mcasp_remove, + .driver = { + .name = "davinci-mcasp", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_mcasp_init(void) +{ + return platform_driver_register(&davinci_mcasp_driver); +} +module_init(davinci_mcasp_init); + +static void __exit davinci_mcasp_exit(void) +{ + platform_driver_unregister(&davinci_mcasp_driver); +} +module_exit(davinci_mcasp_exit); + +MODULE_AUTHOR("Steve Chen"); +MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h new file mode 100644 index 000000000000..554354c1cc2f --- /dev/null +++ b/sound/soc/davinci/davinci-mcasp.h @@ -0,0 +1,60 @@ +/* + * ALSA SoC McASP Audio Layer for TI DAVINCI processor + * + * MCASP related definitions + * + * Author: Nirmal Pandey <n-pandey@ti.com>, + * Suresh Rajashekara <suresh.r@ti.com> + * Steve Chen <schen@.mvista.com> + * + * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com> + * Copyright: (C) 2009 Texas Instruments, India + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef DAVINCI_MCASP_H +#define DAVINCI_MCASP_H + +#include <linux/io.h> +#include <mach/asp.h> +#include "davinci-pcm.h" + +extern struct snd_soc_dai davinci_mcasp_dai[]; + +#define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_96000 +#define DAVINCI_MCASP_I2S_DAI 0 +#define DAVINCI_MCASP_DIT_DAI 1 + +enum { + DAVINCI_AUDIO_WORD_8 = 0, + DAVINCI_AUDIO_WORD_12, + DAVINCI_AUDIO_WORD_16, + DAVINCI_AUDIO_WORD_20, + DAVINCI_AUDIO_WORD_24, + DAVINCI_AUDIO_WORD_32, + DAVINCI_AUDIO_WORD_28, /* This is only valid for McASP */ +}; + +struct davinci_audio_dev { + void __iomem *base; + int sample_rate; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; + unsigned int codec_fmt; + + /* McASP specific data */ + int tdm_slots; + u8 op_mode; + u8 num_serializer; + u8 *serial_dir; + u8 version; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; +}; + +#endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a05996588489..091dacb78b4d 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -67,6 +67,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; unsigned int data_type; + unsigned short acnt; unsigned int count; period_size = snd_pcm_lib_period_bytes(substream); @@ -91,11 +92,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } + acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); edma_set_src_index(lch, src_bidx, 0); edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) @@ -206,6 +208,7 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) /* Copy self-linked parameter RAM entry into master channel */ edma_read_slot(prtd->slave_lch, &temp); edma_write_slot(prtd->master_lch, &temp); + davinci_pcm_enqueue_dma(substream); return 0; } @@ -243,6 +246,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) int ret = 0; snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); if (prtd == NULL) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 62cb4eb07e34..63d96253c73a 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,17 +12,20 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <mach/edma.h> +#include <mach/asp.h> + + struct davinci_pcm_dma_params { - char *name; /* stream identifier */ - int channel; /* sync dma channel ID */ - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned int data_type; /* xfer data type */ + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + unsigned short acnt; + dma_addr_t dma_addr; /* device physical address for DMA */ + enum dma_event_q eventq_no; /* event queue number */ + unsigned char data_type; /* xfer data type */ + unsigned char convert_mono_stereo; }; -struct evm_snd_platform_data { - int tx_dma_ch; - int rx_dma_ch; -}; extern struct snd_soc_platform davinci_soc_platform; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index f0a2d4071998..9ff62e3a9b1d 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -69,6 +69,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task)) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 7eb549985d49..c4ae3e096bb9 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/of_device.h> #include <linux/of_platform.h> +#include <linux/delay.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -112,7 +113,7 @@ static void psc_ac97_cold_reset(struct snd_ac97 *ac97) out_8(®s->op1, MPC52xx_PSC_OP_RES); udelay(10); out_8(®s->op0, MPC52xx_PSC_OP_RES); - udelay(50); + msleep(1); psc_ac97_warm_reset(ac97); } diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig new file mode 100644 index 000000000000..a700562e8692 --- /dev/null +++ b/sound/soc/imx/Kconfig @@ -0,0 +1,21 @@ +config SND_MX1_MX2_SOC + tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" + depends on ARCH_MX2 || ARCH_MX1 + select SND_PCM + help + Say Y or M if you want to add support for codecs attached to + the MX1 or MX2 SSI interface. + +config SND_MXC_SOC_SSI + tristate + +config SND_SOC_MX27VIS_WM8974 + tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" + depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 + select SND_MXC_SOC_SSI + select SND_SOC_WM8974 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with WM8974. + + diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile new file mode 100644 index 000000000000..c2ffd2c8df5a --- /dev/null +++ b/sound/soc/imx/Makefile @@ -0,0 +1,10 @@ +# i.MX Platform Support +snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o +snd-soc-mxc-ssi-objs := mxc-ssi.o + +obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o +obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o + +# i.MX Machine Support +snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o +obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c new file mode 100644 index 000000000000..b83866529397 --- /dev/null +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -0,0 +1,488 @@ +/* + * mx1_mx2-pcm.c -- ALSA SoC interface for Freescale i.MX1x, i.MX2x CPUs + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin@vista-silicon.com + * + * Based on mxc-pcm.c by Liam Girdwood. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma-mx1-mx2.h> + +#include "mx1_mx2-pcm.h" + + +static const struct snd_pcm_hardware mx1_mx2_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = 32 * 1024, + .period_bytes_min = 64, + .period_bytes_max = 8 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +struct mx1_mx2_runtime_data { + int dma_ch; + int active; + unsigned int period; + unsigned int periods; + int tx_spin; + spinlock_t dma_lock; + struct mx1_mx2_pcm_dma_params *dma_params; +}; + + +/** + * This function stops the current dma transfer for playback + * and clears the dma pointers. + * + * @param substream pointer to the structure of the current stream. + * + */ +static int audio_stop_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned long flags; + + spin_lock_irqsave(&prtd->dma_lock, flags); + + pr_debug("%s\n", __func__); + + prtd->active = 0; + prtd->period = 0; + prtd->periods = 0; + + /* this stops the dma channel and clears the buffer ptrs */ + + imx_dma_disable(prtd->dma_ch); + + spin_unlock_irqrestore(&prtd->dma_lock, flags); + + return 0; +} + +/** + * This function is called whenever a new audio block needs to be + * transferred to the codec. The function receives the address and the size + * of the new block and start a new DMA transfer. + * + * @param substream pointer to the structure of the current stream. + * + */ +static int dma_new_period(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned int dma_size; + unsigned int offset; + int ret = 0; + dma_addr_t mem_addr; + unsigned int dev_addr; + + if (prtd->active) { + dma_size = frames_to_bytes(runtime, runtime->period_size); + offset = dma_size * prtd->period; + + pr_debug("%s: period (%d) out of (%d)\n", __func__, + prtd->period, + runtime->periods); + pr_debug("period_size %d frames\n offset %d bytes\n", + (unsigned int)runtime->period_size, + offset); + pr_debug("dma_size %d bytes\n", dma_size); + + snd_BUG_ON(dma_size > mx1_mx2_pcm_hardware.period_bytes_max); + + mem_addr = (dma_addr_t)(runtime->dma_addr + offset); + dev_addr = prtd->dma_params->per_address; + pr_debug("%s: mem_addr is %x\n dev_addr is %x\n", + __func__, mem_addr, dev_addr); + + ret = imx_dma_setup_single(prtd->dma_ch, mem_addr, + dma_size, dev_addr, + prtd->dma_params->transfer_type); + if (ret < 0) { + printk(KERN_ERR "Error %d configuring DMA\n", ret); + return ret; + } + imx_dma_enable(prtd->dma_ch); + + pr_debug("%s: transfer enabled\nmem_addr = %x\n", + __func__, (unsigned int) mem_addr); + pr_debug("dev_addr = %x\ndma_size = %d\n", + (unsigned int) dev_addr, dma_size); + + prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ + prtd->period++; + prtd->period %= runtime->periods; + } + return ret; +} + + +/** + * This is a callback which will be called + * when a TX transfer finishes. The call occurs + * in interrupt context. + * + * @param dat pointer to the structure of the current stream. + * + */ +static void audio_dma_irq(int channel, void *data) +{ + struct snd_pcm_substream *substream; + struct snd_pcm_runtime *runtime; + struct mx1_mx2_runtime_data *prtd; + unsigned int dma_size; + unsigned int previous_period; + unsigned int offset; + + substream = data; + runtime = substream->runtime; + prtd = runtime->private_data; + previous_period = prtd->periods; + dma_size = frames_to_bytes(runtime, runtime->period_size); + offset = dma_size * previous_period; + + prtd->tx_spin = 0; + prtd->periods++; + prtd->periods %= runtime->periods; + + pr_debug("%s: irq per %d offset %x\n", __func__, prtd->periods, offset); + + /* + * If we are getting a callback for an active stream then we inform + * the PCM middle layer we've finished a period + */ + if (prtd->active) + snd_pcm_period_elapsed(substream); + + /* + * Trig next DMA transfer + */ + dma_new_period(substream); +} + +/** + * This function configures the hardware to allow audio + * playback operations. It is called by ALSA framework. + * + * @param substream pointer to the structure of the current stream. + * + * @return 0 on success, -1 otherwise. + */ +static int +snd_mx1_mx2_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + prtd->period = 0; + prtd->periods = 0; + + return 0; +} + +static int mx1_mx2_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_ERR "%s: Error %d failed to malloc pcm pages \n", + __func__, ret); + return ret; + } + + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_addr 0x(%x)\n", + __func__, (unsigned int)runtime->dma_addr); + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_area 0x(%x)\n", + __func__, (unsigned int)runtime->dma_area); + pr_debug("%s: snd_imx1_mx2_audio_hw_params runtime->dma_bytes 0x(%x)\n", + __func__, (unsigned int)runtime->dma_bytes); + + return ret; +} + +static int mx1_mx2_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + imx_dma_free(prtd->dma_ch); + + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int mx1_mx2_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct mx1_mx2_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + prtd->tx_spin = 0; + /* requested stream startup */ + prtd->active = 1; + pr_debug("%s: starting dma_new_period\n", __func__); + ret = dma_new_period(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + /* requested stream shutdown */ + pr_debug("%s: stopping dma transfer\n", __func__); + ret = audio_stop_dma(substream); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static snd_pcm_uframes_t +mx1_mx2_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + unsigned int offset = 0; + + /* tx_spin value is used here to check if a transfer is active */ + if (prtd->tx_spin) { + offset = (runtime->period_size * (prtd->periods)) + + (runtime->period_size >> 1); + if (offset >= runtime->buffer_size) + offset = runtime->period_size >> 1; + } else { + offset = (runtime->period_size * (prtd->periods)); + if (offset >= runtime->buffer_size) + offset = 0; + } + pr_debug("%s: pointer offset %x\n", __func__, offset); + + return offset; +} + +static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mx1_mx2_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int ret; + + snd_soc_set_runtime_hwparams(substream, &mx1_mx2_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + prtd = kzalloc(sizeof(struct mx1_mx2_runtime_data), GFP_KERNEL); + if (prtd == NULL) { + ret = -ENOMEM; + goto out; + } + + runtime->private_data = prtd; + + if (!dma_data) + return -ENODEV; + + prtd->dma_params = dma_data; + + pr_debug("%s: Requesting dma channel (%s)\n", __func__, + prtd->dma_params->name); + prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, + DMA_PRIO_HIGH); + if (prtd->dma_ch < 0) { + printk(KERN_ERR "Error %d requesting dma channel\n", ret); + return ret; + } + imx_dma_config_burstlen(prtd->dma_ch, + prtd->dma_params->watermark_level); + + ret = imx_dma_config_channel(prtd->dma_ch, + prtd->dma_params->per_config, + prtd->dma_params->mem_config, + prtd->dma_params->event_id, 0); + + if (ret) { + pr_debug(KERN_ERR "Error %d configuring dma channel %d\n", + ret, prtd->dma_ch); + return ret; + } + + pr_debug("%s: Setting tx dma callback function\n", __func__); + ret = imx_dma_setup_handlers(prtd->dma_ch, + audio_dma_irq, NULL, + (void *)substream); + if (ret < 0) { + printk(KERN_ERR "Error %d setting dma callback function\n", ret); + return ret; + } + return 0; + + out: + return ret; +} + +static int mx1_mx2_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mx1_mx2_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int mx1_mx2_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops mx1_mx2_pcm_ops = { + .open = mx1_mx2_pcm_open, + .close = mx1_mx2_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mx1_mx2_pcm_hw_params, + .hw_free = mx1_mx2_pcm_hw_free, + .prepare = snd_mx1_mx2_prepare, + .trigger = mx1_mx2_pcm_trigger, + .pointer = mx1_mx2_pcm_pointer, + .mmap = mx1_mx2_pcm_mmap, +}; + +static u64 mx1_mx2_pcm_dmamask = 0xffffffff; + +static int mx1_mx2_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = mx1_mx2_pcm_hardware.buffer_bytes_max; + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Reserve uncached-buffered memory area for DMA */ + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + pr_debug("%s: preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + __func__, (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void mx1_mx2_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static int mx1_mx2_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &mx1_mx2_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + pr_debug("%s: preallocate playback buffer\n", __func__); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = mx1_mx2_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + pr_debug("%s: preallocate capture buffer\n", __func__); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform mx1_mx2_soc_platform = { + .name = "mx1_mx2-audio", + .pcm_ops = &mx1_mx2_pcm_ops, + .pcm_new = mx1_mx2_pcm_new, + .pcm_free = mx1_mx2_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(mx1_mx2_soc_platform); + +static int __init mx1_mx2_soc_platform_init(void) +{ + return snd_soc_register_platform(&mx1_mx2_soc_platform); +} +module_init(mx1_mx2_soc_platform_init); + +static void __exit mx1_mx2_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&mx1_mx2_soc_platform); +} +module_exit(mx1_mx2_soc_platform_exit); + +MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com"); +MODULE_DESCRIPTION("Freescale i.MX2x, i.MX1x PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mx1_mx2-pcm.h b/sound/soc/imx/mx1_mx2-pcm.h new file mode 100644 index 000000000000..2e528106570b --- /dev/null +++ b/sound/soc/imx/mx1_mx2-pcm.h @@ -0,0 +1,26 @@ +/* + * mx1_mx2-pcm.h :- ASoC platform header for Freescale i.MX1x, i.MX2x + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MX1_MX2_PCM_H +#define _MX1_MX2_PCM_H + +/* DMA information for mx1_mx2 platforms */ +struct mx1_mx2_pcm_dma_params { + char *name; /* stream identifier */ + unsigned int transfer_type; /* READ or WRITE DMA transfer */ + dma_addr_t per_address; /* physical address of SSI fifo */ + int event_id; /* fixed DMA number for SSI fifo */ + int watermark_level; /* SSI fifo watermark level */ + int per_config; /* DMA Config flags for peripheral */ + int mem_config; /* DMA Config flags for RAM */ + }; + +/* platform data */ +extern struct snd_soc_platform mx1_mx2_soc_platform; + +#endif diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c new file mode 100644 index 000000000000..e4dcb539108a --- /dev/null +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -0,0 +1,317 @@ +/* + * mx27vis_wm8974.c -- SoC audio for mx27vis + * + * Copyright 2009 Vista Silicon S.L. + * Author: Javier Martin + * javier.martin@vista-silicon.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + + +#include "../codecs/wm8974.h" +#include "mx1_mx2-pcm.h" +#include "mxc-ssi.h" +#include <mach/gpio.h> +#include <mach/iomux.h> + +#define IGNORED_ARG 0 + + +static struct snd_soc_card mx27vis; + +/** + * This function connects SSI1 (HPCR1) as slave to + * SSI1 external signals (PPCR1) + * As slave, HPCR1 must set TFSDIR and TCLKDIR as inputs from + * port 4 + */ +void audmux_connect_1_4(void) +{ + pr_debug("AUDMUX: normal operation mode\n"); + /* Reset HPCR1 and PPCR1 */ + + DAM_HPCR1 = 0x00000000; + DAM_PPCR1 = 0x00000000; + + /* set to synchronous */ + DAM_HPCR1 |= AUDMUX_HPCR_SYN; + DAM_PPCR1 |= AUDMUX_PPCR_SYN; + + + /* set Rx sources 1 <--> 4 */ + DAM_HPCR1 |= AUDMUX_HPCR_RXDSEL(3); /* port 4 */ + DAM_PPCR1 |= AUDMUX_PPCR_RXDSEL(0); /* port 1 */ + + /* set Tx frame and Clock direction and source 4 --> 1 output */ + DAM_HPCR1 |= AUDMUX_HPCR_TFSDIR | AUDMUX_HPCR_TCLKDIR; + DAM_HPCR1 |= AUDMUX_HPCR_TFCSEL(3); /* TxDS and TxCclk from port 4 */ + + return; +} + +static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0, fmt = 0, mclk = 0; + int ret = 0; + + /* + * The WM8974 is better at generating accurate audio clocks than the + * MX27 SSI controller, so we will use it as master when we can. + */ + switch (params_rate(params)) { + case 8000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + mclk = WM8974_MCLKDIV_12; + pll_out = 24576000; + break; + case 16000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + pll_out = 12288000; + break; + case 48000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + pll_out = 12288000; + break; + case 96000: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 12288000; + break; + case 11025: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_16; + pll_out = 11289600; + break; + case 22050: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_8; + pll_out = 11289600; + break; + case 44100: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_4; + mclk = WM8974_MCLKDIV_2; + pll_out = 11289600; + break; + case 88200: + fmt = SND_SOC_DAIFMT_CBM_CFM; + bclk = WM8974_BCLKDIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = codec_dai->ops->set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from codec DAI configuration\n"); + return ret; + } + + /* set cpu DAI configuration */ + ret = cpu_dai->ops->set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_SYNC | fmt); + if (ret < 0) { + printk(KERN_ERR "Error from cpu DAI configuration\n"); + return ret; + } + + /* Put DC field of STCCR to 1 (not zero) */ + ret = cpu_dai->ops->set_tdm_slot(cpu_dai, 0, 2); + + /* set the SSI system clock as input */ + ret = cpu_dai->ops->set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "Error when setting system SSI clk\n"); + return ret; + } + + /* set codec BCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_BCLKDIV, bclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting BCLK division\n"); + return ret; + } + + + /* codec PLL input is 25 MHz */ + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + 25000000, pll_out); + if (ret < 0) { + printk(KERN_ERR "Error when setting PLL input\n"); + return ret; + } + + /*set codec MCLK division for sample rate */ + ret = codec_dai->ops->set_clkdiv(codec_dai, WM8974_MCLKDIV, mclk); + if (ret < 0) { + printk(KERN_ERR "Error when setting MCLK division\n"); + return ret; + } + + return 0; +} + +static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); +} + +/* + * mx27vis WM8974 HiFi DAI opserations. + */ +static struct snd_soc_ops mx27vis_hifi_ops = { + .hw_params = mx27vis_hifi_hw_params, + .hw_free = mx27vis_hifi_hw_free, +}; + + +static int mx27vis_suspend(struct platform_device *pdev, pm_message_t state) +{ + return 0; +} + +static int mx27vis_resume(struct platform_device *pdev) +{ + return 0; +} + +static int mx27vis_probe(struct platform_device *pdev) +{ + int ret = 0; + + ret = get_ssi_clk(0, &pdev->dev); + + if (ret < 0) { + printk(KERN_ERR "%s: cant get ssi clock\n", __func__); + return ret; + } + + + return 0; +} + +static int mx27vis_remove(struct platform_device *pdev) +{ + put_ssi_clk(0); + return 0; +} + +static struct snd_soc_dai_link mx27vis_dai[] = { +{ /* Hifi Playback*/ + .name = "WM8974", + .stream_name = "WM8974 HiFi", + .cpu_dai = &imx_ssi_pcm_dai[0], + .codec_dai = &wm8974_dai, + .ops = &mx27vis_hifi_ops, +}, +}; + +static struct snd_soc_card mx27vis = { + .name = "mx27vis", + .platform = &mx1_mx2_soc_platform, + .probe = mx27vis_probe, + .remove = mx27vis_remove, + .suspend_pre = mx27vis_suspend, + .resume_post = mx27vis_resume, + .dai_link = mx27vis_dai, + .num_links = ARRAY_SIZE(mx27vis_dai), +}; + +static struct snd_soc_device mx27vis_snd_devdata = { + .card = &mx27vis, + .codec_dev = &soc_codec_dev_wm8974, +}; + +static struct platform_device *mx27vis_snd_device; + +/* Temporal definition of board specific behaviour */ +void gpio_ssi_active(int ssi_num) +{ + int ret = 0; + + unsigned int ssi1_pins[] = { + PC20_PF_SSI1_FS, + PC21_PF_SSI1_RXD, + PC22_PF_SSI1_TXD, + PC23_PF_SSI1_CLK, + }; + unsigned int ssi2_pins[] = { + PC24_PF_SSI2_FS, + PC25_PF_SSI2_RXD, + PC26_PF_SSI2_TXD, + PC27_PF_SSI2_CLK, + }; + if (ssi_num == 0) + ret = mxc_gpio_setup_multiple_pins(ssi1_pins, + ARRAY_SIZE(ssi1_pins), "USB OTG"); + else + ret = mxc_gpio_setup_multiple_pins(ssi2_pins, + ARRAY_SIZE(ssi2_pins), "USB OTG"); + if (ret) + printk(KERN_ERR "Error requesting ssi %x pins\n", ssi_num); +} + + +static int __init mx27vis_init(void) +{ + int ret; + + mx27vis_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_snd_device, &mx27vis_snd_devdata); + mx27vis_snd_devdata.dev = &mx27vis_snd_device->dev; + ret = platform_device_add(mx27vis_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_snd_device); + } + + /* WM8974 uses SSI1 (HPCR1) via AUDMUX port 4 for audio (PPCR1) */ + gpio_ssi_active(0); + audmux_connect_1_4(); + + return ret; +} + +static void __exit mx27vis_exit(void) +{ + /* We should call some "ssi_gpio_inactive()" properly */ +} + +module_init(mx27vis_init); +module_exit(mx27vis_exit); + + +MODULE_AUTHOR("Javier Martin, javier.martin@vista-silicon.com"); +MODULE_DESCRIPTION("ALSA SoC WM8974 mx27vis"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c new file mode 100644 index 000000000000..3806ff2c0cd4 --- /dev/null +++ b/sound/soc/imx/mxc-ssi.c @@ -0,0 +1,868 @@ +/* + * mxc-ssi.c -- SSI driver for Freescale IMX + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * + * Based on mxc-alsa-mc13783 (C) 2006 Freescale. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * Need to rework SSI register defs when new defs go into mainline. + * Add support for TDM and FIFO 1. + * Add support for i.mx3x DMA interface. + * + */ + + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <mach/dma-mx1-mx2.h> +#include <asm/mach-types.h> + +#include "mxc-ssi.h" +#include "mx1_mx2-pcm.h" + +#define SSI1_PORT 0 +#define SSI2_PORT 1 + +static int ssi_active[2] = {0, 0}; + +/* DMA information for mx1_mx2 platforms */ +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out0 = { + .name = "SSI1 PCM Stereo out 0", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI1_BASE_ADDR + STX0, + .event_id = DMA_REQ_SSI1_TX0, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_out1 = { + .name = "SSI1 PCM Stereo out 1", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI1_BASE_ADDR + STX1, + .event_id = DMA_REQ_SSI1_TX1, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in0 = { + .name = "SSI1 PCM Stereo in 0", + .transfer_type = DMA_MODE_READ, + .per_address = SSI1_BASE_ADDR + SRX0, + .event_id = DMA_REQ_SSI1_RX0, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi1_pcm_stereo_in1 = { + .name = "SSI1 PCM Stereo in 1", + .transfer_type = DMA_MODE_READ, + .per_address = SSI1_BASE_ADDR + SRX1, + .event_id = DMA_REQ_SSI1_RX1, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out0 = { + .name = "SSI2 PCM Stereo out 0", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI2_BASE_ADDR + STX0, + .event_id = DMA_REQ_SSI2_TX0, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_out1 = { + .name = "SSI2 PCM Stereo out 1", + .transfer_type = DMA_MODE_WRITE, + .per_address = SSI2_BASE_ADDR + STX1, + .event_id = DMA_REQ_SSI2_TX1, + .watermark_level = TXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in0 = { + .name = "SSI2 PCM Stereo in 0", + .transfer_type = DMA_MODE_READ, + .per_address = SSI2_BASE_ADDR + SRX0, + .event_id = DMA_REQ_SSI2_RX0, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct mx1_mx2_pcm_dma_params imx_ssi2_pcm_stereo_in1 = { + .name = "SSI2 PCM Stereo in 1", + .transfer_type = DMA_MODE_READ, + .per_address = SSI2_BASE_ADDR + SRX1, + .event_id = DMA_REQ_SSI2_RX1, + .watermark_level = RXFIFO_WATERMARK, + .per_config = IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + .mem_config = IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, +}; + +static struct clk *ssi_clk0, *ssi_clk1; + +int get_ssi_clk(int ssi, struct device *dev) +{ + switch (ssi) { + case 0: + ssi_clk0 = clk_get(dev, "ssi1"); + if (IS_ERR(ssi_clk0)) + return PTR_ERR(ssi_clk0); + return 0; + case 1: + ssi_clk1 = clk_get(dev, "ssi2"); + if (IS_ERR(ssi_clk1)) + return PTR_ERR(ssi_clk1); + return 0; + default: + return -EINVAL; + } +} +EXPORT_SYMBOL(get_ssi_clk); + +void put_ssi_clk(int ssi) +{ + switch (ssi) { + case 0: + clk_put(ssi_clk0); + ssi_clk0 = NULL; + break; + case 1: + clk_put(ssi_clk1); + ssi_clk1 = NULL; + break; + } +} +EXPORT_SYMBOL(put_ssi_clk); + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 scr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + scr = SSI1_SCR; + pr_debug("%s: SCR for SSI1 is %x\n", __func__, scr); + } else { + scr = SSI2_SCR; + pr_debug("%s: SCR for SSI2 is %x\n", __func__, scr); + } + + if (scr & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) { + scr |= SSI_SCR_SYS_CLK_EN; + pr_debug("%s: clk of is output\n", __func__); + } else { + scr &= ~SSI_SCR_SYS_CLK_EN; + pr_debug("%s: clk of is input\n", __func__); + } + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + pr_debug("%s: writeback of SSI1_SCR\n", __func__); + SSI1_SCR = scr; + } else { + pr_debug("%s: writeback of SSI2_SCR\n", __func__); + SSI2_SCR = scr; + } + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + u32 stccr, srccr; + + pr_debug("%s\n", __func__); + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) + return 0; + srccr = SSI1_STCCR; + stccr = SSI1_STCCR; + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) + return 0; + srccr = SSI2_STCCR; + stccr = SSI2_STCCR; + } + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCCR = stccr; + SSI1_SRCCR = srccr; + } else { + SSI2_STCCR = stccr; + SSI2_SRCCR = srccr; + } + return 0; +} + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int mask, int slots) +{ + u32 stmsk, srmsk, stccr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + stccr = SSI1_STCCR; + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + stccr = SSI2_STCCR; + } + + stmsk = srmsk = mask; + stccr &= ~SSI_STCCR_DC_MASK; + stccr |= SSI_STCCR_DC(slots - 1); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STMSK = stmsk; + SSI1_SRMSK = srmsk; + SSI1_SRCCR = SSI1_STCCR = stccr; + } else { + SSI2_STMSK = stmsk; + SSI2_SRMSK = srmsk; + SSI2_SRCCR = SSI2_STCCR = stccr; + } + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + u32 stcr = 0, srcr = 0, scr; + + /* + * This is done to avoid this function to modify + * previous set values in stcr + */ + stcr = SSI1_STCR; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + scr = SSI1_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); + else + scr = SSI2_SCR & ~(SSI_SCR_SYN | SSI_SCR_NET); + + if (scr & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + stcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + srcr |= SSI_SRCR_RFSI | SSI_SRCR_REFS | SSI_SRCR_RXBIT0; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + stcr |= SSI_STCR_TXBIT0; + srcr |= SSI_SRCR_RXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + stcr |= SSI_STCR_TFSL; + srcr |= SSI_SRCR_RFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + stcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + srcr |= SSI_SRCR_RFSL | SSI_SRCR_REFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + stcr |= SSI_STCR_TFSI; + stcr &= ~SSI_STCR_TSCKP; + srcr |= SSI_SRCR_RFSI; + srcr &= ~SSI_SRCR_RSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + stcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + srcr &= ~(SSI_SRCR_RSCKP | SSI_SRCR_RFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + stcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + srcr |= SSI_SRCR_RFSI | SSI_SRCR_RSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + stcr &= ~SSI_STCR_TFSI; + stcr |= SSI_STCR_TSCKP; + srcr &= ~SSI_SRCR_RFSI; + srcr |= SSI_SRCR_RSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + stcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + srcr |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + stcr |= SSI_STCR_TFDIR; + srcr |= SSI_SRCR_RFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + stcr |= SSI_STCR_TXDIR; + srcr |= SSI_SRCR_RXDIR; + break; + } + + /* sync */ + if (!(fmt & SND_SOC_DAIFMT_ASYNC)) + scr |= SSI_SCR_SYN; + + /* tdm - only for stereo atm */ + if (fmt & SND_SOC_DAIFMT_TDM) + scr |= SSI_SCR_NET; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCR = stcr; + SSI1_SRCR = srcr; + SSI1_SCR = scr; + } else { + SSI2_STCR = stcr; + SSI2_SRCR = srcr; + SSI2_SCR = scr; + } + + return 0; +} + +static int imx_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* set up TX DMA params */ + switch (cpu_dai->id) { + case IMX_DAI_SSI0: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out0; + break; + case IMX_DAI_SSI1: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_out1; + break; + case IMX_DAI_SSI2: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out0; + break; + case IMX_DAI_SSI3: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_out1; + } + pr_debug("%s: (playback)\n", __func__); + } else { + /* set up RX DMA params */ + switch (cpu_dai->id) { + case IMX_DAI_SSI0: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in0; + break; + case IMX_DAI_SSI1: + cpu_dai->dma_data = &imx_ssi1_pcm_stereo_in1; + break; + case IMX_DAI_SSI2: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in0; + break; + case IMX_DAI_SSI3: + cpu_dai->dma_data = &imx_ssi2_pcm_stereo_in1; + } + pr_debug("%s: (capture)\n", __func__); + } + + /* + * we cant really change any SSI values after SSI is enabled + * need to fix in software for max flexibility - lrg + */ + if (cpu_dai->active) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + + /* reset the SSI port - Sect 45.4.4 */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + + if (!ssi_clk0) + return -EINVAL; + + if (ssi_active[SSI1_PORT]++) { + pr_debug("%s: exit before reset\n", __func__); + return 0; + } + + /* SSI1 Reset */ + SSI1_SCR = 0; + + SSI1_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | + SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | + SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | + SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); + } else { + + if (!ssi_clk1) + return -EINVAL; + + if (ssi_active[SSI2_PORT]++) { + pr_debug("%s: exit before reset\n", __func__); + return 0; + } + + /* SSI2 Reset */ + SSI2_SCR = 0; + + SSI2_SFCSR = SSI_SFCSR_RFWM1(RXFIFO_WATERMARK) | + SSI_SFCSR_RFWM0(RXFIFO_WATERMARK) | + SSI_SFCSR_TFWM1(TXFIFO_WATERMARK) | + SSI_SFCSR_TFWM0(TXFIFO_WATERMARK); + } + + return 0; +} + +int imx_ssi_hw_tx_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 stccr, stcr, sier; + + pr_debug("%s\n", __func__); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + stccr = SSI1_STCCR & ~SSI_STCCR_WL_MASK; + stcr = SSI1_STCR; + sier = SSI1_SIER; + } else { + stccr = SSI2_STCCR & ~SSI_STCCR_WL_MASK; + stcr = SSI2_STCR; + sier = SSI2_SIER; + } + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + stccr |= SSI_STCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + stccr |= SSI_STCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + stccr |= SSI_STCCR_WL(24); + break; + } + + /* enable interrupts */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + stcr |= SSI_STCR_TFEN0; + else + stcr |= SSI_STCR_TFEN1; + sier |= SSI_SIER_TDMAE; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_STCR = stcr; + SSI1_STCCR = stccr; + SSI1_SIER = sier; + } else { + SSI2_STCR = stcr; + SSI2_STCCR = stccr; + SSI2_SIER = sier; + } + + return 0; +} + +int imx_ssi_hw_rx_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 srccr, srcr, sier; + + pr_debug("%s\n", __func__); + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + srccr = SSI1_SRCCR & ~SSI_SRCCR_WL_MASK; + srcr = SSI1_SRCR; + sier = SSI1_SIER; + } else { + srccr = SSI2_SRCCR & ~SSI_SRCCR_WL_MASK; + srcr = SSI2_SRCR; + sier = SSI2_SIER; + } + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + srccr |= SSI_SRCCR_WL(24); + break; + } + + /* enable interrupts */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + srcr |= SSI_SRCR_RFEN0; + else + srcr |= SSI_SRCR_RFEN1; + sier |= SSI_SIER_RDMAE; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + SSI1_SRCR = srcr; + SSI1_SRCCR = srccr; + SSI1_SIER = sier; + } else { + SSI2_SRCR = srcr; + SSI2_SRCCR = srccr; + SSI2_SIER = sier; + } + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + int ret; + + /* cant change any parameters when SSI is running */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + if (SSI1_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + } else { + if (SSI2_SCR & SSI_SCR_SSIEN) { + printk(KERN_WARNING "Warning ssi already enabled\n"); + return 0; + } + } + + /* + * Configure both tx and rx params with the same settings. This is + * really a harware restriction because SSI must be disabled until + * we can change those values. If there is an active audio stream in + * one direction, enabling the other direction with different + * settings would mean disturbing the running one. + */ + ret = imx_ssi_hw_tx_params(substream, params); + if (ret < 0) + return ret; + return imx_ssi_hw_rx_params(substream, params); +} + +int imx_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + pr_debug("%s\n", __func__); + + /* Enable clks here to follow SSI recommended init sequence */ + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { + ret = clk_enable(ssi_clk0); + if (ret < 0) + printk(KERN_ERR "Unable to enable ssi_clk0\n"); + } else { + ret = clk_enable(ssi_clk1); + if (ret < 0) + printk(KERN_ERR "Unable to enable ssi_clk1\n"); + } + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + u32 scr; + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + scr = SSI1_SCR; + else + scr = SSI2_SCR; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE | SSI_SCR_SSIEN; + else + scr |= SSI_SCR_RE | SSI_SCR_SSIEN; + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + break; + default: + return -EINVAL; + } + + if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) + SSI1_SCR = scr; + else + SSI2_SCR = scr; + + return 0; +} + +static void imx_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + /* shutdown SSI if neither Tx or Rx is active */ + if (!cpu_dai->active) { + + if (cpu_dai->id == IMX_DAI_SSI0 || + cpu_dai->id == IMX_DAI_SSI2) { + + if (--ssi_active[SSI1_PORT] > 1) + return; + + SSI1_SCR = 0; + clk_disable(ssi_clk0); + } else { + if (--ssi_active[SSI2_PORT]) + return; + SSI2_SCR = 0; + clk_disable(ssi_clk1); + } + } +} + +#ifdef CONFIG_PM +static int imx_ssi_suspend(struct platform_device *dev, + struct snd_soc_dai *dai) +{ + return 0; +} + +static int imx_ssi_resume(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + return 0; +} + +#else +#define imx_ssi_suspend NULL +#define imx_ssi_resume NULL +#endif + +#define IMX_SSI_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define IMX_SSI_BITS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .startup = imx_ssi_startup, + .shutdown = imx_ssi_shutdown, + .trigger = imx_ssi_trigger, + .prepare = imx_ssi_prepare, + .hw_params = imx_ssi_hw_params, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_fmt = imx_ssi_set_dai_fmt, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, +}; + +struct snd_soc_dai imx_ssi_pcm_dai[] = { +{ + .name = "imx-i2s-1-0", + .id = IMX_DAI_SSI0, + .suspend = imx_ssi_suspend, + .resume = imx_ssi_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-2-0", + .id = IMX_DAI_SSI1, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-1-1", + .id = IMX_DAI_SSI2, + .suspend = imx_ssi_suspend, + .resume = imx_ssi_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +{ + .name = "imx-i2s-2-1", + .id = IMX_DAI_SSI3, + .playback = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .formats = IMX_SSI_BITS, + .rates = IMX_SSI_RATES,}, + .ops = &imx_ssi_pcm_dai_ops, +}, +}; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int __init imx_ssi_init(void) +{ + return snd_soc_register_dais(imx_ssi_pcm_dai, + ARRAY_SIZE(imx_ssi_pcm_dai)); +} + +static void __exit imx_ssi_exit(void) +{ + snd_soc_unregister_dais(imx_ssi_pcm_dai, + ARRAY_SIZE(imx_ssi_pcm_dai)); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); +MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com"); +MODULE_DESCRIPTION("i.MX ASoC I2S driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/imx/mxc-ssi.h b/sound/soc/imx/mxc-ssi.h new file mode 100644 index 000000000000..12bbdc9c7ecd --- /dev/null +++ b/sound/soc/imx/mxc-ssi.h @@ -0,0 +1,238 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#include <mach/hardware.h> + +/* SSI regs definition - MOVE to /arch/arm/plat-mxc/include/mach/ when stable */ +#define SSI1_IO_BASE_ADDR IO_ADDRESS(SSI1_BASE_ADDR) +#define SSI2_IO_BASE_ADDR IO_ADDRESS(SSI2_BASE_ADDR) + +#define STX0 0x00 +#define STX1 0x04 +#define SRX0 0x08 +#define SRX1 0x0c +#define SCR 0x10 +#define SISR 0x14 +#define SIER 0x18 +#define STCR 0x1c +#define SRCR 0x20 +#define STCCR 0x24 +#define SRCCR 0x28 +#define SFCSR 0x2c +#define STR 0x30 +#define SOR 0x34 +#define SACNT 0x38 +#define SACADD 0x3c +#define SACDAT 0x40 +#define SATAG 0x44 +#define STMSK 0x48 +#define SRMSK 0x4c + +#define SSI1_STX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX0))) +#define SSI1_STX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STX1))) +#define SSI1_SRX0 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX0))) +#define SSI1_SRX1 (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRX1))) +#define SSI1_SCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SCR))) +#define SSI1_SISR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SISR))) +#define SSI1_SIER (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SIER))) +#define SSI1_STCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCR))) +#define SSI1_SRCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCR))) +#define SSI1_STCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STCCR))) +#define SSI1_SRCCR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRCCR))) +#define SSI1_SFCSR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SFCSR))) +#define SSI1_STR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STR))) +#define SSI1_SOR (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SOR))) +#define SSI1_SACNT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACNT))) +#define SSI1_SACADD (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACADD))) +#define SSI1_SACDAT (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SACDAT))) +#define SSI1_SATAG (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SATAG))) +#define SSI1_STMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + STMSK))) +#define SSI1_SRMSK (*((volatile u32 *)(SSI1_IO_BASE_ADDR + SRMSK))) + + +#define SSI2_STX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX0))) +#define SSI2_STX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STX1))) +#define SSI2_SRX0 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX0))) +#define SSI2_SRX1 (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRX1))) +#define SSI2_SCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SCR))) +#define SSI2_SISR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SISR))) +#define SSI2_SIER (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SIER))) +#define SSI2_STCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCR))) +#define SSI2_SRCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCR))) +#define SSI2_STCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STCCR))) +#define SSI2_SRCCR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRCCR))) +#define SSI2_SFCSR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SFCSR))) +#define SSI2_STR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STR))) +#define SSI2_SOR (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SOR))) +#define SSI2_SACNT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACNT))) +#define SSI2_SACADD (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACADD))) +#define SSI2_SACDAT (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SACDAT))) +#define SSI2_SATAG (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SATAG))) +#define SSI2_STMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + STMSK))) +#define SSI2_SRMSK (*((volatile u32 *)(SSI2_IO_BASE_ADDR + SRMSK))) + +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 15) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 15) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + + +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) + +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (x << 4) +#define SSI_SACNT_RD (x << 3) +#define SSI_SACNT_TIF (x << 2) +#define SSI_SACNT_FV (x << 1) +#define SSI_SACNT_AC97EN (x << 0) + +/* Watermarks for FIFO's */ +#define TXFIFO_WATERMARK 0x4 +#define RXFIFO_WATERMARK 0x4 + +/* i.MX DAI SSP ID's */ +#define IMX_DAI_SSI0 0 /* SSI1 FIFO 0 */ +#define IMX_DAI_SSI1 1 /* SSI1 FIFO 1 */ +#define IMX_DAI_SSI2 2 /* SSI2 FIFO 0 */ +#define IMX_DAI_SSI3 3 /* SSI2 FIFO 1 */ + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + + +/* SSI Div 2 */ +#define IMX_SSI_DIV_2_OFF (~SSI_STCCR_DIV2) +#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2 + +extern struct snd_soc_dai imx_ssi_pcm_dai[4]; +extern int get_ssi_clk(int ssi, struct device *dev); +extern void put_ssi_clk(int ssi); +#endif diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index b771238662b6..2dee9839be86 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on SND_OMAP_SOC && MACH_AMS_DELTA + select SND_OMAP_SOC_MCBSP + select SND_SOC_CX20442 + help + Say Y if you want to add support for SoC audio on Amstrad Delta. + config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C @@ -72,4 +80,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE help Say Y if you want to add support for SoC audio on the Beagleboard. +config SND_OMAP_SOC_ZOOM2 + tristate "SoC Audio support for Zoom2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on Zoom2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index a37f49862389..02d69471dcb5 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o @@ -14,8 +15,10 @@ snd-soc-omap3evm-objs := omap3evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o +snd-soc-zoom2-objs := zoom2.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o @@ -23,3 +26,4 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o +obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c new file mode 100644 index 000000000000..5a5166ac7279 --- /dev/null +++ b/sound/soc/omap/ams-delta.c @@ -0,0 +1,646 @@ +/* + * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone + * + * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> + * + * Initially based on sound/soc/omap/osk5912.x + * Copyright (C) 2008 Mistral Solutions + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/gpio.h> +#include <linux/spinlock.h> +#include <linux/tty.h> + +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include <mach/board-ams-delta.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/cx20442.h" + + +/* Board specific DAPM widgets */ + const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { + /* Handset */ + SND_SOC_DAPM_MIC("Mouthpiece", NULL), + SND_SOC_DAPM_HP("Earpiece", NULL), + /* Handsfree/Speakerphone */ + SND_SOC_DAPM_MIC("Microphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +/* How they are connected to codec pins */ +static const struct snd_soc_dapm_route ams_delta_audio_map[] = { + {"TELIN", NULL, "Mouthpiece"}, + {"Earpiece", NULL, "TELOUT"}, + + {"MIC", NULL, "Microphone"}, + {"Speaker", NULL, "SPKOUT"}, +}; + +/* + * Controls, functional after the modem line discipline is activated. + */ + +/* Virtual switch: audio input/output constellations */ +static const char *ams_delta_audio_mode[] = + {"Mixed", "Handset", "Handsfree", "Speakerphone"}; + +/* Selection <-> pin translation */ +#define AMS_DELTA_MOUTHPIECE 0 +#define AMS_DELTA_EARPIECE 1 +#define AMS_DELTA_MICROPHONE 2 +#define AMS_DELTA_SPEAKER 3 +#define AMS_DELTA_AGC 4 + +#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \ + (1 << AMS_DELTA_MICROPHONE)) +#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \ + (1 << AMS_DELTA_EARPIECE)) +#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \ + (1 << AMS_DELTA_SPEAKER)) +#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) + +unsigned short ams_delta_audio_mode_pins[] = { + AMS_DELTA_MIXED, + AMS_DELTA_HANDSET, + AMS_DELTA_HANDSFREE, + AMS_DELTA_SPEAKERPHONE, +}; + +static unsigned short ams_delta_audio_agc; + +static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *control = (struct soc_enum *)kcontrol->private_value; + unsigned short pins; + int pin, changed = 0; + + /* Refuse any mode changes if we are not able to control the codec. */ + if (!codec->control_data) + return -EUNATCH; + + if (ucontrol->value.enumerated.item[0] >= control->max) + return -EINVAL; + + mutex_lock(&codec->mutex); + + /* Translate selection to bitmap */ + pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]]; + + /* Setup pins after corresponding bits if changed */ + pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + else + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } + pin = !!(pins & (1 << AMS_DELTA_EARPIECE)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Earpiece"); + else + snd_soc_dapm_disable_pin(codec, "Earpiece"); + } + pin = !!(pins & (1 << AMS_DELTA_MICROPHONE)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Microphone"); + else + snd_soc_dapm_disable_pin(codec, "Microphone"); + } + pin = !!(pins & (1 << AMS_DELTA_SPEAKER)); + if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) { + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + } + pin = !!(pins & (1 << AMS_DELTA_AGC)); + if (pin != ams_delta_audio_agc) { + ams_delta_audio_agc = pin; + changed = 1; + if (pin) + snd_soc_dapm_enable_pin(codec, "AGCIN"); + else + snd_soc_dapm_disable_pin(codec, "AGCIN"); + } + if (changed) + snd_soc_dapm_sync(codec); + + mutex_unlock(&codec->mutex); + + return changed; +} + +static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned short pins, mode; + + pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") << + AMS_DELTA_MOUTHPIECE) | + (snd_soc_dapm_get_pin_status(codec, "Earpiece") << + AMS_DELTA_EARPIECE)); + if (pins) + pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE); + else + pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE) | + (snd_soc_dapm_get_pin_status(codec, "Speaker") << + AMS_DELTA_SPEAKER) | + (ams_delta_audio_agc << AMS_DELTA_AGC)); + + for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++) + if (pins == ams_delta_audio_mode_pins[mode]) + break; + + if (mode >= ARRAY_SIZE(ams_delta_audio_mode)) + return -EINVAL; + + ucontrol->value.enumerated.item[0] = mode; + + return 0; +} + +static const struct soc_enum ams_delta_audio_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode), + ams_delta_audio_mode), +}; + +static const struct snd_kcontrol_new ams_delta_audio_controls[] = { + SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0], + ams_delta_get_audio_mode, ams_delta_set_audio_mode), +}; + +/* Hook switch */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = { + { + .gpio = 4, + .name = "hook_switch", + .report = SND_JACK_HEADSET, + .invert = 1, + .debounce_time = 150, + } +}; + +/* After we are able to control the codec over the modem, + * the hook switch can be used for dynamic DAPM reconfiguration. */ +static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { + /* Handset */ + { + .pin = "Mouthpiece", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Earpiece", + .mask = SND_JACK_HEADPHONE, + }, + /* Handsfree */ + { + .pin = "Microphone", + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, +}; + + +/* + * Modem line discipline, required for making above controls functional. + * Activated from userspace with ldattach, possibly invoked from udev rule. + */ + +/* To actually apply any modem controlled configuration changes to the codec, + * we must connect codec DAI pins to the modem for a moment. Be carefull not + * to interfere with our digital mute function that shares the same hardware. */ +static struct timer_list cx81801_timer; +static bool cx81801_cmd_pending; +static bool ams_delta_muted; +static DEFINE_SPINLOCK(ams_delta_lock); + +static void cx81801_timeout(unsigned long data) +{ + int muted; + + spin_lock(&ams_delta_lock); + cx81801_cmd_pending = 0; + muted = ams_delta_muted; + spin_unlock(&ams_delta_lock); + + /* Reconnect the codec DAI back from the modem to the CPU DAI + * only if digital mute still off */ + if (!muted) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); +} + +/* Line discipline .open() */ +static int cx81801_open(struct tty_struct *tty) +{ + return v253_ops.open(tty); +} + +/* Line discipline .close() */ +static void cx81801_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + del_timer_sync(&cx81801_timer); + + v253_ops.close(tty); + + /* Prevent the hook switch from further changing the DAPM pins */ + INIT_LIST_HEAD(&ams_delta_hook_switch.pins); + + /* Revert back to default audio input/output constellation */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earpiece"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_sync(codec); +} + +/* Line discipline .hangup() */ +static int cx81801_hangup(struct tty_struct *tty) +{ + cx81801_close(tty); + return 0; +} + +/* Line discipline .recieve_buf() */ +static void cx81801_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) +{ + struct snd_soc_codec *codec = tty->disc_data; + const unsigned char *c; + int apply, ret; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Initialize timer used for config pulse generation */ + setup_timer(&cx81801_timer, cx81801_timeout, 0); + + v253_ops.receive_buf(tty, cp, fp, count); + + /* Link hook switch to DAPM pins */ + ret = snd_soc_jack_add_pins(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_pins), + ams_delta_hook_switch_pins); + if (ret) + dev_warn(codec->socdev->card->dev, + "Failed to link hook switch to DAPM pins, " + "will continue with hook switch unlinked.\n"); + + return; + } + + v253_ops.receive_buf(tty, cp, fp, count); + + for (c = &cp[count - 1]; c >= cp; c--) { + if (*c != '\r') + continue; + /* Complete modem response received, apply config to codec */ + + spin_lock_bh(&ams_delta_lock); + mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150)); + apply = !ams_delta_muted && !cx81801_cmd_pending; + cx81801_cmd_pending = 1; + spin_unlock_bh(&ams_delta_lock); + + /* Apply config pulse by connecting the codec to the modem + * if not already done */ + if (apply) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + AMS_DELTA_LATCH2_MODEM_CODEC); + break; + } +} + +/* Line discipline .write_wakeup() */ +static void cx81801_wakeup(struct tty_struct *tty) +{ + v253_ops.write_wakeup(tty); +} + +static struct tty_ldisc_ops cx81801_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx81801", + .owner = THIS_MODULE, + .open = cx81801_open, + .close = cx81801_close, + .hangup = cx81801_hangup, + .receive_buf = cx81801_receive, + .write_wakeup = cx81801_wakeup, +}; + + +/* + * Even if not very usefull, the sound card can still work without any of the + * above functonality activated. You can still control its audio input/output + * constellation and speakerphone gain from userspace by issueing AT commands + * over the modem port. + */ + +static int ams_delta_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set cpu DAI configuration */ + return snd_soc_dai_set_fmt(rtd->dai->cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_ops ams_delta_ops = { + .hw_params = ams_delta_hw_params, +}; + + +/* Board specific codec bias level control */ +static int ams_delta_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + struct snd_soc_codec *codec = card->codec; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + AMS_DELTA_LATCH2_MODEM_NRESET); + break; + case SND_SOC_BIAS_OFF: + if (codec->bias_level != SND_SOC_BIAS_OFF) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, + 0); + } + codec->bias_level = level; + + return 0; +} + +/* Digital mute implemented using modem/CPU multiplexer. + * Shares hardware with codec config pulse generation */ +static bool ams_delta_muted = 1; + +static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute) +{ + int apply; + + if (ams_delta_muted == mute) + return 0; + + spin_lock_bh(&ams_delta_lock); + ams_delta_muted = mute; + apply = !cx81801_cmd_pending; + spin_unlock_bh(&ams_delta_lock); + + if (apply) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0); + return 0; +} + +/* Our codec DAI probably doesn't have its own .ops structure */ +static struct snd_soc_dai_ops ams_delta_dai_ops = { + .digital_mute = ams_delta_digital_mute, +}; + +/* Will be used if the codec ever has its own digital_mute function */ +static int ams_delta_startup(struct snd_pcm_substream *substream) +{ + return ams_delta_digital_mute(NULL, 0); +} + +static void ams_delta_shutdown(struct snd_pcm_substream *substream) +{ + ams_delta_digital_mute(NULL, 1); +} + + +/* + * Card initialization + */ + +static int ams_delta_cx20442_init(struct snd_soc_codec *codec) +{ + struct snd_soc_dai *codec_dai = codec->dai; + struct snd_soc_card *card = codec->socdev->card; + int ret; + /* Codec is ready, now add/activate board specific controls */ + + /* Set up digital mute if not provided by the codec */ + if (!codec_dai->ops) { + codec_dai->ops = &ams_delta_dai_ops; + } else if (!codec_dai->ops->digital_mute) { + codec_dai->ops->digital_mute = ams_delta_digital_mute; + } else { + ams_delta_ops.startup = ams_delta_startup; + ams_delta_ops.shutdown = ams_delta_shutdown; + } + + /* Set codec bias level */ + ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + + /* Add hook switch - can be used to control the codec from userspace + * even if line discipline fails */ + ret = snd_soc_jack_new(card, "hook_switch", + SND_JACK_HEADSET, &ams_delta_hook_switch); + if (ret) + dev_warn(card->dev, + "Failed to allocate resources for hook switch, " + "will continue without one.\n"); + else { + ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + if (ret) + dev_warn(card->dev, + "Failed to set up hook switch GPIO line, " + "will continue with hook switch inactive.\n"); + } + + /* Register optional line discipline for over the modem control */ + ret = tty_register_ldisc(N_V253, &cx81801_ops); + if (ret) { + dev_warn(card->dev, + "Failed to register line discipline, " + "will continue without any controls.\n"); + return 0; + } + + /* Add board specific DAPM widgets and routes */ + ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets, + ARRAY_SIZE(ams_delta_dapm_widgets)); + if (ret) { + dev_warn(card->dev, + "Failed to register DAPM controls, " + "will continue without any.\n"); + return 0; + } + + ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map, + ARRAY_SIZE(ams_delta_audio_map)); + if (ret) { + dev_warn(card->dev, + "Failed to set up DAPM routes, " + "will continue with codec default map.\n"); + return 0; + } + + /* Set up initial pin constellation */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earpiece"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + snd_soc_dapm_sync(codec); + + /* Add virtual switch */ + ret = snd_soc_add_controls(codec, ams_delta_audio_controls, + ARRAY_SIZE(ams_delta_audio_controls)); + if (ret) + dev_warn(card->dev, + "Failed to register audio mode control, " + "will continue without it.\n"); + + return 0; +} + +/* DAI glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ams_delta_dai_link = { + .name = "CX20442", + .stream_name = "CX20442", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &cx20442_dai, + .init = ams_delta_cx20442_init, + .ops = &ams_delta_ops, +}; + +/* Audio card driver */ +static struct snd_soc_card ams_delta_audio_card = { + .name = "AMS_DELTA", + .platform = &omap_soc_platform, + .dai_link = &ams_delta_dai_link, + .num_links = 1, + .set_bias_level = ams_delta_set_bias_level, +}; + +/* Audio subsystem */ +static struct snd_soc_device ams_delta_snd_soc_device = { + .card = &ams_delta_audio_card, + .codec_dev = &cx20442_codec_dev, +}; + +/* Module init/exit */ +static struct platform_device *ams_delta_audio_platform_device; +static struct platform_device *cx20442_platform_device; + +static int __init ams_delta_module_init(void) +{ + int ret; + + if (!(machine_is_ams_delta())) + return -ENODEV; + + ams_delta_audio_platform_device = + platform_device_alloc("soc-audio", -1); + if (!ams_delta_audio_platform_device) + return -ENOMEM; + + platform_set_drvdata(ams_delta_audio_platform_device, + &ams_delta_snd_soc_device); + ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev; + *(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1; + + ret = platform_device_add(ams_delta_audio_platform_device); + if (ret) + goto err; + + /* + * Codec platform device could be registered from elsewhere (board?), + * but I do it here as it makes sense only if used with the card. + */ + cx20442_platform_device = platform_device_register_simple("cx20442", + -1, NULL, 0); + return 0; +err: + platform_device_put(ams_delta_audio_platform_device); + return ret; +} +module_init(ams_delta_module_init); + +static void __exit ams_delta_module_exit(void) +{ + struct snd_soc_codec *codec; + struct tty_struct *tty; + + if (ams_delta_audio_card.codec) { + codec = ams_delta_audio_card.codec; + + if (codec->control_data) { + tty = codec->control_data; + + tty_hangup(tty); + } + } + + if (tty_unregister_ldisc(N_V253) != 0) + dev_warn(&ams_delta_audio_platform_device->dev, + "failed to unregister V253 line discipline\n"); + + snd_soc_jack_free_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios); + + /* Keep modem power on */ + ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + platform_device_unregister(cx20442_platform_device); + platform_device_unregister(ams_delta_audio_platform_device); +} +module_exit(ams_delta_module_exit); + +MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>"); +MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index b60b1dfbc435..0a505938e42b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -22,6 +22,7 @@ */ #include <linux/clk.h> +#include <linux/i2c.h> #include <linux/platform_device.h> #include <sound/core.h> #include <sound/pcm.h> @@ -322,8 +323,6 @@ static struct snd_soc_card snd_soc_n810 = { /* Audio private data */ static struct aic3x_setup_data n810_aic33_setup = { - .i2c_bus = 2, - .i2c_address = 0x18, .gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED, .gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT, }; @@ -337,6 +336,13 @@ static struct snd_soc_device n810_snd_devdata = { static struct platform_device *n810_snd_device; +/* temporary i2c device creation until this can be moved into the machine + * support file. +*/ +static struct i2c_board_info i2c_device[] = { + { I2C_BOARD_INFO("tlv320aic3x", 0x1b), } +}; + static int __init n810_soc_init(void) { int err; @@ -345,6 +351,8 @@ static int __init n810_soc_init(void) if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) return -ENODEV; + i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device)); + n810_snd_device = platform_device_alloc("soc-audio", -1); if (!n810_snd_device) return -ENOMEM; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a5d46a7b196a..3341f49402ca 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -139,27 +139,67 @@ static const unsigned long omap34xx_mcbsp_port[][2] = { static const unsigned long omap34xx_mcbsp_port[][2] = {}; #endif +static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id); + int samples; + + /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ + if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) + samples = snd_pcm_lib_period_bytes(substream) >> 1; + else + samples = 1; + + /* Configure McBSP internal buffer usage */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1); + else + omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1); +} + static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + int bus_id = mcbsp_data->bus_id; int err = 0; - if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) { + if (!cpu_dai->active) + err = omap_mcbsp_request(bus_id); + + if (cpu_is_omap343x()) { + int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id); + int max_period; + /* * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. * Set constraint for minimum buffer size to the same than FIFO * size in order to avoid underruns in playback startup because * HW is keeping the DMA request active until FIFO is filled. */ - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX); - } + if (bus_id == 1) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + 4096, UINT_MAX); - if (!cpu_dai->active) - err = omap_mcbsp_request(mcbsp_data->bus_id); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + max_period = omap_mcbsp_get_max_tx_threshold(bus_id); + else + max_period = omap_mcbsp_get_max_rx_threshold(bus_id); + + max_period++; + max_period <<= 1; + + if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + 32, max_period); + } return err; } @@ -183,21 +223,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); - int err = 0; + int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcbsp_data->active++) - omap_mcbsp_start(mcbsp_data->bus_id); + mcbsp_data->active++; + omap_mcbsp_start(mcbsp_data->bus_id, play, !play); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcbsp_data->active) - omap_mcbsp_stop(mcbsp_data->bus_id); + omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + mcbsp_data->active--; break; default: err = -EINVAL; @@ -215,7 +255,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; - int wlen, channels, wpf; + int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; unsigned long port; unsigned int format; @@ -231,6 +271,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else if (cpu_is_omap343x()) { dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap34xx_mcbsp_port[bus_id][substream->stream]; + omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold = + omap_mcbsp_set_threshold; + /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ + if (omap_mcbsp_get_dma_op_mode(bus_id) == + MCBSP_DMA_MODE_THRESHOLD) + sync_mode = OMAP_DMA_SYNC_FRAME; } else { return -ENODEV; } @@ -238,6 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, substream->stream ? "Audio Capture" : "Audio Playback"; omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; + omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { @@ -321,11 +368,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; regs->spcr1 |= RINTM(3); - regs->rcr2 |= RFIG; - regs->xcr2 |= XFIG; + /* RFIG and XFIG are not defined in 34xx */ + if (!cpu_is_omap34xx()) { + regs->rcr2 |= RFIG; + regs->xcr2 |= XFIG; + } if (cpu_is_omap2430() || cpu_is_omap34xx()) { - regs->xccr = DXENDLY(1) | XDMAEN; - regs->rccr = RFULL_CYCLE | RDMAEN; + regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE; + regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE; } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -462,6 +512,40 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data, return 0; } +static int omap_mcbsp_dai_set_rcvr_src(struct omap_mcbsp_data *mcbsp_data, + int clk_id) +{ + int sel_bit, set = 0; + u16 reg = OMAP2_CONTROL_DEVCONF0; + + if (cpu_class_is_omap1()) + return -EINVAL; /* TODO: Can this be implemented for OMAP1? */ + if (mcbsp_data->bus_id != 0) + return -EINVAL; + + switch (clk_id) { + case OMAP_MCBSP_CLKR_SRC_CLKX: + set = 1; + case OMAP_MCBSP_CLKR_SRC_CLKR: + sel_bit = 3; + break; + case OMAP_MCBSP_FSR_SRC_FSX: + set = 1; + case OMAP_MCBSP_FSR_SRC_FSR: + sel_bit = 4; + break; + default: + return -EINVAL; + } + + if (set) + omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg); + else + omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg); + + return 0; +} + static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) @@ -484,6 +568,13 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; break; + + case OMAP_MCBSP_CLKR_SRC_CLKR: + case OMAP_MCBSP_CLKR_SRC_CLKX: + case OMAP_MCBSP_FSR_SRC_FSR: + case OMAP_MCBSP_FSR_SRC_FSX: + err = omap_mcbsp_dai_set_rcvr_src(mcbsp_data, clk_id); + break; default: err = -ENODEV; } diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index c8147aace813..647d2f981ab0 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -32,6 +32,10 @@ enum omap_mcbsp_clksrg_clk { OMAP_MCBSP_SYSCLK_CLK, /* Internal ICLK */ OMAP_MCBSP_SYSCLK_CLKX_EXT, /* External CLKX pin */ OMAP_MCBSP_SYSCLK_CLKR_EXT, /* External CLKR pin */ + OMAP_MCBSP_CLKR_SRC_CLKR, /* CLKR from CLKR pin */ + OMAP_MCBSP_CLKR_SRC_CLKX, /* CLKR from CLKX pin */ + OMAP_MCBSP_FSR_SRC_FSR, /* FSR from FSR pin */ + OMAP_MCBSP_FSR_SRC_FSX, /* FSR from FSX pin */ }; /* McBSP dividers */ diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 84a1950880eb..5735945788bf 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -59,16 +59,31 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data) struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if (cpu_is_omap1510()) { + if ((cpu_is_omap1510()) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { /* - * OMAP1510 doesn't support DMA chaining so have to restart - * the transfer after all periods are transferred + * OMAP1510 doesn't fully support DMA progress counter + * and there is no software emulation implemented yet, + * so have to maintain our own playback progress counter + * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); + if ((stat == OMAP_DMA_LAST_IRQ) && + (prtd->period_index == runtime->periods - 1)) { + /* we are in sync, do nothing */ + spin_unlock_irqrestore(&prtd->lock, flags); + return; + } if (prtd->period_index >= 0) { - if (++prtd->period_index == runtime->periods) { + if (stat & OMAP_DMA_BLOCK_IRQ) { + /* end of buffer reached, loop back */ + prtd->period_index = 0; + } else if (stat & OMAP_DMA_LAST_IRQ) { + /* update the counter for the last period */ + prtd->period_index = runtime->periods - 1; + } else if (++prtd->period_index >= runtime->periods) { + /* end of buffer missed? loop back */ prtd->period_index = 0; - omap_start_dma(prtd->dma_ch); } } spin_unlock_irqrestore(&prtd->lock, flags); @@ -100,7 +115,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!err && !cpu_is_omap1510()) { + if (!err) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -119,8 +134,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream) if (prtd->dma_data == NULL) return 0; - if (!cpu_is_omap1510()) - omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); + omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); omap_free_dma(prtd->dma_ch); prtd->dma_data = NULL; @@ -148,7 +162,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) */ dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; dma_params.trigger = dma_data->dma_req; - dma_params.sync_mode = OMAP_DMA_SYNC_ELEMENT; + dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dma_params.src_amode = OMAP_DMA_AMODE_POST_INC; dma_params.dst_amode = OMAP_DMA_AMODE_CONSTANT; @@ -174,7 +188,15 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + if ((cpu_is_omap1510()) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | + OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); + else + omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ); + + omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); + omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); return 0; } @@ -183,6 +205,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; struct omap_runtime_data *prtd = runtime->private_data; + struct omap_pcm_dma_data *dma_data = prtd->dma_data; unsigned long flags; int ret = 0; @@ -192,6 +215,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->period_index = 0; + /* Configure McBSP internal buffer usage */ + if (dma_data->set_threshold) + dma_data->set_threshold(substream); + omap_start_dma(prtd->dma_ch); break; @@ -288,7 +315,7 @@ static struct snd_pcm_ops omap_pcm_ops = { .mmap = omap_pcm_mmap, }; -static u64 omap_pcm_dmamask = DMA_BIT_MASK(32); +static u64 omap_pcm_dmamask = DMA_BIT_MASK(64); static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) @@ -330,7 +357,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, +static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -338,7 +365,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &omap_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + card->dev->coherent_dma_mask = DMA_BIT_MASK(64); if (dai->playback.channels_min) { ret = omap_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 8d9d26916b05..38a821dd4118 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,6 +29,8 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ + int sync_mode; /* DMA sync mode */ + void (*set_threshold)(struct snd_pcm_substream *substream); }; extern struct snd_soc_platform omap_soc_platform; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index b719e5db4f57..4a3f62d1f295 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,6 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -39,6 +40,11 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) + static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -96,7 +102,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFM); + SND_SOC_DAIFMT_CBM_CFM); if (ret) { printk(KERN_ERR "can't set codec DAI configuration\n"); return ret; @@ -280,6 +286,7 @@ static struct snd_soc_card snd_soc_sdp3430 = { static struct twl4030_setup_data twl4030_setup = { .ramp_delay_value = 3, .sysclk = 26000, + .hs_extmute = 1, }; /* Audio subsystem */ @@ -294,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -312,6 +320,14 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + /* Set TWL4030 GPIO6 as EXTMUTE signal */ + twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); + ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c new file mode 100644 index 000000000000..f90b45f56220 --- /dev/null +++ b/sound/soc/omap/zoom2.c @@ -0,0 +1,314 @@ +/* + * zoom2.c -- SoC audio for Zoom2 + * + * Author: Misael Lopez Cruz <x0052729@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <mach/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define ZOOM2_HEADSET_MUX_GPIO (OMAP_MAX_GPIO_LINES + 15) +#define ZOOM2_HEADSET_EXTMUTE_GPIO 153 + +static int zoom2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_ops = { + .hw_params = zoom2_hw_params, +}; + +static int zoom2_hw_voice_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops zoom2_voice_ops = { + .hw_params = zoom2_hw_voice_params, +}; + +/* Zoom2 machine DAPM */ +static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Aux In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Mic Bias 1"}, + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 1", NULL, "Ext Mic"}, + {"Mic Bias 2", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Stereophone: HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Aux In: AUXL, AUXR */ + {"Aux In", NULL, "AUXL"}, + {"Aux In", NULL, "AUXR"}, +}; + +static int zoom2_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add Zoom2 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets, + ARRAY_SIZE(zoom2_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up Zoom2 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Zoom2 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + snd_soc_dapm_enable_pin(codec, "Aux In"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); + + snd_soc_dapm_nc_pin(codec, "OUTL"); + snd_soc_dapm_nc_pin(codec, "OUTR"); + snd_soc_dapm_nc_pin(codec, "EARPIECE"); + snd_soc_dapm_nc_pin(codec, "PREDRIVEL"); + snd_soc_dapm_nc_pin(codec, "PREDRIVER"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Enable voice interface */ + reg = codec->read(codec, TWL4030_REG_VOICE_IF); + reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN; + codec->write(codec, TWL4030_REG_VOICE_IF, reg); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link zoom2_dai[] = { + { + .name = "TWL4030 I2S", + .stream_name = "TWL4030 Audio", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .init = zoom2_twl4030_init, + .ops = &zoom2_ops, + }, + { + .name = "TWL4030 PCM", + .stream_name = "TWL4030 Voice", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai[TWL4030_DAI_VOICE], + .init = zoom2_twl4030_voice_init, + .ops = &zoom2_voice_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_zoom2 = { + .name = "Zoom2", + .platform = &omap_soc_platform, + .dai_link = zoom2_dai, + .num_links = ARRAY_SIZE(zoom2_dai), +}; + +/* EXTMUTE callback function */ +void zoom2_set_hs_extmute(int mute) +{ + gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute); +} + +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 3, /* 161 ms */ + .sysclk = 26000, + .hs_extmute = 1, + .set_hs_extmute = zoom2_set_hs_extmute, +}; + +/* Audio subsystem */ +static struct snd_soc_device zoom2_snd_devdata = { + .card = &snd_soc_zoom2, + .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, +}; + +static struct platform_device *zoom2_snd_device; + +static int __init zoom2_soc_init(void) +{ + int ret; + + if (!machine_is_omap_zoom2()) { + pr_debug("Not Zoom2!\n"); + return -ENODEV; + } + printk(KERN_INFO "Zoom2 SoC init\n"); + + zoom2_snd_device = platform_device_alloc("soc-audio", -1); + if (!zoom2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata); + zoom2_snd_devdata.dev = &zoom2_snd_device->dev; + *(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ + *(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + + ret = platform_device_add(zoom2_snd_device); + if (ret) + goto err1; + + BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); + gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); + + BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0); + gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0); + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(zoom2_snd_device); + + return ret; +} +module_init(zoom2_soc_init); + +static void __exit zoom2_soc_exit(void) +{ + gpio_free(ZOOM2_HEADSET_MUX_GPIO); + gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO); + + platform_device_unregister(zoom2_snd_device); +} +module_exit(zoom2_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>"); +MODULE_DESCRIPTION("ALSA SoC Zoom2"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 326955dea36c..9f7c61e23daf 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -20,12 +20,14 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/gpio.h> +#include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/uda1380.h> #include <mach/magician.h> #include <asm/mach-types.h> @@ -188,7 +190,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -447,34 +449,47 @@ static struct snd_soc_card snd_soc_card_magician = { .platform = &pxa2xx_soc_platform, }; -/* magician audio private data */ -static struct uda1380_setup_data magician_uda1380_setup = { - .i2c_address = 0x18, - .dac_clk = UDA1380_DAC_CLK_WSPLL, -}; - /* magician audio subsystem */ static struct snd_soc_device magician_snd_devdata = { .card = &snd_soc_card_magician, .codec_dev = &soc_codec_dev_uda1380, - .codec_data = &magician_uda1380_setup, }; static struct platform_device *magician_snd_device; +/* + * FIXME: move into magician board file once merged into the pxa tree + */ +static struct uda1380_platform_data uda1380_info = { + .gpio_power = EGPIO_MAGICIAN_CODEC_POWER, + .gpio_reset = EGPIO_MAGICIAN_CODEC_RESET, + .dac_clk = UDA1380_DAC_CLK_WSPLL, +}; + +static struct i2c_board_info i2c_board_info[] = { + { + I2C_BOARD_INFO("uda1380", 0x18), + .platform_data = &uda1380_info, + }, +}; + static int __init magician_init(void) { int ret; + struct i2c_adapter *adapter; + struct i2c_client *client; if (!machine_is_magician()) return -ENODEV; - ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER"); - if (ret) - goto err_request_power; - ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET"); - if (ret) - goto err_request_reset; + adapter = i2c_get_adapter(0); + if (!adapter) + return -ENODEV; + client = i2c_new_device(adapter, i2c_board_info); + i2c_put_adapter(adapter); + if (!client) + return -ENODEV; + ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER"); if (ret) goto err_request_spk; @@ -491,14 +506,8 @@ static int __init magician_init(void) if (ret) goto err_request_in_sel1; - gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1); gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0); - /* we may need to have the clock running here - pH5 */ - gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1); - udelay(5); - gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0); - magician_snd_device = platform_device_alloc("soc-audio", -1); if (!magician_snd_device) { ret = -ENOMEM; @@ -526,10 +535,6 @@ err_request_mic: err_request_ep: gpio_free(EGPIO_MAGICIAN_SPK_POWER); err_request_spk: - gpio_free(EGPIO_MAGICIAN_CODEC_RESET); -err_request_reset: - gpio_free(EGPIO_MAGICIAN_CODEC_POWER); -err_request_power: return ret; } @@ -540,15 +545,12 @@ static void __exit magician_exit(void) gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0); gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0); gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0); - gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0); gpio_free(EGPIO_MAGICIAN_IN_SEL1); gpio_free(EGPIO_MAGICIAN_IN_SEL0); gpio_free(EGPIO_MAGICIAN_MIC_POWER); gpio_free(EGPIO_MAGICIAN_EP_POWER); gpio_free(EGPIO_MAGICIAN_SPK_POWER); - gpio_free(EGPIO_MAGICIAN_CODEC_RESET); - gpio_free(EGPIO_MAGICIAN_CODEC_POWER); } module_init(magician_init); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index e6102fda0a7f..1f96e3227be5 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -17,13 +17,12 @@ #include <linux/moduleparam.h> #include <linux/device.h> #include <linux/gpio.h> -#include <linux/interrupt.h> -#include <linux/irq.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/audio.h> @@ -33,90 +32,31 @@ #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static int palm27x_jack_func = 1; -static int palm27x_spk_func = 1; -static int palm27x_ep_gpio = -1; +static struct snd_soc_jack hs_jack; -static void palm27x_ext_control(struct snd_soc_codec *codec) -{ - if (!palm27x_spk_func) - snd_soc_dapm_enable_pin(codec, "Speaker"); - else - snd_soc_dapm_disable_pin(codec, "Speaker"); - - if (!palm27x_jack_func) - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - else - snd_soc_dapm_disable_pin(codec, "Headphone Jack"); - - snd_soc_dapm_sync(codec); -} - -static int palm27x_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->card->codec; - - /* check the jack status at stream startup */ - palm27x_ext_control(codec); - return 0; -} - -static struct snd_soc_ops palm27x_ops = { - .startup = palm27x_startup, +/* Headphones jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, }; -static irqreturn_t palm27x_interrupt(int irq, void *v) -{ - palm27x_spk_func = gpio_get_value(palm27x_ep_gpio); - palm27x_jack_func = !palm27x_spk_func; - return IRQ_HANDLED; -} - -static int palm27x_get_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = palm27x_jack_func; - return 0; -} - -static int palm27x_set_jack(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - if (palm27x_jack_func == ucontrol->value.integer.value[0]) - return 0; - - palm27x_jack_func = ucontrol->value.integer.value[0]; - palm27x_ext_control(codec); - return 1; -} - -static int palm27x_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = palm27x_spk_func; - return 0; -} - -static int palm27x_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - if (palm27x_spk_func == ucontrol->value.integer.value[0]) - return 0; - - palm27x_spk_func = ucontrol->value.integer.value[0]; - palm27x_ext_control(codec); - return 1; -} +/* Headphones jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + [0] = { + /* gpio is set on per-platform basis */ + .name = "hp-gpio", + .report = SND_JACK_HEADPHONE, + .debounce_time = 200, + }, +}; -/* PalmTX machine dapm widgets */ +/* Palm27x machine dapm widgets */ static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext. Speaker", NULL), + SND_SOC_DAPM_MIC("Ext. Microphone", NULL), }; /* PalmTX audio map */ @@ -126,46 +66,66 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Headphone Jack", NULL, "HPOUTR"}, /* ext speaker connected to ROUT2, LOUT2 */ - {"Speaker", NULL, "LOUT2"}, - {"Speaker", NULL, "ROUT2"}, -}; + {"Ext. Speaker", NULL, "LOUT2"}, + {"Ext. Speaker", NULL, "ROUT2"}, -static const char *jack_function[] = {"Headphone", "Off"}; -static const char *spk_function[] = {"On", "Off"}; -static const struct soc_enum palm27x_enum[] = { - SOC_ENUM_SINGLE_EXT(2, jack_function), - SOC_ENUM_SINGLE_EXT(2, spk_function), + /* mic connected to MIC1 */ + {"Ext. Microphone", NULL, "MIC1"}, }; -static const struct snd_kcontrol_new palm27x_controls[] = { - SOC_ENUM_EXT("Jack Function", palm27x_enum[0], palm27x_get_jack, - palm27x_set_jack), - SOC_ENUM_EXT("Speaker Function", palm27x_enum[1], palm27x_get_spk, - palm27x_set_spk), -}; +static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_codec *codec) { int err; + /* add palm27x specific widgets */ + err = snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, + ARRAY_SIZE(palm27x_dapm_widgets)); + if (err) + return err; + + /* set up palm27x specific audio path audio_map */ + err = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + if (err) + return err; + + /* connected pins */ + if (machine_is_palmld()) + snd_soc_dapm_enable_pin(codec, "MIC1"); + snd_soc_dapm_enable_pin(codec, "HPOUTL"); + snd_soc_dapm_enable_pin(codec, "HPOUTR"); + snd_soc_dapm_enable_pin(codec, "LOUT2"); + snd_soc_dapm_enable_pin(codec, "ROUT2"); + + /* not connected pins */ snd_soc_dapm_nc_pin(codec, "OUT3"); snd_soc_dapm_nc_pin(codec, "MONOOUT"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + err = snd_soc_dapm_sync(codec); + if (err) + return err; - /* add palm27x specific controls */ - err = snd_soc_add_controls(codec, palm27x_controls, - ARRAY_SIZE(palm27x_controls)); - if (err < 0) + /* Jack detection API stuff */ + err = snd_soc_jack_new(&palm27x_asoc, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack); + if (err) return err; - /* add palm27x specific widgets */ - snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, - ARRAY_SIZE(palm27x_dapm_widgets)); + err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (err) + return err; - /* set up palm27x specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + err = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); - snd_soc_dapm_sync(codec); - return 0; + return err; } static struct snd_soc_dai_link palm27x_dai[] = { @@ -175,14 +135,12 @@ static struct snd_soc_dai_link palm27x_dai[] = { .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], .init = palm27x_ac97_init, - .ops = &palm27x_ops, }, { .name = "AC97 Aux", .stream_name = "AC97 Aux", .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], - .ops = &palm27x_ops, }, }; @@ -208,27 +166,17 @@ static int palm27x_asoc_probe(struct platform_device *pdev) machine_is_palmld() || machine_is_palmte2())) return -ENODEV; - if (pdev->dev.platform_data) - palm27x_ep_gpio = ((struct palm27x_asoc_info *) - (pdev->dev.platform_data))->jack_gpio; - - ret = gpio_request(palm27x_ep_gpio, "Headphone Jack"); - if (ret) - return ret; - ret = gpio_direction_input(palm27x_ep_gpio); - if (ret) - goto err_alloc; + if (!pdev->dev.platform_data) { + dev_err(&pdev->dev, "please supply platform_data\n"); + return -ENODEV; + } - if (request_irq(gpio_to_irq(palm27x_ep_gpio), palm27x_interrupt, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - "Headphone jack", NULL)) - goto err_alloc; + hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *) + (pdev->dev.platform_data))->jack_gpio; palm27x_snd_device = platform_device_alloc("soc-audio", -1); - if (!palm27x_snd_device) { - ret = -ENOMEM; - goto err_dev; - } + if (!palm27x_snd_device) + return -ENOMEM; platform_set_drvdata(palm27x_snd_device, &palm27x_snd_devdata); palm27x_snd_devdata.dev = &palm27x_snd_device->dev; @@ -241,18 +189,12 @@ static int palm27x_asoc_probe(struct platform_device *pdev) put_device: platform_device_put(palm27x_snd_device); -err_dev: - free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); -err_alloc: - gpio_free(palm27x_ep_gpio); return ret; } static int __devexit palm27x_asoc_remove(struct platform_device *pdev) { - free_irq(gpio_to_irq(palm27x_ep_gpio), NULL); - gpio_free(palm27x_ep_gpio); platform_device_unregister(palm27x_snd_device); return 0; } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 19c45409d94c..5b9ed6464789 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); + + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } @@ -457,31 +470,27 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; case SND_SOC_DAIFMT_DSP_A: @@ -489,22 +498,6 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - sspsp |= SSPSP_SFRMP; - break; - case SND_SOC_DAIFMT_NB_IF: - break; - case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(2); - break; - case SND_SOC_DAIFMT_IB_NF: - sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; - break; - default: - return -EINVAL; - } break; default: diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index d9c94d71fa61..e9ae7b3a7e00 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -22,6 +22,7 @@ #include <mach/hardware.h> #include <mach/regs-ac97.h> #include <mach/dma.h> +#include <mach/audio.h> #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" @@ -241,9 +242,18 @@ EXPORT_SYMBOL_GPL(soc_ac97_ops); static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int i; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; - for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) + if (pdev->id >= 0) { + dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); + return -ENXIO; + } + + for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) { pxa_ac97_dai[i].dev = &pdev->dev; + if (pdata && pdata->codec_pdata[0]) + pxa_ac97_dai[i].ac97_pdata = pdata->codec_pdata[0]; + } /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index df494d1e346f..923428fc1adb 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,7 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 + depends on ARCH_S3C2410 || ARCH_S3C64XX + select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97 or I2S interfaces. You will also need to @@ -38,6 +39,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753 Say Y if you want to add support for SoC audio on smdk2440 with the WM8753. +config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 + tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + This driver provides audio support for the Openmoko Neo FreeRunner + smartphone. + config SND_S3C24XX_SOC_JIVE_WM8750 tristate "SoC I2S Audio support for Jive" depends on SND_S3C24XX_SOC && MACH_JIVE @@ -57,7 +67,7 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710 config SND_S3C24XX_SOC_LN2440SBC_ALC650 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on SND_S3C24XX_SOC + depends on SND_S3C24XX_SOC && ARCH_S3C2410 select SND_S3C2443_SOC_AC97 select SND_SOC_AC97_CODEC help @@ -66,7 +76,26 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650 config SND_S3C24XX_SOC_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_S3C24XX_SOC + depends on SND_S3C24XX_SOC && ARCH_S3C2410 select SND_S3C24XX_SOC_I2S select SND_SOC_L3 select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 07a93a2ebe5f..99f5a7dd3fc6 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -16,12 +16,21 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o +snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o + diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c new file mode 100644 index 000000000000..0c52e36ddd87 --- /dev/null +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -0,0 +1,498 @@ +/* + * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) + * + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory <graeme@openmoko.org> + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory <linux@wolfsonmicro.com> + * Copyright 2009 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include <plat/regs-iis.h> + +#include <mach/regs-clock.h> +#include <asm/io.h> +#include <mach/gta02.h> +#include "../codecs/wm8753.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card neo1973_gta02; + +static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, + WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_gta02_hifi_ops = { + .hw_params = neo1973_gta02_hifi_hw_params, + .hw_free = neo1973_gta02_hifi_hw_free, +}; + +static int neo1973_gta02_voice_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, + 12288000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, + pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); +} + +static struct snd_soc_ops neo1973_gta02_voice_ops = { + .hw_params = neo1973_gta02_voice_hw_params, + .hw_free = neo1973_gta02_voice_hw_free, +}; + +#define LM4853_AMP 1 +#define LM4853_SPK 2 + +static u8 lm4853_state; + +/* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ +static int lm4853_set_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) + lm4853_state |= LM4853_AMP; + else + lm4853_state &= ~LM4853_AMP; + + return 0; +} + +static int lm4853_get_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; + + return 0; +} + +static int lm4853_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) { + lm4853_state |= LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 0); + } else { + lm4853_state &= ~LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 1); + } + + return 0; +} + +static int lm4853_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; + + return 0; +} + +static int lm4853_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), +}; + + +/* example machine audio_mapnections */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Connections to the lm4853 amp */ + {"Stereo Out", NULL, "LOUT1"}, + {"Stereo Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Call Speaker */ + {"Handset Spk", NULL, "LOUT2"}, + {"Handset Spk", NULL, "ROUT2"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { + SOC_DAPM_PIN_SWITCH("Stereo Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Spk"), + + /* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ + SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, + lm4853_get_state, + lm4853_set_state), + SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, + lm4853_get_spk, + lm4853_set_spk), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 GTA02. + */ +static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); + + /* Add neo1973 gta02 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* add neo1973 gta02 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, + ARRAY_SIZE(wm8753_neo1973_gta02_controls)); + + if (err < 0) + return err; + + /* set up neo1973 gta02 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to default off mode */ + snd_soc_dapm_disable_pin(codec, "Stereo Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_gta02_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_gta02_wm8753_init, + .ops = &neo1973_gta02_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_gta02_voice_ops, +}, +}; + +static struct snd_soc_card neo1973_gta02 = { + .name = "neo1973-gta02", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_gta02_dai, + .num_links = ARRAY_SIZE(neo1973_gta02_dai), +}; + +static struct snd_soc_device neo1973_gta02_snd_devdata = { + .card = &neo1973_gta02, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *neo1973_gta02_snd_device; + +static int __init neo1973_gta02_init(void) +{ + int ret; + + if (!machine_is_neo1973_gta02()) { + printk(KERN_INFO + "Only GTA02 is supported by this ASoC driver\n"); + return -ENODEV; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&bt_dai); + if (ret) + return ret; + + neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_gta02_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_gta02_snd_device, + &neo1973_gta02_snd_devdata); + neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; + ret = platform_device_add(neo1973_gta02_snd_device); + + if (ret) { + platform_device_put(neo1973_gta02_snd_device); + return ret; + } + + /* Initialise GPIOs used by amp */ + ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_unregister_device; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_free_gpio_hp_in; + } + + ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_hp_in; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(GTA02_GPIO_AMP_SHUT); +err_free_gpio_hp_in: + gpio_free(GTA02_GPIO_HP_IN); +err_unregister_device: + platform_device_unregister(neo1973_gta02_snd_device); + return ret; +} +module_init(neo1973_gta02_init); + +static void __exit neo1973_gta02_exit(void) +{ + snd_soc_unregister_dai(&bt_dai); + platform_device_unregister(neo1973_gta02_snd_device); + gpio_free(GTA02_GPIO_HP_IN); + gpio_free(GTA02_GPIO_AMP_SHUT); +} +module_exit(neo1973_gta02_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 1a283170ca92..aa7af0b8d421 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -36,6 +36,7 @@ #include <mach/dma.h> #include "s3c-i2s-v2.h" +#include "s3c24xx-pcm.h" #undef S3C_IIS_V2_SUPPORTED @@ -357,19 +358,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, #endif #ifdef CONFIG_PLAT_S3C64XX - iismod &= ~0x606; + iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK); /* Sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: /* 8 bit sample, 16fs BCLK */ - iismod |= 0x2004; + iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS); break; case SNDRV_PCM_FORMAT_S16_LE: /* 16 bit sample, 32fs BCLK */ break; case SNDRV_PCM_FORMAT_S24_LE: /* 24 bit sample, 48fs BCLK */ - iismod |= 0x4002; + iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS); break; } #endif @@ -387,6 +388,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); @@ -416,6 +419,14 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, s3c2412_snd_txctrl(i2s, 1); local_irq_restore(irqs); + + /* + * Load the next buffer to DMA to meet the reqirement + * of the auto reload mechanism of S3C24XX. + * This call won't bother S3C64XX. + */ + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + break; case SNDRV_PCM_TRIGGER_STOP: diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 3f03d5ddfacd..fc1beb0930b9 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97; static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; -static DECLARE_MUTEX(ac97_mutex); +static DEFINE_MUTEX(ac97_mutex); static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, u32 ac_codec_cmd; u32 stat, addr, data; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," " rep addr = %02x\n", reg, addr); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); return (unsigned short)data; } @@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, u32 ac_glbctrl; u32 ac_codec_cmd; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); } @@ -290,6 +290,9 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); switch (cmd) { @@ -312,6 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, } writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + return 0; } @@ -334,6 +339,9 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); switch (cmd) { @@ -349,6 +357,8 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, } writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + return 0; } diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 556e35f0ab73..40e2c4790f0d 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -279,6 +279,9 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int channel = ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); @@ -296,6 +299,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, s3c24xx_snd_rxctrl(1); else s3c24xx_snd_txctrl(1); + + s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index eecfa5eba06b..5cbbdc80fde3 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -255,7 +255,6 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: @@ -318,6 +317,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c new file mode 100644 index 000000000000..1966e0d5652d --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -0,0 +1,394 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +/* This is added as a manual control as the speaker amps create clicks + * when their power state is changed, which are far more noticeable than + * anything produced by the CODEC itself. + */ +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, curently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} +EXPORT_SYMBOL_GPL(simtec_audio_core_probe); + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h new file mode 100644 index 000000000000..2714203af161 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c new file mode 100644 index 000000000000..8346bd96eaf5 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -0,0 +1,153 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic Jack" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are + * not using the DAPM to power it up and down as there it makes + * a click when powering up. */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c new file mode 100644 index 000000000000..25797e096175 --- /dev/null +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index b5f95f9781c1..c1b40ac22c05 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -14,6 +14,7 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -189,8 +190,6 @@ static struct snd_soc_card snd_soc_card_s6105 = { /* s6105 audio private data */ static struct aic3x_setup_data s6105_aic3x_setup = { - .i2c_bus = 0, - .i2c_address = 0x18, }; /* s6105 audio subsystem */ @@ -211,10 +210,19 @@ static struct s6000_snd_platform_data __initdata s6105_snd_data = { static struct platform_device *s6105_snd_device; +/* temporary i2c device creation until this can be moved into the machine + * support file. +*/ +static struct i2c_board_info i2c_device[] = { + { I2C_BOARD_INFO("tlv320aic33", 0x18), } +}; + static int __init s6105_init(void) { int ret; + i2c_register_board_info(0, i2c_device, ARRAY_SIZE(i2c_device)); + s6105_snd_device = platform_device_alloc("soc-audio", -1); if (!s6105_snd_device) return -ENOMEM; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 54bd604012af..9154b4363db3 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -20,7 +20,12 @@ config SND_SOC_SH4_HAC config SND_SOC_SH4_SSI tristate - +config SND_SOC_SH4_FSI + tristate "SH4 FSI support" + depends on CPU_SUBTYPE_SH7724 + select SH_DMA + help + This option enables FSI sound support ## ## Boards @@ -35,4 +40,12 @@ config SND_SH7760_AC97 This option enables generic sound support for the first AC97 unit of the SH7760. +config SND_FSI_AK4642 + bool "FSI-AK4642 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_AK4642 + help + This option enables generic sound support for the + FSI - AK4642 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a8e8ab81cc6a..a6997872f24e 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -5,10 +5,14 @@ obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o ## audio units found on some SH-4 snd-soc-hac-objs := hac.o snd-soc-ssi-objs := ssi.o +snd-soc-fsi-objs := fsi.o obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o +obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o +snd-soc-fsi-ak4642-objs := fsi-ak4642.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o +obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c new file mode 100644 index 000000000000..c7af09729c6e --- /dev/null +++ b/sound/soc/sh/fsi-ak4642.c @@ -0,0 +1,107 @@ +/* + * FSI-AK464x sound support for ms7724se + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori@renesas.com> + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file "COPYING" in the main directory of this archive + * for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include <../sound/soc/codecs/ak4642.h> + +static struct snd_soc_dai_link fsi_dai_link = { + .name = "AK4642", + .stream_name = "AK4642", + .cpu_dai = &fsi_soc_dai[0], /* fsi */ + .codec_dai = &ak4642_dai, + .ops = NULL, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device fsi_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_ak4642, +}; + +#define AK4642_BUS 0 +#define AK4642_ADR 0x12 +static int ak4642_add_i2c_device(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = AK4642_ADR; + strlcpy(info.type, "ak4642", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(AK4642_BUS); + if (!adapter) { + printk(KERN_DEBUG "can't get i2c adapter\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_DEBUG "can't add i2c device\n"); + return -ENODEV; + } + + return 0; +} + +static struct platform_device *fsi_snd_device; + +static int __init fsi_ak4642_init(void) +{ + int ret = -ENOMEM; + + ak4642_add_i2c_device(); + + fsi_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_snd_device) + goto out; + + platform_set_drvdata(fsi_snd_device, + &fsi_snd_devdata); + fsi_snd_devdata.dev = &fsi_snd_device->dev; + ret = platform_device_add(fsi_snd_device); + + if (ret) + platform_device_put(fsi_snd_device); + +out: + return ret; +} + +static void __exit fsi_ak4642_exit(void) +{ + platform_device_unregister(fsi_snd_device); +} + +module_init(fsi_ak4642_init); +module_exit(fsi_ak4642_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH4 FSI-AK4642 sound card"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c new file mode 100644 index 000000000000..44123248b630 --- /dev/null +++ b/sound/soc/sh/fsi.c @@ -0,0 +1,1004 @@ +/* + * Fifo-attached Serial Interface (FSI) support for SH7724 + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori@renesas.com> + * + * Based on ssi.c + * Copyright (c) 2007 Manuel Lauss <mano@roarinelk.homelinux.net> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/list.h> +#include <linux/clk.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/sh_fsi.h> +#include <asm/atomic.h> +#include <asm/dma.h> +#include <asm/dma-sh.h> + +#define DO_FMT 0x0000 +#define DOFF_CTL 0x0004 +#define DOFF_ST 0x0008 +#define DI_FMT 0x000C +#define DIFF_CTL 0x0010 +#define DIFF_ST 0x0014 +#define CKG1 0x0018 +#define CKG2 0x001C +#define DIDT 0x0020 +#define DODT 0x0024 +#define MUTE_ST 0x0028 +#define REG_END MUTE_ST + +#define INT_ST 0x0200 +#define IEMSK 0x0204 +#define IMSK 0x0208 +#define MUTE 0x020C +#define CLK_RST 0x0210 +#define SOFT_RST 0x0214 +#define MREG_START INT_ST +#define MREG_END SOFT_RST + +/* DO_FMT */ +/* DI_FMT */ +#define CR_FMT(param) ((param) << 4) +# define CR_MONO 0x0 +# define CR_MONO_D 0x1 +# define CR_PCM 0x2 +# define CR_I2S 0x3 +# define CR_TDM 0x4 +# define CR_TDM_D 0x5 + +/* DOFF_CTL */ +/* DIFF_CTL */ +#define IRQ_HALF 0x00100000 +#define FIFO_CLR 0x00000001 + +/* DOFF_ST */ +#define ERR_OVER 0x00000010 +#define ERR_UNDER 0x00000001 + +/* CLK_RST */ +#define B_CLK 0x00000010 +#define A_CLK 0x00000001 + +/* INT_ST */ +#define INT_B_IN (1 << 12) +#define INT_B_OUT (1 << 8) +#define INT_A_IN (1 << 4) +#define INT_A_OUT (1 << 0) + +#define FSI_RATES SNDRV_PCM_RATE_8000_96000 + +#define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) + +/************************************************************************ + + + struct + + +************************************************************************/ +struct fsi_priv { + void __iomem *base; + struct snd_pcm_substream *substream; + + int fifo_max; + int chan; + int dma_chan; + + int byte_offset; + int period_len; + int buffer_len; + int periods; +}; + +struct fsi_master { + void __iomem *base; + int irq; + struct clk *clk; + struct fsi_priv fsia; + struct fsi_priv fsib; + struct sh_fsi_platform_info *info; +}; + +static struct fsi_master *master; + +/************************************************************************ + + + basic read write function + + +************************************************************************/ +static int __fsi_reg_write(u32 reg, u32 data) +{ + /* valid data area is 24bit */ + data &= 0x00ffffff; + + return ctrl_outl(data, reg); +} + +static u32 __fsi_reg_read(u32 reg) +{ + return ctrl_inl(reg); +} + +static int __fsi_reg_mask_set(u32 reg, u32 mask, u32 data) +{ + u32 val = __fsi_reg_read(reg); + + val &= ~mask; + val |= data & mask; + + return __fsi_reg_write(reg, val); +} + +static int fsi_reg_write(struct fsi_priv *fsi, u32 reg, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_write((u32)(fsi->base + reg), data); +} + +static u32 fsi_reg_read(struct fsi_priv *fsi, u32 reg) +{ + if (reg > REG_END) + return 0; + + return __fsi_reg_read((u32)(fsi->base + reg)); +} + +static int fsi_reg_mask_set(struct fsi_priv *fsi, u32 reg, u32 mask, u32 data) +{ + if (reg > REG_END) + return -1; + + return __fsi_reg_mask_set((u32)(fsi->base + reg), mask, data); +} + +static int fsi_master_write(u32 reg, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_write((u32)(master->base + reg), data); +} + +static u32 fsi_master_read(u32 reg) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return 0; + + return __fsi_reg_read((u32)(master->base + reg)); +} + +static int fsi_master_mask_set(u32 reg, u32 mask, u32 data) +{ + if ((reg < MREG_START) || + (reg > MREG_END)) + return -1; + + return __fsi_reg_mask_set((u32)(master->base + reg), mask, data); +} + +/************************************************************************ + + + basic function + + +************************************************************************/ +static struct fsi_priv *fsi_get(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd; + struct fsi_priv *fsi = NULL; + + if (!substream || !master) + return NULL; + + rtd = substream->private_data; + switch (rtd->dai->cpu_dai->id) { + case 0: + fsi = &master->fsia; + break; + case 1: + fsi = &master->fsib; + break; + } + + return fsi; +} + +static int fsi_is_port_a(struct fsi_priv *fsi) +{ + /* return + * 1 : port a + * 0 : port b + */ + + if (fsi == &master->fsia) + return 1; + + return 0; +} + +static u32 fsi_get_info_flags(struct fsi_priv *fsi) +{ + int is_porta = fsi_is_port_a(fsi); + + return is_porta ? master->info->porta_flags : + master->info->portb_flags; +} + +static int fsi_is_master_mode(struct fsi_priv *fsi, int is_play) +{ + u32 mode; + u32 flags = fsi_get_info_flags(fsi); + + mode = is_play ? SH_FSI_OUT_SLAVE_MODE : SH_FSI_IN_SLAVE_MODE; + + /* return + * 1 : master mode + * 0 : slave mode + */ + + return (mode & flags) != mode; +} + +static u32 fsi_port_ab_io_bit(struct fsi_priv *fsi, int is_play) +{ + int is_porta = fsi_is_port_a(fsi); + u32 data; + + if (is_porta) + data = is_play ? (1 << 0) : (1 << 4); + else + data = is_play ? (1 << 8) : (1 << 12); + + return data; +} + +static void fsi_stream_push(struct fsi_priv *fsi, + struct snd_pcm_substream *substream, + u32 buffer_len, + u32 period_len) +{ + fsi->substream = substream; + fsi->buffer_len = buffer_len; + fsi->period_len = period_len; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static void fsi_stream_pop(struct fsi_priv *fsi) +{ + fsi->substream = NULL; + fsi->buffer_len = 0; + fsi->period_len = 0; + fsi->byte_offset = 0; + fsi->periods = 0; +} + +static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) +{ + u32 status; + u32 reg = is_play ? DOFF_ST : DIFF_ST; + int residue; + + status = fsi_reg_read(fsi, reg); + residue = 0x1ff & (status >> 8); + residue *= fsi->chan; + + return residue; +} + +static int fsi_get_residue(struct fsi_priv *fsi, int is_play) +{ + int residue; + int width; + struct snd_pcm_runtime *runtime; + + runtime = fsi->substream->runtime; + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + if (2 == width) + residue = fsi_get_fifo_residue(fsi, is_play); + else + residue = get_dma_residue(fsi->dma_chan); + + return residue; +} + +/************************************************************************ + + + basic dma function + + +************************************************************************/ +#define PORTA_DMA 0 +#define PORTB_DMA 1 + +static int fsi_get_dma_chan(void) +{ + if (0 != request_dma(PORTA_DMA, "fsia")) + return -EIO; + + if (0 != request_dma(PORTB_DMA, "fsib")) { + free_dma(PORTA_DMA); + return -EIO; + } + + master->fsia.dma_chan = PORTA_DMA; + master->fsib.dma_chan = PORTB_DMA; + + return 0; +} + +static void fsi_free_dma_chan(void) +{ + dma_wait_for_completion(PORTA_DMA); + dma_wait_for_completion(PORTB_DMA); + free_dma(PORTA_DMA); + free_dma(PORTB_DMA); + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; +} + +/************************************************************************ + + + ctrl function + + +************************************************************************/ +static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, data); + fsi_master_mask_set(IEMSK, data, data); +} + +static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) +{ + u32 data = fsi_port_ab_io_bit(fsi, is_play); + + fsi_master_mask_set(IMSK, data, 0); + fsi_master_mask_set(IEMSK, data, 0); +} + +static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) +{ + u32 val = fsi_is_port_a(fsi) ? (1 << 0) : (1 << 4); + + if (enable) + fsi_master_mask_set(CLK_RST, val, val); + else + fsi_master_mask_set(CLK_RST, val, 0); +} + +static void fsi_irq_init(struct fsi_priv *fsi, int is_play) +{ + u32 data; + u32 ctrl; + + data = fsi_port_ab_io_bit(fsi, is_play); + ctrl = is_play ? DOFF_CTL : DIFF_CTL; + + /* set IMSK */ + fsi_irq_disable(fsi, is_play); + + /* set interrupt generation factor */ + fsi_reg_write(fsi, ctrl, IRQ_HALF); + + /* clear FIFO */ + fsi_reg_mask_set(fsi, ctrl, FIFO_CLR, FIFO_CLR); + + /* clear interrupt factor */ + fsi_master_mask_set(INT_ST, data, 0); +} + +static void fsi_soft_all_reset(void) +{ + u32 status = fsi_master_read(SOFT_RST); + + /* port AB reset */ + status &= 0x000000ff; + fsi_master_write(SOFT_RST, status); + mdelay(10); + + /* soft reset */ + status &= 0x000000f0; + fsi_master_write(SOFT_RST, status); + status |= 0x00000001; + fsi_master_write(SOFT_RST, status); + mdelay(10); +} + +static void fsi_16data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u16 *dma_start; + u32 snd; + int i; + + /* get dma start position for FSI */ + dma_start = (u16 *)runtime->dma_area; + dma_start += fsi->byte_offset / 2; + + /* + * soft dma + * FSI can not use DMA when 16bpp + */ + for (i = 0; i < send; i++) { + snd = (u32)dma_start[i]; + fsi_reg_write(fsi, DODT, snd << 8); + } +} + +static void fsi_32data_push(struct fsi_priv *fsi, + struct snd_pcm_runtime *runtime, + int send) +{ + u32 *dma_start; + + /* get dma start position for FSI */ + dma_start = (u32 *)runtime->dma_area; + dma_start += fsi->byte_offset / 4; + + dma_wait_for_completion(fsi->dma_chan); + dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); + dma_write(fsi->dma_chan, (u32)dma_start, + (u32)(fsi->base + DODT), send * 4); +} + +/* playback interrupt */ +static int fsi_data_push(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int send; + int fifo_free; + int width; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get send size for alsa */ + send = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get FIFO free size */ + fifo_free = (fsi->fifo_max * fsi->chan) - fsi_get_fifo_residue(fsi, 1); + + /* size check */ + if (fifo_free < send) + send = fifo_free; + + if (2 == width) + fsi_16data_push(fsi, runtime, send); + else if (4 == width) + fsi_32data_push(fsi, runtime, send); + else + return -EINVAL; + + fsi->byte_offset += send * width; + + fsi_irq_enable(fsi, 1); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + +static irqreturn_t fsi_interrupt(int irq, void *data) +{ + u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; + u32 int_st = fsi_master_read(INT_ST); + + /* clear irq status */ + fsi_master_write(SOFT_RST, status); + fsi_master_write(SOFT_RST, status | 0x00000010); + + if (int_st & INT_A_OUT) + fsi_data_push(&master->fsia); + if (int_st & INT_B_OUT) + fsi_data_push(&master->fsib); + + fsi_master_write(INT_ST, 0x0000000); + + return IRQ_HANDLED; +} + +/************************************************************************ + + + dai ops + + +************************************************************************/ +static int fsi_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + const char *msg; + u32 flags = fsi_get_info_flags(fsi); + u32 fmt; + u32 reg; + u32 data; + int is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int is_master; + int ret = 0; + + clk_enable(master->clk); + + /* CKG1 */ + data = is_play ? (1 << 0) : (1 << 4); + is_master = fsi_is_master_mode(fsi, is_play); + if (is_master) + fsi_reg_mask_set(fsi, CKG1, data, data); + else + fsi_reg_mask_set(fsi, CKG1, data, 0); + + /* clock inversion (CKG2) */ + data = 0; + switch (SH_FSI_INVERSION_MASK & flags) { + case SH_FSI_LRM_INV: + data = 1 << 12; + break; + case SH_FSI_BRM_INV: + data = 1 << 8; + break; + case SH_FSI_LRS_INV: + data = 1 << 4; + break; + case SH_FSI_BRS_INV: + data = 1 << 0; + break; + } + fsi_reg_write(fsi, CKG2, data); + + /* do fmt, di fmt */ + data = 0; + reg = is_play ? DO_FMT : DI_FMT; + fmt = is_play ? SH_FSI_GET_OFMT(flags) : SH_FSI_GET_IFMT(flags); + switch (fmt) { + case SH_FSI_FMT_MONO: + msg = "MONO"; + data = CR_FMT(CR_MONO); + fsi->chan = 1; + break; + case SH_FSI_FMT_MONO_DELAY: + msg = "MONO Delay"; + data = CR_FMT(CR_MONO_D); + fsi->chan = 1; + break; + case SH_FSI_FMT_PCM: + msg = "PCM"; + data = CR_FMT(CR_PCM); + fsi->chan = 2; + break; + case SH_FSI_FMT_I2S: + msg = "I2S"; + data = CR_FMT(CR_I2S); + fsi->chan = 2; + break; + case SH_FSI_FMT_TDM: + msg = "TDM"; + data = CR_FMT(CR_TDM) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + case SH_FSI_FMT_TDM_DELAY: + msg = "TDM Delay"; + data = CR_FMT(CR_TDM_D) | (fsi->chan - 1); + fsi->chan = is_play ? + SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags); + break; + default: + dev_err(dai->dev, "unknown format.\n"); + return -EINVAL; + } + + switch (fsi->chan) { + case 1: + fsi->fifo_max = 256; + break; + case 2: + fsi->fifo_max = 128; + break; + case 3: + case 4: + fsi->fifo_max = 64; + break; + case 5: + case 6: + case 7: + case 8: + fsi->fifo_max = 32; + break; + default: + dev_err(dai->dev, "channel size error.\n"); + return -EINVAL; + } + + fsi_reg_write(fsi, reg, data); + dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n", + msg, fsi->chan, fsi->dma_chan); + + /* + * clear clk reset if master mode + */ + if (is_master) + fsi_clk_ctrl(fsi, 1); + + /* irq setting */ + fsi_irq_init(fsi, is_play); + + return ret; +} + +static void fsi_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + fsi_irq_disable(fsi, is_play); + fsi_clk_ctrl(fsi, 0); + + clk_disable(master->clk); +} + +static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct fsi_priv *fsi = fsi_get(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int ret = 0; + + /* capture not supported */ + if (!is_play) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + fsi_stream_push(fsi, substream, + frames_to_bytes(runtime, runtime->buffer_size), + frames_to_bytes(runtime, runtime->period_size)); + ret = fsi_data_push(fsi); + break; + case SNDRV_PCM_TRIGGER_STOP: + fsi_irq_disable(fsi, is_play); + fsi_stream_pop(fsi); + break; + } + + return ret; +} + +static struct snd_soc_dai_ops fsi_dai_ops = { + .startup = fsi_dai_startup, + .shutdown = fsi_dai_shutdown, + .trigger = fsi_dai_trigger, +}; + +/************************************************************************ + + + pcm ops + + +************************************************************************/ +static struct snd_pcm_hardware fsi_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE, + .formats = FSI_FMTS, + .rates = FSI_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8192, + .periods_min = 1, + .periods_max = 32, + .fifo_size = 256, +}; + +static int fsi_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &fsi_pcm_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + + return ret; +} + +static int fsi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int fsi_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct fsi_priv *fsi = fsi_get(substream); + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + long location; + + location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + if (location < 0) + location = 0; + + return bytes_to_frames(runtime, location); +} + +static struct snd_pcm_ops fsi_pcm_ops = { + .open = fsi_pcm_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = fsi_hw_params, + .hw_free = fsi_hw_free, + .pointer = fsi_pointer, +}; + +/************************************************************************ + + + snd_soc_platform + + +************************************************************************/ +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static void fsi_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int fsi_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + /* + * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + return snd_pcm_lib_preallocate_pages_for_all( + pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); +} + +/************************************************************************ + + + alsa struct + + +************************************************************************/ +struct snd_soc_dai fsi_soc_dai[] = { + { + .name = "FSIA", + .id = 0, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + }, + { + .name = "FSIB", + .id = 1, + .playback = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, + /* capture not supported */ + .ops = &fsi_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(fsi_soc_dai); + +struct snd_soc_platform fsi_soc_platform = { + .name = "fsi-pcm", + .pcm_ops = &fsi_pcm_ops, + .pcm_new = fsi_pcm_new, + .pcm_free = fsi_pcm_free, +}; +EXPORT_SYMBOL_GPL(fsi_soc_platform); + +/************************************************************************ + + + platform function + + +************************************************************************/ +static int fsi_probe(struct platform_device *pdev) +{ + struct resource *res; + char clk_name[8]; + unsigned int irq; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + irq = platform_get_irq(pdev, 0); + if (!res || !irq) { + dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); + ret = -ENODEV; + goto exit; + } + + master = kzalloc(sizeof(*master), GFP_KERNEL); + if (!master) { + dev_err(&pdev->dev, "Could not allocate master\n"); + ret = -ENOMEM; + goto exit; + } + + master->base = ioremap_nocache(res->start, resource_size(res)); + if (!master->base) { + ret = -ENXIO; + dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); + goto exit_kfree; + } + + master->irq = irq; + master->info = pdev->dev.platform_data; + master->fsia.base = master->base; + master->fsib.base = master->base + 0x40; + + master->fsia.dma_chan = -1; + master->fsib.dma_chan = -1; + + ret = fsi_get_dma_chan(); + if (ret < 0) { + dev_err(&pdev->dev, "cannot get dma api\n"); + goto exit_iounmap; + } + + /* FSI is based on SPU mstp */ + snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); + master->clk = clk_get(NULL, clk_name); + if (IS_ERR(master->clk)) { + dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); + ret = -EIO; + goto exit_free_dma; + } + + fsi_soc_dai[0].dev = &pdev->dev; + fsi_soc_dai[1].dev = &pdev->dev; + + fsi_soft_all_reset(); + + ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); + if (ret) { + dev_err(&pdev->dev, "irq request err\n"); + goto exit_free_dma; + } + + ret = snd_soc_register_platform(&fsi_soc_platform); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd soc register\n"); + goto exit_free_irq; + } + + return snd_soc_register_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + +exit_free_irq: + free_irq(irq, master); +exit_free_dma: + fsi_free_dma_chan(); +exit_iounmap: + iounmap(master->base); +exit_kfree: + kfree(master); + master = NULL; +exit: + return ret; +} + +static int fsi_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&fsi_soc_platform); + + clk_put(master->clk); + + fsi_free_dma_chan(); + + free_irq(master->irq, master); + + iounmap(master->base); + kfree(master); + master = NULL; + return 0; +} + +static struct platform_driver fsi_driver = { + .driver = { + .name = "sh_fsi", + }, + .probe = fsi_probe, + .remove = fsi_remove, +}; + +static int __init fsi_mobile_init(void) +{ + return platform_driver_register(&fsi_driver); +} + +static void __exit fsi_mobile_exit(void) +{ + platform_driver_unregister(&fsi_driver); +} +module_init(fsi_mobile_init); +module_exit(fsi_mobile_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c new file mode 100644 index 000000000000..c8ceddc2a26c --- /dev/null +++ b/sound/soc/soc-cache.c @@ -0,0 +1,218 @@ +/* + * soc-cache.c -- ASoC register cache helpers + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + +static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *cache = codec->reg_cache; + u8 data[2]; + int ret; + + BUG_ON(codec->volatile_register); + + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +#if defined(CONFIG_SPI_MASTER) +static int snd_soc_7_9_spi_write(void *control_data, const char *data, + int len) +{ + struct spi_device *spi = control_data; + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#else +#define snd_soc_7_9_spi_write NULL +#endif + +static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *reg_cache = codec->reg_cache; + u8 data[3]; + + data[0] = reg; + data[1] = (value >> 8) & 0xff; + data[2] = value & 0xff; + + if (!snd_soc_codec_volatile_register(codec, reg)) + reg_cache[reg] = value; + + if (codec->hw_write(codec->control_data, data, 3) == 3) + return 0; + else + return -EIO; +} + +static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size || + snd_soc_codec_volatile_register(codec, reg)) + return codec->hw_read(codec, reg); + else + return cache[reg]; +} + +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, + unsigned int r) +{ + struct i2c_msg xfer[2]; + u8 reg = r; + u16 data; + int ret; + struct i2c_client *client = codec->control_data; + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 1; + xfer[0].buf = ® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 2; + xfer[1].buf = (u8 *)&data; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret != 2) { + dev_err(&client->dev, "i2c_transfer() returned %d\n", ret); + return 0; + } + + return (data >> 8) | ((data & 0xff) << 8); +} +#else +#define snd_soc_8_16_read_i2c NULL +#endif + +static struct { + int addr_bits; + int data_bits; + int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); + int (*spi_write)(void *, const char *, int); + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); +} io_types[] = { + { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, + { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, + snd_soc_8_16_read_i2c }, +}; + +/** + * snd_soc_codec_set_cache_io: Set up standard I/O functions. + * + * @codec: CODEC to configure. + * @type: Type of cache. + * @addr_bits: Number of bits of register address data. + * @data_bits: Number of bits of data per register. + * @control: Control bus used. + * + * Register formats are frequently shared between many I2C and SPI + * devices. In order to promote code reuse the ASoC core provides + * some standard implementations of CODEC read and write operations + * which can be set up using this function. + * + * The caller is responsible for allocating and initialising the + * actual cache. + * + * Note that at present this code cannot be used by CODECs with + * volatile registers. + */ +int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, + int addr_bits, int data_bits, + enum snd_soc_control_type control) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(io_types); i++) + if (io_types[i].addr_bits == addr_bits && + io_types[i].data_bits == data_bits) + break; + if (i == ARRAY_SIZE(io_types)) { + printk(KERN_ERR + "No I/O functions for %d bit address %d bit data\n", + addr_bits, data_bits); + return -EINVAL; + } + + codec->write = io_types[i].write; + codec->read = io_types[i].read; + + switch (control) { + case SND_SOC_CUSTOM: + break; + + case SND_SOC_I2C: +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) + codec->hw_write = (hw_write_t)i2c_master_send; +#endif + if (io_types[i].i2c_read) + codec->hw_read = io_types[i].i2c_read; + break; + + case SND_SOC_SPI: + if (io_types[i].spi_write) + codec->hw_write = io_types[i].spi_write; + break; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1d70829464ef..7ff04ad2a97e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -28,6 +28,7 @@ #include <linux/bitops.h> #include <linux/debugfs.h> #include <linux/platform_device.h> +#include <sound/ac97_codec.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -619,8 +620,9 @@ static struct snd_pcm_ops soc_pcm_ops = { #ifdef CONFIG_PM /* powers down audio subsystem for suspend */ -static int soc_suspend(struct platform_device *pdev, pm_message_t state) +static int soc_suspend(struct device *dev) { + struct platform_device *pdev = to_platform_device(dev); struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = card->platform; @@ -656,7 +658,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) snd_pcm_suspend_all(card->dai_link[i].pcm); if (card->suspend_pre) - card->suspend_pre(pdev, state); + card->suspend_pre(pdev, PMSG_SUSPEND); for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; @@ -682,7 +684,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) } if (codec_dev->suspend) - codec_dev->suspend(pdev, state); + codec_dev->suspend(pdev, PMSG_SUSPEND); for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; @@ -691,7 +693,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) } if (card->suspend_post) - card->suspend_post(pdev, state); + card->suspend_post(pdev, PMSG_SUSPEND); return 0; } @@ -765,8 +767,9 @@ static void soc_resume_deferred(struct work_struct *work) } /* powers up audio subsystem after a suspend */ -static int soc_resume(struct platform_device *pdev) +static int soc_resume(struct device *dev) { + struct platform_device *pdev = to_platform_device(dev); struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_card *card = socdev->card; struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; @@ -788,6 +791,44 @@ static int soc_resume(struct platform_device *pdev) return 0; } +/** + * snd_soc_suspend_device: Notify core of device suspend + * + * @dev: Device being suspended. + * + * In order to ensure that the entire audio subsystem is suspended in a + * coordinated fashion ASoC devices should suspend themselves when + * called by ASoC. When the standard kernel suspend process asks the + * device to suspend it should call this function to initiate a suspend + * of the entire ASoC card. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_suspend_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_suspend_device); + +/** + * snd_soc_resume_device: Notify core of device resume + * + * @dev: Device being resumed. + * + * In order to ensure that the entire audio subsystem is resumed in a + * coordinated fashion ASoC devices should resume themselves when called + * by ASoC. When the standard kernel resume process asks the device + * to resume it should call this function. Once all the components of + * the card have notified that they are ready to be resumed the card + * will be resumed. + * + * \note Currently this function is stubbed out. + */ +int snd_soc_resume_device(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_resume_device); #else #define soc_suspend NULL #define soc_resume NULL @@ -981,16 +1022,39 @@ static int soc_remove(struct platform_device *pdev) return 0; } +static int soc_poweroff(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; + + if (!card->instantiated) + return 0; + + /* Flush out pmdown_time work - we actually do want to run it + * now, we're shutting down so no imminent restart. */ + run_delayed_work(&card->delayed_work); + + snd_soc_dapm_shutdown(socdev); + + return 0; +} + +static struct dev_pm_ops soc_pm_ops = { + .suspend = soc_suspend, + .resume = soc_resume, + .poweroff = soc_poweroff, +}; + /* ASoC platform driver */ static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", .owner = THIS_MODULE, + .pm = &soc_pm_ops, }, .probe = soc_probe, .remove = soc_remove, - .suspend = soc_suspend, - .resume = soc_resume, }; /* create a new pcm */ @@ -1062,6 +1126,23 @@ static int soc_new_pcm(struct snd_soc_device *socdev, return ret; } +/** + * snd_soc_codec_volatile_register: Report if a register is volatile. + * + * @codec: CODEC to query. + * @reg: Register to query. + * + * Boolean function indiciating if a CODEC register is volatile. + */ +int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) +{ + if (codec->volatile_register) + return codec->volatile_register(reg); + else + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); + /* codec register dump */ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { @@ -1075,6 +1156,9 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) count += sprintf(buf, "%s registers\n", codec->name); for (i = 0; i < codec->reg_cache_size; i += step) { + if (codec->readable_register && !codec->readable_register(i)) + continue; + count += sprintf(buf + count, "%2x: ", i); if (count >= PAGE_SIZE - 1) break; @@ -1183,10 +1267,18 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) if (!codec->debugfs_pop_time) printk(KERN_WARNING "Failed to create pop time debugfs file\n"); + + codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root); + if (!codec->debugfs_dapm) + printk(KERN_WARNING + "Failed to create DAPM debugfs directory\n"); + + snd_soc_dapm_debugfs_init(codec); } static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { + debugfs_remove_recursive(codec->debugfs_dapm); debugfs_remove(codec->debugfs_pop_time); debugfs_remove(codec->debugfs_reg); } @@ -1264,10 +1356,10 @@ EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); * Returns 1 for change else 0. */ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1294,10 +1386,10 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, - unsigned short mask, unsigned short value) + unsigned int mask, unsigned int value) { int change; - unsigned short old, new; + unsigned int old, new; mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); @@ -1381,8 +1473,11 @@ int snd_soc_init_card(struct snd_soc_device *socdev) continue; } } - if (card->dai_link[i].codec_dai->ac97_control) + if (card->dai_link[i].codec_dai->ac97_control) { ac97 = 1; + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } } snprintf(codec->card->shortname, sizeof(codec->card->shortname), "%s", card->name); @@ -1586,7 +1681,7 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1615,8 +1710,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask, bitmask; + unsigned int val; + unsigned int mask, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1652,7 +1747,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1691,8 +1786,8 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; + unsigned int val; + unsigned int mask; if (ucontrol->value.enumerated.item[0] > e->max - 1) return -EINVAL; @@ -1852,7 +1947,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val = (ucontrol->value.integer.value[0] & mask); if (invert) @@ -1918,7 +2013,7 @@ int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; int max = mc->max; - unsigned int mask = (1<<fls(max))-1; + unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; ucontrol->value.integer.value[0] = @@ -1958,7 +2053,7 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] & mask); @@ -2050,7 +2145,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; int min = mc->min; - unsigned short val; + unsigned int val; val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; @@ -2136,17 +2231,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 21c69074aa17..0d8b08ef8731 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -37,6 +37,7 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -52,19 +53,41 @@ /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { - snd_soc_dapm_pre, snd_soc_dapm_supply, snd_soc_dapm_micbias, - snd_soc_dapm_mic, snd_soc_dapm_mux, snd_soc_dapm_value_mux, - snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, - snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, - snd_soc_dapm_post + [snd_soc_dapm_pre] = 0, + [snd_soc_dapm_supply] = 1, + [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_aif_in] = 3, + [snd_soc_dapm_aif_out] = 3, + [snd_soc_dapm_mic] = 4, + [snd_soc_dapm_mux] = 5, + [snd_soc_dapm_value_mux] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mixer] = 7, + [snd_soc_dapm_mixer_named_ctl] = 7, + [snd_soc_dapm_pga] = 8, + [snd_soc_dapm_adc] = 9, + [snd_soc_dapm_hp] = 10, + [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_post] = 11, }; static int dapm_down_seq[] = { - snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, - snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer, - snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias, - snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_supply, - snd_soc_dapm_post + [snd_soc_dapm_pre] = 0, + [snd_soc_dapm_adc] = 1, + [snd_soc_dapm_hp] = 2, + [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_pga] = 4, + [snd_soc_dapm_mixer_named_ctl] = 5, + [snd_soc_dapm_mixer] = 5, + [snd_soc_dapm_dac] = 6, + [snd_soc_dapm_mic] = 7, + [snd_soc_dapm_micbias] = 8, + [snd_soc_dapm_mux] = 9, + [snd_soc_dapm_value_mux] = 9, + [snd_soc_dapm_aif_in] = 10, + [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_supply] = 11, + [snd_soc_dapm_post] = 12, }; static void pop_wait(u32 pop_time) @@ -130,8 +153,12 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, if (card->set_bias_level) ret = card->set_bias_level(card, level); - if (ret == 0 && codec->set_bias_level) - ret = codec->set_bias_level(codec, level); + if (ret == 0) { + if (codec->set_bias_level) + ret = codec->set_bias_level(codec, level); + else + codec->bias_level = level; + } return ret; } @@ -206,6 +233,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: p->connect = 1; break; /* does effect routing - dynamically connected */ @@ -268,7 +297,7 @@ static int dapm_connect_mixer(struct snd_soc_codec *codec, static int dapm_update_bits(struct snd_soc_dapm_widget *widget) { int change, power; - unsigned short old, new; + unsigned int old, new; struct snd_soc_codec *codec = widget->codec; /* check for valid widgets */ @@ -479,8 +508,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) if (widget->id == snd_soc_dapm_supply) return 0; - if (widget->id == snd_soc_dapm_adc && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -519,8 +554,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 0; /* active stream ? */ - if (widget->id == snd_soc_dapm_dac && widget->active) - return 1; + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + if (widget->active) + return 1; + default: + break; + } if (widget->connected) { /* connected pin ? */ @@ -689,53 +730,211 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) return power; } -/* - * Scan a single DAPM widget for a complete audio path and update the - * power status appropriately. - */ -static int dapm_power_widget(struct snd_soc_codec *codec, int event, - struct snd_soc_dapm_widget *w) +static int dapm_seq_compare(struct snd_soc_dapm_widget *a, + struct snd_soc_dapm_widget *b, + int sort[]) { - int ret; + if (sort[a->id] != sort[b->id]) + return sort[a->id] - sort[b->id]; + if (a->reg != b->reg) + return a->reg - b->reg; - switch (w->id) { - case snd_soc_dapm_pre: - if (!w->event) - return 0; + return 0; +} - if (event == SND_SOC_DAPM_STREAM_START) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMU); +/* Insert a widget in order into a DAPM power sequence. */ +static void dapm_seq_insert(struct snd_soc_dapm_widget *new_widget, + struct list_head *list, + int sort[]) +{ + struct snd_soc_dapm_widget *w; + + list_for_each_entry(w, list, power_list) + if (dapm_seq_compare(new_widget, w, sort) < 0) { + list_add_tail(&new_widget->power_list, &w->power_list); + return; + } + + list_add_tail(&new_widget->power_list, list); +} + +/* Apply the coalesced changes from a DAPM sequence */ +static void dapm_seq_run_coalesced(struct snd_soc_codec *codec, + struct list_head *pending) +{ + struct snd_soc_dapm_widget *w; + int reg, power, ret; + unsigned int value = 0; + unsigned int mask = 0; + unsigned int cur_mask; + + reg = list_first_entry(pending, struct snd_soc_dapm_widget, + power_list)->reg; + + list_for_each_entry(w, pending, power_list) { + cur_mask = 1 << w->shift; + BUG_ON(reg != w->reg); + + if (w->invert) + power = !w->power; + else + power = w->power; + + mask |= cur_mask; + if (power) + value |= cur_mask; + + pop_dbg(codec->pop_time, + "pop test : Queue %s: reg=0x%x, 0x%x/0x%x\n", + w->name, reg, value, mask); + + /* power up pre event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + pop_dbg(codec->pop_time, "pop test : %s PRE_PMU\n", + w->name); + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); if (ret < 0) - return ret; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, - NULL, SND_SOC_DAPM_PRE_PMD); + pr_err("%s: pre event failed: %d\n", + w->name, ret); + } + + /* power down pre event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + pop_dbg(codec->pop_time, "pop test : %s PRE_PMD\n", + w->name); + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); if (ret < 0) - return ret; + pr_err("%s: pre event failed: %d\n", + w->name, ret); } - return 0; - case snd_soc_dapm_post: - if (!w->event) - return 0; + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !w->power) + dapm_set_pga(w, w->power); + } - if (event == SND_SOC_DAPM_STREAM_START) { + if (reg >= 0) { + pop_dbg(codec->pop_time, + "pop test : Applying 0x%x/0x%x to %x in %dms\n", + value, mask, reg, codec->pop_time); + pop_wait(codec->pop_time); + snd_soc_update_bits(codec, reg, mask, value); + } + + list_for_each_entry(w, pending, power_list) { + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && w->power) + dapm_set_pga(w, w->power); + + /* power up post event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + pop_dbg(codec->pop_time, "pop test : %s POST_PMU\n", + w->name); ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMU); if (ret < 0) - return ret; - } else if (event == SND_SOC_DAPM_STREAM_STOP) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMD); + pr_err("%s: post event failed: %d\n", + w->name, ret); + } + + /* power down post event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + pop_dbg(codec->pop_time, "pop test : %s POST_PMD\n", + w->name); + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); if (ret < 0) - return ret; + pr_err("%s: post event failed: %d\n", + w->name, ret); } - return 0; + } +} - default: - return dapm_generic_apply_power(w); +/* Apply a DAPM power sequence. + * + * We walk over a pre-sorted list of widgets to apply power to. In + * order to minimise the number of writes to the device required + * multiple widgets will be updated in a single write where possible. + * Currently anything that requires more than a single write is not + * handled. + */ +static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, + int event, int sort[]) +{ + struct snd_soc_dapm_widget *w, *n; + LIST_HEAD(pending); + int cur_sort = -1; + int cur_reg = SND_SOC_NOPM; + int ret; + + list_for_each_entry_safe(w, n, list, power_list) { + ret = 0; + + /* Do we need to apply any queued changes? */ + if (sort[w->id] != cur_sort || w->reg != cur_reg) { + if (!list_empty(&pending)) + dapm_seq_run_coalesced(codec, &pending); + + INIT_LIST_HEAD(&pending); + cur_sort = -1; + cur_reg = SND_SOC_NOPM; + } + + switch (w->id) { + case snd_soc_dapm_pre: + if (!w->event) + list_for_each_entry_safe_continue(w, n, list, + power_list); + + if (event == SND_SOC_DAPM_STREAM_START) + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMU); + else if (event == SND_SOC_DAPM_STREAM_STOP) + ret = w->event(w, + NULL, SND_SOC_DAPM_PRE_PMD); + break; + + case snd_soc_dapm_post: + if (!w->event) + list_for_each_entry_safe_continue(w, n, list, + power_list); + + if (event == SND_SOC_DAPM_STREAM_START) + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + else if (event == SND_SOC_DAPM_STREAM_STOP) + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMD); + break; + + case snd_soc_dapm_input: + case snd_soc_dapm_output: + case snd_soc_dapm_hp: + case snd_soc_dapm_mic: + case snd_soc_dapm_line: + case snd_soc_dapm_spk: + /* No register support currently */ + ret = dapm_generic_apply_power(w); + break; + + default: + /* Queue it up for application */ + cur_sort = sort[w->id]; + cur_reg = w->reg; + list_move(&w->power_list, &pending); + break; + } + + if (ret < 0) + pr_err("Failed to apply widget power: %d\n", + ret); } + + if (!list_empty(&pending)) + dapm_seq_run_coalesced(codec, &pending); } /* @@ -751,23 +950,22 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { struct snd_soc_device *socdev = codec->socdev; struct snd_soc_dapm_widget *w; + LIST_HEAD(up_list); + LIST_HEAD(down_list); int ret = 0; - int i, power; + int power; int sys_power = 0; - INIT_LIST_HEAD(&codec->up_list); - INIT_LIST_HEAD(&codec->down_list); - /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ list_for_each_entry(w, &codec->dapm_widgets, list) { switch (w->id) { case snd_soc_dapm_pre: - list_add_tail(&codec->down_list, &w->power_list); + dapm_seq_insert(w, &down_list, dapm_down_seq); break; case snd_soc_dapm_post: - list_add_tail(&codec->up_list, &w->power_list); + dapm_seq_insert(w, &up_list, dapm_up_seq); break; default: @@ -782,16 +980,31 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) continue; if (power) - list_add_tail(&w->power_list, &codec->up_list); + dapm_seq_insert(w, &up_list, dapm_up_seq); else - list_add_tail(&w->power_list, - &codec->down_list); + dapm_seq_insert(w, &down_list, dapm_down_seq); w->power = power; break; } } + /* If there are no DAPM widgets then try to figure out power from the + * event type. + */ + if (list_empty(&codec->dapm_widgets)) { + switch (event) { + case SND_SOC_DAPM_STREAM_START: + case SND_SOC_DAPM_STREAM_RESUME: + sys_power = 1; + break; + case SND_SOC_DAPM_STREAM_NOP: + sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY; + default: + break; + } + } + /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { @@ -802,32 +1015,10 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } /* Power down widgets first; try to avoid amplifying pops. */ - for (i = 0; i < ARRAY_SIZE(dapm_down_seq); i++) { - list_for_each_entry(w, &codec->down_list, power_list) { - /* is widget in stream order */ - if (w->id != dapm_down_seq[i]) - continue; - - ret = dapm_power_widget(codec, event, w); - if (ret != 0) - pr_err("Failed to power down %s: %d\n", - w->name, ret); - } - } + dapm_seq_run(codec, &down_list, event, dapm_down_seq); /* Now power up. */ - for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) { - list_for_each_entry(w, &codec->up_list, power_list) { - /* is widget in stream order */ - if (w->id != dapm_up_seq[i]) - continue; - - ret = dapm_power_widget(codec, event, w); - if (ret != 0) - pr_err("Failed to power up %s: %d\n", - w->name, ret); - } - } + dapm_seq_run(codec, &up_list, event, dapm_up_seq); /* If we just powered the last thing off drop to standby bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { @@ -845,6 +1036,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) pr_err("Failed to apply active bias: %d\n", ret); } + pop_dbg(codec->pop_time, "DAPM sequencing finished, waiting %dms\n", + codec->pop_time); + return 0; } @@ -881,6 +1075,8 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -906,6 +1102,92 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) } #endif +#ifdef CONFIG_DEBUG_FS +static int dapm_widget_power_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t dapm_widget_power_read_file(struct file *file, + char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_dapm_widget *w = file->private_data; + char *buf; + int in, out; + ssize_t ret; + struct snd_soc_dapm_path *p = NULL; + + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + + ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d\n", + w->name, w->power ? "On" : "Off", in, out); + + if (w->active && w->sname) + ret += snprintf(buf, PAGE_SIZE - ret, " stream %s active\n", + w->sname); + + list_for_each_entry(p, &w->sources, list_sink) { + if (p->connect) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + " in %s %s\n", + p->name ? p->name : "static", + p->source->name); + } + list_for_each_entry(p, &w->sinks, list_source) { + if (p->connect) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + " out %s %s\n", + p->name ? p->name : "static", + p->sink->name); + } + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + return ret; +} + +static const struct file_operations dapm_widget_power_fops = { + .open = dapm_widget_power_open_file, + .read = dapm_widget_power_read_file, +}; + +void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_widget *w; + struct dentry *d; + + if (!codec->debugfs_dapm) + return; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (!w->name) + continue; + + d = debugfs_create_file(w->name, 0444, + codec->debugfs_dapm, w, + &dapm_widget_power_fops); + if (!d) + printk(KERN_WARNING + "ASoC: Failed to create %s debugfs file\n", + w->name); + } +} +#else +void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) +{ +} +#endif + /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int mask, @@ -1138,8 +1420,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, if (wsink->id == snd_soc_dapm_input) { if (wsource->id == snd_soc_dapm_micbias || wsource->id == snd_soc_dapm_mic || - wsink->id == snd_soc_dapm_line || - wsink->id == snd_soc_dapm_output) + wsource->id == snd_soc_dapm_line || + wsource->id == snd_soc_dapm_output) wsink->ext = 1; } if (wsource->id == snd_soc_dapm_output) { @@ -1171,6 +1453,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, case snd_soc_dapm_pre: case snd_soc_dapm_post: case snd_soc_dapm_supply: + case snd_soc_dapm_aif_in: + case snd_soc_dapm_aif_out: list_add(&path->list, &codec->dapm_paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -1273,9 +1557,11 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) dapm_new_mux(codec, w); break; case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -1372,7 +1658,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val2, val_mask; + unsigned int val, val2, val_mask; int ret; val = (ucontrol->value.integer.value[0] & mask); @@ -1436,7 +1722,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, bitmask; + unsigned int val, bitmask; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) ; @@ -1464,8 +1750,8 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask, bitmask; + unsigned int val, mux; + unsigned int mask, bitmask; int ret = 0; for (bitmask = 1; bitmask < e->max; bitmask <<= 1) @@ -1523,7 +1809,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short reg_val, val, mux; + unsigned int reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; @@ -1563,8 +1849,8 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val, mux; - unsigned short mask; + unsigned int val, mux; + unsigned int mask; int ret = 0; if (ucontrol->value.enumerated.item[0] > e->max - 1) @@ -1880,6 +2166,36 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_dapm_widget *w; + LIST_HEAD(down_list); + int powerdown = 0; + + list_for_each_entry(w, &codec->dapm_widgets, list) { + if (w->power) { + dapm_seq_insert(w, &down_list, dapm_down_seq); + w->power = 0; + powerdown = 1; + } + } + + /* If there were no widgets to power down we're already in + * standby. + */ + if (powerdown) { + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + dapm_seq_run(codec, &down_list, 0, dapm_down_seq); + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + } + + snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); +} + /* Module information */ MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk"); MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 28346fb2e70c..1d455ab79490 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -73,14 +73,15 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) oldstatus = jack->status; jack->status &= ~mask; - jack->status |= status; + jack->status |= status & mask; - /* The DAPM sync is expensive enough to be worth skipping */ - if (jack->status == oldstatus) + /* The DAPM sync is expensive enough to be worth skipping. + * However, empty mask means pin synchronization is desired. */ + if (mask && (jack->status == oldstatus)) goto out; list_for_each_entry(pin, &jack->pins, list) { - enable = pin->mask & status; + enable = pin->mask & jack->status; if (pin->invert) enable = !enable; @@ -220,6 +221,9 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto err; + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + ret = request_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, @@ -228,8 +232,13 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (ret) goto err; - INIT_WORK(&gpios[i].work, gpio_work); - gpios[i].jack = jack; +#ifdef CONFIG_GPIO_SYSFS + /* Expose GPIO value over sysfs for diagnostic purposes */ + gpio_export(gpios[i].gpio, false); +#endif + + /* Update initial jack status */ + snd_soc_jack_gpio_detect(&gpios[i]); } return 0; @@ -258,6 +267,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, int i; for (i = 0; i < count; i++) { +#ifdef CONFIG_GPIO_SYSFS + gpio_unexport(gpios[i].gpio); +#endif free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); gpio_free(gpios[i].gpio); gpios[i].jack = NULL; diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 938a58a5a244..efed64b8b026 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -297,15 +297,17 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, static bool filter(struct dma_chan *chan, void *param) { struct txx9aclc_dmadata *dmadata = param; - char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */ + char *devname; + bool found = false; - snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name, + devname = kasprintf(GFP_KERNEL, "%s.%d", dmadata->dma_res->name, (int)dmadata->dma_res->start); if (strcmp(dev_name(chan->device->dev), devname) == 0) { chan->private = &dmadata->dma_slave; - return true; + found = true; } - return false; + kfree(devname); + return found; } static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, diff --git a/sound/sound_core.c b/sound/sound_core.c index a41f8b127f49..bb4b88e606bb 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -128,6 +128,46 @@ extern int msnd_pinnacle_init(void); #endif /* + * By default, OSS sound_core claims full legacy minor range (0-255) + * of SOUND_MAJOR to trap open attempts to any sound minor and + * requests modules using custom sound-slot/service-* module aliases. + * The only benefit of doing this is allowing use of custom module + * aliases instead of the standard char-major-* ones. This behavior + * prevents alternative OSS implementation and is scheduled to be + * removed. + * + * CONFIG_SOUND_OSS_CORE_PRECLAIM and soundcore.preclaim_oss kernel + * parameter are added to allow distros and developers to try and + * switch to alternative implementations without needing to rebuild + * the kernel in the meantime. If preclaim_oss is non-zero, the + * kernel will behave the same as before. All SOUND_MAJOR minors are + * preclaimed and the custom module aliases along with standard chrdev + * ones are emitted if a missing device is opened. If preclaim_oss is + * zero, sound_core only grabs what's actually in use and for missing + * devices only the standard chrdev aliases are requested. + * + * All these clutters are scheduled to be removed along with + * sound-slot/service-* module aliases. Please take a look at + * feature-removal-schedule.txt for details. + */ +#ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM +static int preclaim_oss = 1; +#else +static int preclaim_oss = 0; +#endif + +module_param(preclaim_oss, int, 0444); + +static int soundcore_open(struct inode *, struct file *); + +static const struct file_operations soundcore_fops = +{ + /* We must have an owner or the module locking fails */ + .owner = THIS_MODULE, + .open = soundcore_open, +}; + +/* * Low level list operator. Scan the ordered list, find a hole and * join into it. Called with the lock asserted */ @@ -219,8 +259,9 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati if (!s) return -ENOMEM; - + spin_lock(&sound_loader_lock); +retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); @@ -231,11 +272,31 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati else sprintf(s->name, "sound/%s%d", name, r / SOUND_STEP); + if (!preclaim_oss) { + /* + * Something else might have grabbed the minor. If + * first free slot is requested, rescan with @low set + * to the next unit; otherwise, -EBUSY. + */ + r = __register_chrdev(SOUND_MAJOR, s->unit_minor, 1, s->name, + &soundcore_fops); + if (r < 0) { + spin_lock(&sound_loader_lock); + __sound_remove_unit(list, s->unit_minor); + if (index < 0) { + low = s->unit_minor + SOUND_STEP; + goto retry; + } + spin_unlock(&sound_loader_lock); + return -EBUSY; + } + } + device_create(sound_class, dev, MKDEV(SOUND_MAJOR, s->unit_minor), NULL, s->name+6); - return r; + return s->unit_minor; - fail: +fail: kfree(s); return r; } @@ -254,6 +315,9 @@ static void sound_remove_unit(struct sound_unit **list, int unit) p = __sound_remove_unit(list, unit); spin_unlock(&sound_loader_lock); if (p) { + if (!preclaim_oss) + __unregister_chrdev(SOUND_MAJOR, p->unit_minor, 1, + p->name); device_destroy(sound_class, MKDEV(SOUND_MAJOR, p->unit_minor)); kfree(p); } @@ -491,19 +555,6 @@ void unregister_sound_dsp(int unit) EXPORT_SYMBOL(unregister_sound_dsp); -/* - * Now our file operations - */ - -static int soundcore_open(struct inode *, struct file *); - -static const struct file_operations soundcore_fops= -{ - /* We must have an owner or the module locking fails */ - .owner = THIS_MODULE, - .open = soundcore_open, -}; - static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; @@ -539,8 +590,9 @@ static int soundcore_open(struct inode *inode, struct file *file) s = __look_for_unit(chain, unit); if (s) new_fops = fops_get(s->unit_fops); - if (!new_fops) { + if (preclaim_oss && !new_fops) { spin_unlock(&sound_loader_lock); + /* * Please, don't change this order or code. * For ALSA slot means soundcard and OSS emulation code @@ -550,6 +602,17 @@ static int soundcore_open(struct inode *inode, struct file *file) */ request_module("sound-slot-%i", unit>>4); request_module("sound-service-%i-%i", unit>>4, chain); + + /* + * sound-slot/service-* module aliases are scheduled + * for removal in favor of the standard char-major-* + * module aliases. For the time being, generate both + * the legacy and standard module aliases to ease + * transition. + */ + if (request_module("char-major-%d-%d", SOUND_MAJOR, unit) > 0) + request_module("char-major-%d", SOUND_MAJOR); + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -593,7 +656,8 @@ static void cleanup_oss_soundcore(void) static int __init init_oss_soundcore(void) { - if (register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops)==-1) { + if (preclaim_oss && + register_chrdev(SOUND_MAJOR, "sound", &soundcore_fops) == -1) { printk(KERN_ERR "soundcore: sound device already in use.\n"); return -EBUSY; } diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 44b9cdc8a83b..8db0374e10d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1083,6 +1083,8 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri } else urb_packs = 1; urb_packs *= packs_per_ms; + if (subs->syncpipe) + urb_packs = min(urb_packs, 1U << subs->syncinterval); /* decide how many packets to be used */ if (is_playback) { @@ -2124,8 +2126,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %#x (%d bits)\n", - fp->format, snd_pcm_format_width(fp->format)); + snd_iprintf(buffer, " Format: %s\n", + snd_pcm_format_name(fp->format)); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2fb35cc22a30..0eff19ceb7e1 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -45,6 +45,7 @@ #include <linux/slab.h> #include <linux/timer.h> #include <linux/usb.h> +#include <linux/wait.h> #include <sound/core.h> #include <sound/rawmidi.h> #include <sound/asequencer.h> @@ -62,6 +63,9 @@ */ #define ERROR_DELAY_JIFFIES (HZ / 10) +#define OUTPUT_URBS 7 +#define INPUT_URBS 7 + MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("USB Audio/MIDI helper module"); @@ -90,7 +94,7 @@ struct snd_usb_midi_endpoint; struct usb_protocol_ops { void (*input)(struct snd_usb_midi_in_endpoint*, uint8_t*, int); - void (*output)(struct snd_usb_midi_out_endpoint*); + void (*output)(struct snd_usb_midi_out_endpoint *ep, struct urb *urb); void (*output_packet)(struct urb*, uint8_t, uint8_t, uint8_t, uint8_t); void (*init_out_endpoint)(struct snd_usb_midi_out_endpoint*); void (*finish_out_endpoint)(struct snd_usb_midi_out_endpoint*); @@ -116,11 +120,15 @@ struct snd_usb_midi { struct snd_usb_midi_out_endpoint { struct snd_usb_midi* umidi; - struct urb* urb; - int urb_active; + struct out_urb_context { + struct urb *urb; + struct snd_usb_midi_out_endpoint *ep; + } urbs[OUTPUT_URBS]; + unsigned int active_urbs; + unsigned int drain_urbs; int max_transfer; /* size of urb buffer */ struct tasklet_struct tasklet; - + unsigned int next_urb; spinlock_t buffer_lock; struct usbmidi_out_port { @@ -139,11 +147,13 @@ struct snd_usb_midi_out_endpoint { uint8_t data[2]; } ports[0x10]; int current_port; + + wait_queue_head_t drain_wait; }; struct snd_usb_midi_in_endpoint { struct snd_usb_midi* umidi; - struct urb* urb; + struct urb* urbs[INPUT_URBS]; struct usbmidi_in_port { struct snd_rawmidi_substream *substream; u8 running_status_length; @@ -251,10 +261,17 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb) static void snd_usbmidi_out_urb_complete(struct urb* urb) { - struct snd_usb_midi_out_endpoint* ep = urb->context; + struct out_urb_context *context = urb->context; + struct snd_usb_midi_out_endpoint* ep = context->ep; + unsigned int urb_index; spin_lock(&ep->buffer_lock); - ep->urb_active = 0; + urb_index = context - ep->urbs; + ep->active_urbs &= ~(1 << urb_index); + if (unlikely(ep->drain_urbs)) { + ep->drain_urbs &= ~(1 << urb_index); + wake_up(&ep->drain_wait); + } spin_unlock(&ep->buffer_lock); if (urb->status < 0) { int err = snd_usbmidi_urb_error(urb->status); @@ -274,24 +291,38 @@ static void snd_usbmidi_out_urb_complete(struct urb* urb) */ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) { - struct urb* urb = ep->urb; + unsigned int urb_index; + struct urb* urb; unsigned long flags; spin_lock_irqsave(&ep->buffer_lock, flags); - if (ep->urb_active || ep->umidi->chip->shutdown) { + if (ep->umidi->chip->shutdown) { spin_unlock_irqrestore(&ep->buffer_lock, flags); return; } - urb->transfer_buffer_length = 0; - ep->umidi->usb_protocol_ops->output(ep); + urb_index = ep->next_urb; + for (;;) { + if (!(ep->active_urbs & (1 << urb_index))) { + urb = ep->urbs[urb_index].urb; + urb->transfer_buffer_length = 0; + ep->umidi->usb_protocol_ops->output(ep, urb); + if (urb->transfer_buffer_length == 0) + break; - if (urb->transfer_buffer_length > 0) { - dump_urb("sending", urb->transfer_buffer, - urb->transfer_buffer_length); - urb->dev = ep->umidi->chip->dev; - ep->urb_active = snd_usbmidi_submit_urb(urb, GFP_ATOMIC) >= 0; + dump_urb("sending", urb->transfer_buffer, + urb->transfer_buffer_length); + urb->dev = ep->umidi->chip->dev; + if (snd_usbmidi_submit_urb(urb, GFP_ATOMIC) < 0) + break; + ep->active_urbs |= 1 << urb_index; + } + if (++urb_index >= OUTPUT_URBS) + urb_index = 0; + if (urb_index == ep->next_urb) + break; } + ep->next_urb = urb_index; spin_unlock_irqrestore(&ep->buffer_lock, flags); } @@ -306,7 +337,7 @@ static void snd_usbmidi_out_tasklet(unsigned long data) static void snd_usbmidi_error_timer(unsigned long data) { struct snd_usb_midi *umidi = (struct snd_usb_midi *)data; - int i; + unsigned int i, j; spin_lock(&umidi->disc_lock); if (umidi->disconnected) { @@ -317,8 +348,10 @@ static void snd_usbmidi_error_timer(unsigned long data) struct snd_usb_midi_in_endpoint *in = umidi->endpoints[i].in; if (in && in->error_resubmit) { in->error_resubmit = 0; - in->urb->dev = umidi->chip->dev; - snd_usbmidi_submit_urb(in->urb, GFP_ATOMIC); + for (j = 0; j < INPUT_URBS; ++j) { + in->urbs[j]->dev = umidi->chip->dev; + snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); + } } if (umidi->endpoints[i].out) snd_usbmidi_do_output(umidi->endpoints[i].out); @@ -330,13 +363,14 @@ static void snd_usbmidi_error_timer(unsigned long data) static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep, const void *data, int len) { - int err; + int err = 0; void *buf = kmemdup(data, len, GFP_KERNEL); if (!buf) return -ENOMEM; dump_urb("sending", buf, len); - err = usb_bulk_msg(ep->umidi->chip->dev, ep->urb->pipe, buf, len, - NULL, 250); + if (ep->urbs[0].urb) + err = usb_bulk_msg(ep->umidi->chip->dev, ep->urbs[0].urb->pipe, + buf, len, NULL, 250); kfree(buf); return err; } @@ -554,9 +588,9 @@ static void snd_usbmidi_transmit_byte(struct usbmidi_out_port* port, } } -static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_standard_output(struct snd_usb_midi_out_endpoint* ep, + struct urb *urb) { - struct urb* urb = ep->urb; int p; /* FIXME: lower-numbered ports can starve higher-numbered ports */ @@ -613,14 +647,15 @@ static void snd_usbmidi_novation_input(struct snd_usb_midi_in_endpoint* ep, snd_usbmidi_input_data(ep, 0, &buffer[2], buffer[0] - 1); } -static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep, + struct urb *urb) { uint8_t* transfer_buffer; int count; if (!ep->ports[0].active) return; - transfer_buffer = ep->urb->transfer_buffer; + transfer_buffer = urb->transfer_buffer; count = snd_rawmidi_transmit(ep->ports[0].substream, &transfer_buffer[2], ep->max_transfer - 2); @@ -630,7 +665,7 @@ static void snd_usbmidi_novation_output(struct snd_usb_midi_out_endpoint* ep) } transfer_buffer[0] = 0; transfer_buffer[1] = count; - ep->urb->transfer_buffer_length = 2 + count; + urb->transfer_buffer_length = 2 + count; } static struct usb_protocol_ops snd_usbmidi_novation_ops = { @@ -648,20 +683,21 @@ static void snd_usbmidi_raw_input(struct snd_usb_midi_in_endpoint* ep, snd_usbmidi_input_data(ep, 0, buffer, buffer_length); } -static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_raw_output(struct snd_usb_midi_out_endpoint* ep, + struct urb *urb) { int count; if (!ep->ports[0].active) return; count = snd_rawmidi_transmit(ep->ports[0].substream, - ep->urb->transfer_buffer, + urb->transfer_buffer, ep->max_transfer); if (count < 1) { ep->ports[0].active = 0; return; } - ep->urb->transfer_buffer_length = count; + urb->transfer_buffer_length = count; } static struct usb_protocol_ops snd_usbmidi_raw_ops = { @@ -681,23 +717,25 @@ static void snd_usbmidi_us122l_input(struct snd_usb_midi_in_endpoint *ep, snd_usbmidi_input_data(ep, 0, buffer, buffer_length); } -static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep) +static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, + struct urb *urb) { int count; if (!ep->ports[0].active) return; - count = ep->urb->dev->speed == USB_SPEED_HIGH ? 1 : 2; + count = snd_usb_get_speed(ep->umidi->chip->dev) == USB_SPEED_HIGH + ? 1 : 2; count = snd_rawmidi_transmit(ep->ports[0].substream, - ep->urb->transfer_buffer, + urb->transfer_buffer, count); if (count < 1) { ep->ports[0].active = 0; return; } - memset(ep->urb->transfer_buffer + count, 0xFD, 9 - count); - ep->urb->transfer_buffer_length = count; + memset(urb->transfer_buffer + count, 0xFD, 9 - count); + urb->transfer_buffer_length = count; } static struct usb_protocol_ops snd_usbmidi_122l_ops = { @@ -786,10 +824,11 @@ static void snd_usbmidi_emagic_input(struct snd_usb_midi_in_endpoint* ep, } } -static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep, + struct urb *urb) { int port0 = ep->current_port; - uint8_t* buf = ep->urb->transfer_buffer; + uint8_t* buf = urb->transfer_buffer; int buf_free = ep->max_transfer; int length, i; @@ -829,7 +868,7 @@ static void snd_usbmidi_emagic_output(struct snd_usb_midi_out_endpoint* ep) *buf = 0xff; --buf_free; } - ep->urb->transfer_buffer_length = ep->max_transfer - buf_free; + urb->transfer_buffer_length = ep->max_transfer - buf_free; } static struct usb_protocol_ops snd_usbmidi_emagic_ops = { @@ -884,6 +923,35 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, } } +static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) +{ + struct usbmidi_out_port* port = substream->runtime->private_data; + struct snd_usb_midi_out_endpoint *ep = port->ep; + unsigned int drain_urbs; + DEFINE_WAIT(wait); + long timeout = msecs_to_jiffies(50); + + /* + * The substream buffer is empty, but some data might still be in the + * currently active URBs, so we have to wait for those to complete. + */ + spin_lock_irq(&ep->buffer_lock); + drain_urbs = ep->active_urbs; + if (drain_urbs) { + ep->drain_urbs |= drain_urbs; + do { + prepare_to_wait(&ep->drain_wait, &wait, + TASK_UNINTERRUPTIBLE); + spin_unlock_irq(&ep->buffer_lock); + timeout = schedule_timeout(timeout); + spin_lock_irq(&ep->buffer_lock); + drain_urbs &= ep->drain_urbs; + } while (drain_urbs && timeout); + finish_wait(&ep->drain_wait, &wait); + } + spin_unlock_irq(&ep->buffer_lock); +} + static int snd_usbmidi_input_open(struct snd_rawmidi_substream *substream) { return 0; @@ -908,6 +976,7 @@ static struct snd_rawmidi_ops snd_usbmidi_output_ops = { .open = snd_usbmidi_output_open, .close = snd_usbmidi_output_close, .trigger = snd_usbmidi_output_trigger, + .drain = snd_usbmidi_output_drain, }; static struct snd_rawmidi_ops snd_usbmidi_input_ops = { @@ -916,19 +985,26 @@ static struct snd_rawmidi_ops snd_usbmidi_input_ops = { .trigger = snd_usbmidi_input_trigger }; +static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb, + unsigned int buffer_length) +{ + usb_buffer_free(umidi->chip->dev, buffer_length, + urb->transfer_buffer, urb->transfer_dma); + usb_free_urb(urb); +} + /* * Frees an input endpoint. * May be called when ep hasn't been initialized completely. */ static void snd_usbmidi_in_endpoint_delete(struct snd_usb_midi_in_endpoint* ep) { - if (ep->urb) { - usb_buffer_free(ep->umidi->chip->dev, - ep->urb->transfer_buffer_length, - ep->urb->transfer_buffer, - ep->urb->transfer_dma); - usb_free_urb(ep->urb); - } + unsigned int i; + + for (i = 0; i < INPUT_URBS; ++i) + if (ep->urbs[i]) + free_urb_and_buffer(ep->umidi, ep->urbs[i], + ep->urbs[i]->transfer_buffer_length); kfree(ep); } @@ -943,6 +1019,7 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, void* buffer; unsigned int pipe; int length; + unsigned int i; rep->in = NULL; ep = kzalloc(sizeof(*ep), GFP_KERNEL); @@ -950,30 +1027,36 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, return -ENOMEM; ep->umidi = umidi; - ep->urb = usb_alloc_urb(0, GFP_KERNEL); - if (!ep->urb) { - snd_usbmidi_in_endpoint_delete(ep); - return -ENOMEM; + for (i = 0; i < INPUT_URBS; ++i) { + ep->urbs[i] = usb_alloc_urb(0, GFP_KERNEL); + if (!ep->urbs[i]) { + snd_usbmidi_in_endpoint_delete(ep); + return -ENOMEM; + } } if (ep_info->in_interval) pipe = usb_rcvintpipe(umidi->chip->dev, ep_info->in_ep); else pipe = usb_rcvbulkpipe(umidi->chip->dev, ep_info->in_ep); length = usb_maxpacket(umidi->chip->dev, pipe, 0); - buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL, - &ep->urb->transfer_dma); - if (!buffer) { - snd_usbmidi_in_endpoint_delete(ep); - return -ENOMEM; + for (i = 0; i < INPUT_URBS; ++i) { + buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL, + &ep->urbs[i]->transfer_dma); + if (!buffer) { + snd_usbmidi_in_endpoint_delete(ep); + return -ENOMEM; + } + if (ep_info->in_interval) + usb_fill_int_urb(ep->urbs[i], umidi->chip->dev, + pipe, buffer, length, + snd_usbmidi_in_urb_complete, + ep, ep_info->in_interval); + else + usb_fill_bulk_urb(ep->urbs[i], umidi->chip->dev, + pipe, buffer, length, + snd_usbmidi_in_urb_complete, ep); + ep->urbs[i]->transfer_flags = URB_NO_TRANSFER_DMA_MAP; } - if (ep_info->in_interval) - usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer, - length, snd_usbmidi_in_urb_complete, ep, - ep_info->in_interval); - else - usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer, - length, snd_usbmidi_in_urb_complete, ep); - ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; rep->in = ep; return 0; @@ -994,12 +1077,12 @@ static unsigned int snd_usbmidi_count_bits(unsigned int x) */ static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) { - if (ep->urb) { - usb_buffer_free(ep->umidi->chip->dev, ep->max_transfer, - ep->urb->transfer_buffer, - ep->urb->transfer_dma); - usb_free_urb(ep->urb); - } + unsigned int i; + + for (i = 0; i < OUTPUT_URBS; ++i) + if (ep->urbs[i].urb) + free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, + ep->max_transfer); kfree(ep); } @@ -1011,7 +1094,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, struct snd_usb_midi_endpoint* rep) { struct snd_usb_midi_out_endpoint* ep; - int i; + unsigned int i; unsigned int pipe; void* buffer; @@ -1021,38 +1104,46 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, return -ENOMEM; ep->umidi = umidi; - ep->urb = usb_alloc_urb(0, GFP_KERNEL); - if (!ep->urb) { - snd_usbmidi_out_endpoint_delete(ep); - return -ENOMEM; + for (i = 0; i < OUTPUT_URBS; ++i) { + ep->urbs[i].urb = usb_alloc_urb(0, GFP_KERNEL); + if (!ep->urbs[i].urb) { + snd_usbmidi_out_endpoint_delete(ep); + return -ENOMEM; + } + ep->urbs[i].ep = ep; } if (ep_info->out_interval) pipe = usb_sndintpipe(umidi->chip->dev, ep_info->out_ep); else pipe = usb_sndbulkpipe(umidi->chip->dev, ep_info->out_ep); if (umidi->chip->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ - /* FIXME: we need more URBs to get reasonable bandwidth here: */ ep->max_transfer = 4; else ep->max_transfer = usb_maxpacket(umidi->chip->dev, pipe, 1); - buffer = usb_buffer_alloc(umidi->chip->dev, ep->max_transfer, - GFP_KERNEL, &ep->urb->transfer_dma); - if (!buffer) { - snd_usbmidi_out_endpoint_delete(ep); - return -ENOMEM; + for (i = 0; i < OUTPUT_URBS; ++i) { + buffer = usb_buffer_alloc(umidi->chip->dev, + ep->max_transfer, GFP_KERNEL, + &ep->urbs[i].urb->transfer_dma); + if (!buffer) { + snd_usbmidi_out_endpoint_delete(ep); + return -ENOMEM; + } + if (ep_info->out_interval) + usb_fill_int_urb(ep->urbs[i].urb, umidi->chip->dev, + pipe, buffer, ep->max_transfer, + snd_usbmidi_out_urb_complete, + &ep->urbs[i], ep_info->out_interval); + else + usb_fill_bulk_urb(ep->urbs[i].urb, umidi->chip->dev, + pipe, buffer, ep->max_transfer, + snd_usbmidi_out_urb_complete, + &ep->urbs[i]); + ep->urbs[i].urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; } - if (ep_info->out_interval) - usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer, - ep->max_transfer, snd_usbmidi_out_urb_complete, - ep, ep_info->out_interval); - else - usb_fill_bulk_urb(ep->urb, umidi->chip->dev, - pipe, buffer, ep->max_transfer, - snd_usbmidi_out_urb_complete, ep); - ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP; spin_lock_init(&ep->buffer_lock); tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep); + init_waitqueue_head(&ep->drain_wait); for (i = 0; i < 0x10; ++i) if (ep_info->out_cables & (1 << i)) { @@ -1090,7 +1181,7 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi) void snd_usbmidi_disconnect(struct list_head* p) { struct snd_usb_midi* umidi; - int i; + unsigned int i, j; umidi = list_entry(p, struct snd_usb_midi, list); /* @@ -1105,13 +1196,15 @@ void snd_usbmidi_disconnect(struct list_head* p) struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i]; if (ep->out) tasklet_kill(&ep->out->tasklet); - if (ep->out && ep->out->urb) { - usb_kill_urb(ep->out->urb); + if (ep->out) { + for (j = 0; j < OUTPUT_URBS; ++j) + usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); } if (ep->in) - usb_kill_urb(ep->in->urb); + for (j = 0; j < INPUT_URBS; ++j) + usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ if (ep->out) { snd_usbmidi_out_endpoint_delete(ep->out); @@ -1692,20 +1785,25 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, void snd_usbmidi_input_stop(struct list_head* p) { struct snd_usb_midi* umidi; - int i; + unsigned int i, j; umidi = list_entry(p, struct snd_usb_midi, list); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint* ep = &umidi->endpoints[i]; if (ep->in) - usb_kill_urb(ep->in->urb); + for (j = 0; j < INPUT_URBS; ++j) + usb_kill_urb(ep->in->urbs[j]); } } static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) { - if (ep) { - struct urb* urb = ep->urb; + unsigned int i; + + if (!ep) + return; + for (i = 0; i < INPUT_URBS; ++i) { + struct urb* urb = ep->urbs[i]; urb->dev = ep->umidi->chip->dev; snd_usbmidi_submit_urb(urb, GFP_KERNEL); } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ec9cdf986928..ab5a3ac2ac47 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -86,6 +86,7 @@ struct usb_mixer_interface { u8 rc_buffer[6]; u8 audigy2nx_leds[3]; + u8 xonar_u1_status; }; @@ -461,7 +462,7 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) { struct usb_mixer_elem_info *cval = kcontrol->private_data; - DECLARE_TLV_DB_SCALE(scale, 0, 0, 0); + DECLARE_TLV_DB_MINMAX(scale, 0, 0); if (size < sizeof(scale)) return -ENOMEM; @@ -469,7 +470,16 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, * while ALSA TLV contains in 1/100 dB unit */ scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; - scale[3] = (convert_signed_value(cval, cval->res) * 100) / 256; + scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256; + if (scale[3] <= scale[2]) { + /* something is wrong; assume it's either from/to 0dB */ + if (scale[2] < 0) + scale[3] = 0; + else if (scale[2] > 0) + scale[2] = 0; + else /* totally crap, return an error */ + return -EINVAL; + } if (copy_to_user(_tlv, scale, sizeof(scale))) return -EFAULT; return 0; @@ -2033,6 +2043,58 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } +static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + return 0; +} + +static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + u8 old_status, new_status; + int err, changed; + + old_status = mixer->xonar_u1_status; + if (ucontrol->value.integer.value[0]) + new_status = old_status | 0x02; + else + new_status = old_status & ~0x02; + changed = new_status != old_status; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + 50, 0, &new_status, 1, 100); + if (err < 0) + return err; + mixer->xonar_u1_status = new_status; + return changed; +} + +static struct snd_kcontrol_new snd_xonar_u1_output_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Playback Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_xonar_u1_switch_get, + .put = snd_xonar_u1_switch_put, +}; + +static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +{ + int err; + + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + if (err < 0) + return err; + mixer->xonar_u1_status = 0x05; + return 0; +} + int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { @@ -2075,6 +2137,13 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, snd_audigy2nx_proc_read); } + if (mixer->chip->usb_id == USB_ID(0x0b05, 0x1739) || + mixer->chip->usb_id == USB_ID(0x0b05, 0x1743)) { + err = snd_xonar_u1_controls_create(mixer); + if (err < 0) + goto _error; + } + err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops); if (err < 0) goto _error; |