diff options
Diffstat (limited to 'sound')
282 files changed, 15611 insertions, 8699 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 200aca1faa71..1eceb85287c5 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -60,6 +60,8 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" +source "sound/atmel/Kconfig" + source "sound/spi/Kconfig" source "sound/mips/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index c76d70716fa5..ec467decfa79 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -6,7 +6,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - sparc/ spi/ parisc/ pcmcia/ mips/ soc/ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h index ee64f5de8966..6065b0344e23 100644 --- a/sound/aoa/aoa-gpio.h +++ b/sound/aoa/aoa-gpio.h @@ -34,10 +34,12 @@ struct gpio_methods { void (*set_headphone)(struct gpio_runtime *rt, int on); void (*set_speakers)(struct gpio_runtime *rt, int on); void (*set_lineout)(struct gpio_runtime *rt, int on); + void (*set_master)(struct gpio_runtime *rt, int on); int (*get_headphone)(struct gpio_runtime *rt); int (*get_speakers)(struct gpio_runtime *rt); int (*get_lineout)(struct gpio_runtime *rt); + int (*get_master)(struct gpio_runtime *rt); void (*set_hw_reset)(struct gpio_runtime *rt, int on); diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index c93ad5dec66b..de8e03afa97b 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include "../aoa.h" -/* TODO: these are 20 global variables +/* TODO: these are lots of global variables * that aren't used on most machines... * Move them into a dynamically allocated * structure and use that. @@ -23,6 +23,7 @@ /* these are the GPIO numbers (register addresses as offsets into * the GPIO space) */ static int headphone_mute_gpio; +static int master_mute_gpio; static int amp_mute_gpio; static int lineout_mute_gpio; static int hw_reset_gpio; @@ -32,6 +33,7 @@ static int linein_detect_gpio; /* see the SWITCH_GPIO macro */ static int headphone_mute_gpio_activestate; +static int master_mute_gpio_activestate; static int amp_mute_gpio_activestate; static int lineout_mute_gpio_activestate; static int hw_reset_gpio_activestate; @@ -156,6 +158,7 @@ static int ftr_gpio_get_##name(struct gpio_runtime *rt) \ FTR_GPIO(headphone, 0); FTR_GPIO(amp, 1); FTR_GPIO(lineout, 2); +FTR_GPIO(master, 3); static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) { @@ -172,6 +175,8 @@ static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) hw_reset_gpio, v); } +static struct gpio_methods methods; + static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) { int saved; @@ -181,6 +186,8 @@ static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) ftr_gpio_set_headphone(rt, 0); ftr_gpio_set_amp(rt, 0); ftr_gpio_set_lineout(rt, 0); + if (methods.set_master) + ftr_gpio_set_master(rt, 0); rt->implementation_private = saved; } @@ -193,6 +200,8 @@ static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt) ftr_gpio_set_headphone(rt, (s>>0)&1); ftr_gpio_set_amp(rt, (s>>1)&1); ftr_gpio_set_lineout(rt, (s>>2)&1); + if (methods.set_master) + ftr_gpio_set_master(rt, (s>>3)&1); } static void ftr_handle_notify(struct work_struct *work) @@ -231,6 +240,12 @@ static void ftr_gpio_init(struct gpio_runtime *rt) get_gpio("hw-reset", "audio-hw-reset", &hw_reset_gpio, &hw_reset_gpio_activestate); + if (get_gpio("master-mute", NULL, + &master_mute_gpio, + &master_mute_gpio_activestate)) { + methods.set_master = ftr_gpio_set_master; + methods.get_master = ftr_gpio_get_master; + } headphone_detect_node = get_gpio("headphone-detect", NULL, &headphone_detect_gpio, diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index ad60f5d10e82..fbf5c933baa4 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1,16 +1,14 @@ /* - * Apple Onboard Audio driver -- layout fabric + * Apple Onboard Audio driver -- layout/machine id fabric * - * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net> * * GPL v2, can be found in COPYING. * * - * This fabric module looks for sound codecs - * based on the layout-id property in the device tree. - * + * This fabric module looks for sound codecs based on the + * layout-id or device-id property in the device tree. */ - #include <asm/prom.h> #include <linux/list.h> #include <linux/module.h> @@ -63,7 +61,7 @@ struct codec_connect_info { #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) struct layout { - unsigned int layout_id; + unsigned int layout_id, device_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; @@ -111,6 +109,10 @@ MODULE_ALIAS("sound-layout-96"); MODULE_ALIAS("sound-layout-98"); MODULE_ALIAS("sound-layout-100"); +MODULE_ALIAS("aoa-device-id-14"); +MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-35"); + /* onyx with all but microphone connected */ static struct codec_connection onyx_connections_nomic[] = { { @@ -518,6 +520,27 @@ static struct layout layouts[] = { .connections = onyx_connections_noheadphones, }, }, + /* PowerMac3,4 */ + { .device_id = 14, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerMac3,6 */ + { .device_id = 22, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook5,2 */ + { .device_id = 35, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, {} }; @@ -526,7 +549,7 @@ static struct layout *find_layout_by_id(unsigned int id) struct layout *l; l = layouts; - while (l->layout_id) { + while (l->codecs[0].name) { if (l->layout_id == id) return l; l++; @@ -534,6 +557,19 @@ static struct layout *find_layout_by_id(unsigned int id) return NULL; } +static struct layout *find_layout_by_device(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->device_id == id) + return l; + l++; + } + return NULL; +} + static void use_layout(struct layout *l) { int i; @@ -564,6 +600,7 @@ struct layout_dev { struct snd_kcontrol *headphone_ctrl; struct snd_kcontrol *lineout_ctrl; struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *master_ctrl; struct snd_kcontrol *headphone_detected_ctrl; struct snd_kcontrol *lineout_detected_ctrl; @@ -615,6 +652,7 @@ static struct snd_kcontrol_new n##_ctl = { \ AMP_CONTROL(headphone, "Headphone Switch"); AMP_CONTROL(speakers, "Speakers Switch"); AMP_CONTROL(lineout, "Line-Out Switch"); +AMP_CONTROL(master, "Master Switch"); static int detect_choice_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -855,6 +893,11 @@ static void layout_attached_codec(struct aoa_codec *codec) lineout = codec->gpio->methods->get_detect(codec->gpio, AOA_NOTIFY_LINE_OUT); + if (codec->gpio->methods->set_master) { + ctl = snd_ctl_new1(&master_ctl, codec->gpio); + ldev->master_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } while (cc->connected) { if (cc->connected & CC_SPEAKERS) { if (headphones <= 0 && lineout <= 0) @@ -938,8 +981,8 @@ static struct aoa_fabric layout_fabric = { static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) { struct device_node *sound = NULL; - const unsigned int *layout_id; - struct layout *layout; + const unsigned int *id; + struct layout *layout = NULL; struct layout_dev *ldev = NULL; int err; @@ -952,15 +995,18 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) if (sound->type && strcasecmp(sound->type, "soundchip") == 0) break; } - if (!sound) return -ENODEV; + if (!sound) + return -ENODEV; - layout_id = of_get_property(sound, "layout-id", NULL); - if (!layout_id) - goto outnodev; - printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n", - *layout_id); + id = of_get_property(sound, "layout-id", NULL); + if (id) { + layout = find_layout_by_id(*id); + } else { + id = of_get_property(sound, "device-id", NULL); + if (id) + layout = find_layout_by_device(*id); + } - layout = find_layout_by_id(*layout_id); if (!layout) { printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); goto outnodev; @@ -976,6 +1022,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) ldev->layout = layout; ldev->gpio.node = sound->parent; switch (layout->layout_id) { + case 0: /* anything with device_id, not layout_id */ case 41: /* that unknown machine no one seems to have */ case 51: /* PowerBook5,4 */ case 58: /* Mac Mini */ diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index be468edf3ecb..418c84c99d69 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -1,7 +1,7 @@ /* * i2sbus driver * - * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net> * * GPL v2, can be found in COPYING. */ @@ -186,13 +186,25 @@ static int i2sbus_add_dev(struct macio_dev *macio, } } if (i == 1) { - const u32 *layout_id = - of_get_property(sound, "layout-id", NULL); - if (layout_id) { - layout = *layout_id; + const u32 *id = of_get_property(sound, "layout-id", NULL); + + if (id) { + layout = *id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); ok = 1; + } else { + id = of_get_property(sound, "device-id", NULL); + /* + * We probably cannot handle all device-id machines, + * so restrict to those we do handle for now. + */ + if (id && (*id == 22 || *id == 14 || *id == 35)) { + snprintf(dev->sound.modalias, 32, + "aoa-device-id-%d", *id); + ok = 1; + layout = -1; + } } } /* for the time being, until we can handle non-layout-id diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de48d816..885683a3b0bd 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -11,17 +11,6 @@ menuconfig SND_ARM if SND_ARM -config SND_SA11XX_UDA1341 - tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && L3 - select SND_PCM - help - Say Y here if you have a Compaq iPaq H3x00 handheld computer - and want to use its Philips UDA 1341 audio chip. - - To compile this driver as a module, choose M here: the module - will be called snd-sa11xx-uda1341. - config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de11de8a..5a549ed6c8aa 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -2,9 +2,6 @@ # Makefile for ALSA # -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o -snd-sa11xx-uda1341-objs := sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o devdma.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7d39aac9ec14..7fbd68fab944 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--); + } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout); if (!timeout) dev_err(&aaci->dev->dev, @@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & SLFR_1TXB) && timeout--); + } while ((v & SLFR_1TXB) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); @@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) do { cond_resched(); v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); - } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--); + } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on RX valid\n"); diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c33be5..2e6355f4cbb9 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -31,6 +31,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; static struct clk *ac97_clk; static struct clk *ac97conf_clk; +static int reset_gpio; /* * Beware PXA27x bugs: @@ -42,6 +43,45 @@ static struct clk *ac97conf_clk; * 1 jiffy timeout if interrupt never comes). */ +enum { + RESETGPIO_FORCE_HIGH, + RESETGPIO_FORCE_LOW, + RESETGPIO_NORMAL_ALTFUNC +}; + +/** + * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA + * @mode: chosen action + * + * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line + * must be done to insure proper work of AC97 reset line. This function + * computes the correct gpio_mode for further use by reset functions, and + * applied the change through pxa_gpio_mode. + */ +static void set_resetgpio_mode(int resetgpio_action) +{ + int mode = 0; + + if (reset_gpio) + switch (resetgpio_action) { + case RESETGPIO_NORMAL_ALTFUNC: + if (reset_gpio == 113) + mode = 113 | GPIO_OUT | GPIO_DFLT_LOW; + if (reset_gpio == 95) + mode = 95 | GPIO_ALT_FN_1_OUT; + break; + case RESETGPIO_FORCE_LOW: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW; + break; + case RESETGPIO_FORCE_HIGH: + mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH; + break; + }; + + if (mode) + pxa_gpio_mode(mode); +} + unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { unsigned short val = -1; @@ -137,10 +177,10 @@ static inline void pxa_ac97_warm_pxa27x(void) /* warm reset broken on Bulverde, so manually keep AC97 reset high */ - pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); + set_resetgpio_mode(RESETGPIO_FORCE_HIGH); udelay(10); GCR |= GCR_WARM_RST; - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); udelay(500); } @@ -308,8 +348,8 @@ int pxa2xx_ac97_hw_resume(void) pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + /* Use GPIO 113 or 95 as AC97 Reset on Bulverde */ + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); } clk_enable(ac97_clk); return 0; @@ -320,6 +360,27 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume); int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) { int ret; + struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data; + + if (pdata) { + switch (pdata->reset_gpio) { + case 95: + case 113: + reset_gpio = pdata->reset_gpio; + break; + case 0: + reset_gpio = 113; + break; + case -1: + break; + default: + dev_err(&dev->dev, "Invalid reset GPIO %d\n", + pdata->reset_gpio); + } + } else { + if (cpu_is_pxa27x()) + reset_gpio = 113; + } if (cpu_is_pxa25x() || cpu_is_pxa27x()) { pxa_gpio_mode(GPIO31_SYNC_AC97_MD); @@ -330,7 +391,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (cpu_is_pxa27x()) { /* Use GPIO 113 as AC97 Reset on Bulverde */ - pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC); ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c deleted file mode 100644 index 51d708c31e65..000000000000 --- a/sound/arm/sa11xx-uda1341.c +++ /dev/null @@ -1,984 +0,0 @@ -/* - * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard - * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS - * 2002-03-20 Tomas Kasparek playback over ALSA is working - * 2002-03-28 Tomas Kasparek playback over OSS emulation is working - * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA) - * 2002-03-29 Tomas Kasparek capture is working (OSS emulation) - * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates) - * 2003-02-14 Brian Avery fixed full duplex mode, other updates - * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL) - * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel - * working suspend and resume - * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again - * merged HAL layer (patches from Brian) - */ - -/*************************************************************************************************** -* -* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai -* available in the Alsa doc section on the website -* -* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100. -* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated -* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it. -* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the -* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which -* is a mem loc that always decodes to 0's w/ no off chip access. -* -* Some alsa terminology: -* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes -* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte -* buffer and 4 periods in the runtime structure this means we'll get an int every 256 -* bytes or 4 times per buffer. -* A number of the sizes are in frames rather than bytes, use frames_to_bytes and -* bytes_to_frames to convert. The easiest way to tell the units is to look at the -* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t -* -* Notes about the pointer fxn: -* The pointer fxn needs to return the offset into the dma buffer in frames. -* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts. -* -* Notes about pause/resume -* Implementing this would be complicated so it's skipped. The problem case is: -* A full duplex connection is going, then play is paused. At this point you need to start xmitting -* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd -* need to save off the dma info, and restore it properly on a resume. Yeach! -* -* Notes about transfer methods: -* The async write calls fail. I probably need to implement something else to support them? -* -***************************************************************************************************/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/err.h> -#include <linux/platform_device.h> -#include <linux/errno.h> -#include <linux/ioctl.h> -#include <linux/delay.h> -#include <linux/slab.h> - -#ifdef CONFIG_PM -#include <linux/pm.h> -#endif - -#include <mach/hardware.h> -#include <mach/h3600.h> -#include <asm/mach-types.h> -#include <asm/dma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/initval.h> - -#include <linux/l3/l3.h> - -#undef DEBUG_MODE -#undef DEBUG_FUNCTION_NAMES -#include <sound/uda1341.h> - -/* - * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? - * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this - * module for Familiar 0.6.1 - */ - -/* {{{ Type definitions */ - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); - -static char *id; /* ID for this card */ - -module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); - -struct audio_stream { - char *id; /* identification string */ - int stream_id; /* numeric identification */ - dma_device_t dma_dev; /* device identifier for DMA */ -#ifdef HH_VERSION - dmach_t dmach; /* dma channel identification */ -#else - dma_regs_t *dma_regs; /* points to our DMA registers */ -#endif - unsigned int active:1; /* we are using this stream for transfer now */ - int period; /* current transfer period */ - int periods; /* current count of periods registerd in the DMA engine */ - int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ - unsigned int old_offset; - spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ - struct snd_pcm_substream *stream; -}; - -struct sa11xx_uda1341 { - struct snd_card *card; - struct l3_client *uda1341; - struct snd_pcm *pcm; - long samplerate; - struct audio_stream s[2]; /* playback & capture */ -}; - -static unsigned int rates[] = { - 8000, 10666, 10985, 14647, - 16000, 21970, 22050, 24000, - 29400, 32000, 44100, 48000, -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct platform_device *device; - -/* }}} */ - -/* {{{ Clock and sample rate stuff */ - -/* - * Stop-gap solution until rest of hh.org HAL stuff is merged. - */ -#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) -#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) - -#ifdef CONFIG_SA1100_H3XXX -#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) -#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) -#else -#error This driver could serve H3x00 handhelds only! -#endif - -static void sa11xx_uda1341_set_audio_clock(long val) -{ - switch (val) { - case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ - GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - - case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ - GPSR = GPIO_H3600_CLK_SET0; - GPCR = GPIO_H3600_CLK_SET1; - break; - - case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ - GPCR = GPIO_H3600_CLK_SET0; - GPSR = GPIO_H3600_CLK_SET1; - break; - - case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ - GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - } -} - -static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate) -{ - int clk_div = 0; - int clk=0; - - /* We don't want to mess with clocks when frames are in flight */ - Ser4SSCR0 &= ~SSCR0_SSE; - /* wait for any frame to complete */ - udelay(125); - - /* - * We have the following clock sources: - * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz - * Those can be divided either by 256, 384 or 512. - * This makes up 12 combinations for the following samplerates... - */ - if (rate >= 48000) - rate = 48000; - else if (rate >= 44100) - rate = 44100; - else if (rate >= 32000) - rate = 32000; - else if (rate >= 29400) - rate = 29400; - else if (rate >= 24000) - rate = 24000; - else if (rate >= 22050) - rate = 22050; - else if (rate >= 21970) - rate = 21970; - else if (rate >= 16000) - rate = 16000; - else if (rate >= 14647) - rate = 14647; - else if (rate >= 10985) - rate = 10985; - else if (rate >= 10666) - rate = 10666; - else - rate = 8000; - - /* Set the external clock generator */ - - sa11xx_uda1341_set_audio_clock(rate); - - /* Select the clock divisor */ - switch (rate) { - case 8000: - case 10985: - case 22050: - case 24000: - clk = F512; - clk_div = SSCR0_SerClkDiv(16); - break; - case 16000: - case 21970: - case 44100: - case 48000: - clk = F256; - clk_div = SSCR0_SerClkDiv(8); - break; - case 10666: - case 14647: - case 29400: - case 32000: - clk = F384; - clk_div = SSCR0_SerClkDiv(12); - break; - } - - /* FMT setting should be moved away when other FMTs are added (FIXME) */ - l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); - - l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); - Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; - sa11xx_uda1341->samplerate = rate; -} - -/* }}} */ - -/* {{{ HW init and shutdown */ - -static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - unsigned long flags; - - /* Setup DMA stuff */ - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; - - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; - - /* Initialize the UDA1341 internal state */ - - /* Setup the uarts */ - local_irq_save(flags); - GAFR |= (GPIO_SSP_CLK); - GPDR &= ~(GPIO_SSP_CLK); - Ser4SSCR0 = 0; - Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); - Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; - Ser4SSCR0 |= SSCR0_SSE; - local_irq_restore(flags); - - /* Enable the audio power */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* Wait for the UDA1341 to wake up */ - mdelay(1); //FIXME - was removed by Perex - Why? - - /* Initialize the UDA1341 internal state */ - l3_open(sa11xx_uda1341->uda1341); - - /* external clock configuration (after l3_open - regs must be initialized */ - sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); - - /* Wait for the UDA1341 to wake up */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - mdelay(1); - - /* make the left and right channels unswapped (flip the WS latch) */ - Ser4SSDR = 0; - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - /* mute on */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* disable the audio power and all signals leading to the audio chip */ - l3_close(sa11xx_uda1341->uda1341); - Ser4SSCR0 = 0; - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - - /* power off and mute off */ - /* FIXME - is muting off necesary??? */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -/* }}} */ - -/* {{{ DMA staff */ - -/* - * these are the address and sizes used to fill the xmit buffer - * so we can get a clock in record only mode - */ -#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS -#define FORCE_CLOCK_SIZE 4096 // was 2048 - -// FIXME Why this value exactly - wrote comment -#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ - -#ifdef HH_VERSION - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int)) -{ - int ret; - - ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); - if (ret < 0) { - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; - } - sa1100_dma_set_callback(s->dmach, callback); - return 0; -} - -static inline void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dmach); - s->dmach = -1; -} - -#else - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *)) -{ - int ret; - - ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); - if (ret < 0) - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; -} - -static void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dma_regs); - s->dma_regs = 0; -} - -#endif - -static u_int audio_get_dma_pos(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int offset; - unsigned long flags; - dma_addr_t addr; - - // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel - spin_lock_irqsave(&s->dma_lock, flags); -#ifdef HH_VERSION - sa1100_dma_get_current(s->dmach, NULL, &addr); -#else - addr = sa1100_get_dma_pos((s)->dma_regs); -#endif - offset = addr - runtime->dma_addr; - spin_unlock_irqrestore(&s->dma_lock, flags); - - offset = bytes_to_frames(runtime,offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -/* - * this stops the dma and clears the dma ptrs - */ -static void audio_stop_dma(struct audio_stream *s) -{ - unsigned long flags; - - spin_lock_irqsave(&s->dma_lock, flags); - s->active = 0; - s->period = 0; - /* this stops the dma channel and clears the buffer ptrs */ -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - sa1100_clear_dma(s->dma_regs); -#endif - spin_unlock_irqrestore(&s->dma_lock, flags); -} - -static void audio_process_dma(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime; - unsigned int dma_size; - unsigned int offset; - int ret; - - /* we are requested to process synchronization DMA transfer */ - if (s->tx_spin) { - if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK)) - return; - /* fill the xmit dma buffers and return */ -#ifdef HH_VERSION - sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); -#else - while (1) { - ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); - if (ret) - return; - } -#endif - return; - } - - /* must be set here - only valid for running streams, not for forced_clock dma fills */ - runtime = substream->runtime; - while (s->active && s->periods < runtime->periods) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - if (s->old_offset) { - /* a little trick, we need resume from old position */ - offset = frames_to_bytes(runtime, s->old_offset - 1); - s->old_offset = 0; - s->periods = 0; - s->period = offset / dma_size; - offset %= dma_size; - dma_size = dma_size - offset; - if (!dma_size) - continue; /* special case */ - } else { - offset = dma_size * s->period; - snd_BUG_ON(dma_size > DMA_BUF_SIZE); - } -#ifdef HH_VERSION - ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size); - if (ret) - return; //FIXME -#else - ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size); - if (ret) { - printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret); - return; - } -#endif - - s->period++; - s->period %= runtime->periods; - s->periods++; - } -} - -#ifdef HH_VERSION -static void audio_dma_callback(void *data, int size) -#else -static void audio_dma_callback(void *data) -#endif -{ - struct audio_stream *s = data; - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - spin_lock(&s->dma_lock); - if (!s->tx_spin && s->periods > 0) - s->periods--; - audio_process_dma(s); - spin_unlock(&s->dma_lock); -} - -/* }}} */ - -/* {{{ PCM setting */ - -/* {{{ trigger & timer */ - -static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - int stream_id = substream->pstr->stream; - struct audio_stream *s = &chip->s[stream_id]; - struct audio_stream *s1 = &chip->s[stream_id ^ 1]; - int err = 0; - - /* note local interrupts are already disabled in the midlevel code */ - spin_lock(&s->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* now we need to make sure a record only stream has a clock */ - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s1->tx_spin = 1; - audio_process_dma(s1); - } - /* this case is when you were recording then you turn on a - * playback stream so we stop (also clears it) the dma first, - * clear the sync flag and then we let it turned on - */ - else { - s->tx_spin = 0; - } - - /* requested stream startup */ - s->active = 1; - audio_process_dma(s); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - audio_stop_dma(s); - - /* - * now we need to make sure a record only stream has a clock - * so if we're stopping a playback with an active capture - * we need to turn the 0 fill dma on for the xmit side - */ - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s->tx_spin = 1; - audio_process_dma(s); - } - /* - * we killed a capture only stream, so we should also kill - * the zero fill transmit - */ - else { - if (s1->tx_spin) { - s1->tx_spin = 0; - audio_stop_dma(s1); - } - } - - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - s->active = 0; -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - s->periods = 0; - break; - case SNDRV_PCM_TRIGGER_RESUME: - s->active = 1; - s->tx_spin = 0; - audio_process_dma(s); - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->active = 0; - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) { - if (s1->active) { - s->tx_spin = 1; - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - audio_process_dma(s); - } - } else { - if (s1->tx_spin) { - s1->tx_spin = 0; -#ifdef HH_VERSION - sa1100_dma_flush_all(s1->dmach); -#else - //FIXME - DMA API -#endif - } - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - s->active = 1; - if (s->old_offset) { - s->tx_spin = 0; - audio_process_dma(s); - break; - } - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } -#ifdef HH_VERSION - sa1100_dma_resume(s->dmach); -#else - //FIXME - DMA API -#endif - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&s->dma_lock); - return err; -} - -static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audio_stream *s = &chip->s[substream->pstr->stream]; - - /* set requested samplerate */ - sa11xx_uda1341_set_samplerate(chip, runtime->rate); - - /* set requestd format when available */ - /* set FMT here !!! FIXME */ - - s->period = 0; - s->periods = 0; - - return 0; -} - -static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - return audio_get_dma_pos(&chip->s[substream->pstr->stream]); -} - -/* }}} */ - -static struct snd_pcm_hardware snd_sa11xx_uda1341_capture = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_sa11xx_uda1341_playback = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int stream_id = substream->pstr->stream; - int err; - - chip->s[stream_id].stream = substream; - - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = snd_sa11xx_uda1341_playback; - else - runtime->hw = snd_sa11xx_uda1341_capture; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) - return err; - - return 0; -} - -static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - - chip->s[substream->pstr->stream].stream = NULL; - return 0; -} - -/* {{{ HW params & free */ - -static int snd_sa11xx_uda1341_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 snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -/* }}} */ - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0) - return err; - - /* - * this sets up our initial buffers and sets the dma_type to isa. - * isa works but I'm not sure why (or if) it's the right choice - * this may be too large, trying it for now - */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, 64*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops); - pcm->private_data = sa11xx_uda1341; - pcm->info_flags = 0; - strcpy(pcm->name, "UDA1341 PCM"); - - sa11xx_uda1341_audio_init(sa11xx_uda1341); - - /* setup DMA controller */ - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback); - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback); - - sa11xx_uda1341->pcm = pcm; - - return 0; -} - -/* }}} */ - -/* {{{ module init & exit */ - -#ifdef CONFIG_PM - -static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr, - pm_message_t state) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); -#ifdef HH_VERSION - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - l3_command(chip->uda1341, CMD_SUSPEND, NULL); - sa11xx_uda1341_audio_shutdown(chip); - - return 0; -} - -static int snd_sa11xx_uda1341_resume(struct platform_device *devptr) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - sa11xx_uda1341_audio_init(chip); - l3_command(chip->uda1341, CMD_RESUME, NULL); -#ifdef HH_VERSION - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif /* COMFIG_PM */ - -void snd_sa11xx_uda1341_free(struct snd_card *card) -{ - struct sa11xx_uda1341 *chip = card->private_data; - - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); -} - -static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) -{ - int err; - struct snd_card *card; - struct sa11xx_uda1341 *chip; - - /* register the soundcard */ - err = snd_card_create(-1, id, THIS_MODULE, - sizeof(struct sa11xx_uda1341), &card); - if (err < 0) - return err; - - chip = card->private_data; - spin_lock_init(&chip->s[0].dma_lock); - spin_lock_init(&chip->s[1].dma_lock); - - card->private_free = snd_sa11xx_uda1341_free; - chip->card = card; - chip->samplerate = AUDIO_RATE_DEFAULT; - - // mixer - if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341))) - goto nodev; - - // PCM - if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0) - goto nodev; - - strcpy(card->driver, "UDA1341"); - strcpy(card->shortname, "H3600 UDA1341TS"); - sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); - - snd_card_set_dev(card, &devptr->dev); - - if ((err = snd_card_register(card)) == 0) { - printk( KERN_INFO "iPAQ audio support initialized\n" ); - platform_set_drvdata(devptr, card); - return 0; - } - - nodev: - snd_card_free(card); - return err; -} - -static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr) -{ - snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; -} - -#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341" - -static struct platform_driver sa11xx_uda1341_driver = { - .probe = sa11xx_uda1341_probe, - .remove = __devexit_p(sa11xx_uda1341_remove), -#ifdef CONFIG_PM - .suspend = snd_sa11xx_uda1341_suspend, - .resume = snd_sa11xx_uda1341_resume, -#endif - .driver = { - .name = SA11XX_UDA1341_DRIVER, - }, -}; - -static int __init sa11xx_uda1341_init(void) -{ - int err; - - if (!machine_is_h3xxx()) - return -ENODEV; - if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) - return err; - device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (!IS_ERR(device)) { - if (platform_get_drvdata(device)) - return 0; - platform_device_unregister(device); - err = -ENODEV; - } else - err = PTR_ERR(device); - platform_driver_unregister(&sa11xx_uda1341_driver); - return err; -} - -static void __exit sa11xx_uda1341_exit(void) -{ - platform_device_unregister(device); - platform_driver_unregister(&sa11xx_uda1341_driver); -} - -module_init(sa11xx_uda1341_init); -module_exit(sa11xx_uda1341_exit); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig new file mode 100644 index 000000000000..6c228a91940d --- /dev/null +++ b/sound/atmel/Kconfig @@ -0,0 +1,19 @@ +menu "Atmel devices (AVR32 and AT91)" + depends on AVR32 || ARCH_AT91 + +config SND_ATMEL_ABDAC + tristate "Atmel Audio Bitstream DAC (ABDAC) driver" + select SND_PCM + depends on DW_DMAC && AVR32 + help + ALSA sound driver for the Atmel Audio Bitstream DAC (ABDAC). + +config SND_ATMEL_AC97C + tristate "Atmel AC97 Controller (AC97C) driver" + select SND_PCM + select SND_AC97_CODEC + depends on DW_DMAC && AVR32 + help + ALSA sound driver for the Atmel AC97 controller. + +endmenu diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile new file mode 100644 index 000000000000..219dcfac6086 --- /dev/null +++ b/sound/atmel/Makefile @@ -0,0 +1,5 @@ +snd-atmel-abdac-objs := abdac.o +snd-atmel-ac97c-objs := ac97c.o + +obj-$(CONFIG_SND_ATMEL_ABDAC) += snd-atmel-abdac.o +obj-$(CONFIG_SND_ATMEL_AC97C) += snd-atmel-ac97c.o diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c new file mode 100644 index 000000000000..28b3c7f7cfe6 --- /dev/null +++ b/sound/atmel/abdac.c @@ -0,0 +1,602 @@ +/* + * Driver for the Atmel on-chip Audio Bitstream DAC (ABDAC) + * + * Copyright (C) 2006-2009 Atmel Corporation + * + * 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/clk.h> +#include <linux/bitmap.h> +#include <linux/dw_dmac.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/atmel-abdac.h> + +/* DAC register offsets */ +#define DAC_DATA 0x0000 +#define DAC_CTRL 0x0008 +#define DAC_INT_MASK 0x000c +#define DAC_INT_EN 0x0010 +#define DAC_INT_DIS 0x0014 +#define DAC_INT_CLR 0x0018 +#define DAC_INT_STATUS 0x001c + +/* Bitfields in CTRL */ +#define DAC_SWAP_OFFSET 30 +#define DAC_SWAP_SIZE 1 +#define DAC_EN_OFFSET 31 +#define DAC_EN_SIZE 1 + +/* Bitfields in INT_MASK/INT_EN/INT_DIS/INT_STATUS/INT_CLR */ +#define DAC_UNDERRUN_OFFSET 28 +#define DAC_UNDERRUN_SIZE 1 +#define DAC_TX_READY_OFFSET 29 +#define DAC_TX_READY_SIZE 1 + +/* Bit manipulation macros */ +#define DAC_BIT(name) \ + (1 << DAC_##name##_OFFSET) +#define DAC_BF(name, value) \ + (((value) & ((1 << DAC_##name##_SIZE) - 1)) \ + << DAC_##name##_OFFSET) +#define DAC_BFEXT(name, value) \ + (((value) >> DAC_##name##_OFFSET) \ + & ((1 << DAC_##name##_SIZE) - 1)) +#define DAC_BFINS(name, value, old) \ + (((old) & ~(((1 << DAC_##name##_SIZE) - 1) \ + << DAC_##name##_OFFSET)) \ + | DAC_BF(name, value)) + +/* Register access macros */ +#define dac_readl(port, reg) \ + __raw_readl((port)->regs + DAC_##reg) +#define dac_writel(port, reg, value) \ + __raw_writel((value), (port)->regs + DAC_##reg) + +/* + * ABDAC supports a maximum of 6 different rates from a generic clock. The + * generic clock has a power of two divider, which gives 6 steps from 192 kHz + * to 5112 Hz. + */ +#define MAX_NUM_RATES 6 +/* ALSA seems to use rates between 192000 Hz and 5112 Hz. */ +#define RATE_MAX 192000 +#define RATE_MIN 5112 + +enum { + DMA_READY = 0, +}; + +struct atmel_abdac_dma { + struct dma_chan *chan; + struct dw_cyclic_desc *cdesc; +}; + +struct atmel_abdac { + struct clk *pclk; + struct clk *sample_clk; + struct platform_device *pdev; + struct atmel_abdac_dma dma; + + struct snd_pcm_hw_constraint_list constraints_rates; + struct snd_pcm_substream *substream; + struct snd_card *card; + struct snd_pcm *pcm; + + void __iomem *regs; + unsigned long flags; + unsigned int rates[MAX_NUM_RATES]; + unsigned int rates_num; + int irq; +}; + +#define get_dac(card) ((struct atmel_abdac *)(card)->private_data) + +/* This function is called by the DMA driver. */ +static void atmel_abdac_dma_period_done(void *arg) +{ + struct atmel_abdac *dac = arg; + snd_pcm_period_elapsed(dac->substream); +} + +static int atmel_abdac_prepare_dma(struct atmel_abdac *dac, + struct snd_pcm_substream *substream, + enum dma_data_direction direction) +{ + struct dma_chan *chan = dac->dma.chan; + struct dw_cyclic_desc *cdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long buffer_len, period_len; + + /* + * We don't do DMA on "complex" transfers, i.e. with + * non-halfword-aligned buffers or lengths. + */ + if (runtime->dma_addr & 1 || runtime->buffer_size & 1) { + dev_dbg(&dac->pdev->dev, "too complex transfer\n"); + return -EINVAL; + } + + buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + period_len = frames_to_bytes(runtime, runtime->period_size); + + cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, + period_len, DMA_TO_DEVICE); + if (IS_ERR(cdesc)) { + dev_dbg(&dac->pdev->dev, "could not prepare cyclic DMA\n"); + return PTR_ERR(cdesc); + } + + cdesc->period_callback = atmel_abdac_dma_period_done; + cdesc->period_callback_param = dac; + + dac->dma.cdesc = cdesc; + + set_bit(DMA_READY, &dac->flags); + + return 0; +} + +static struct snd_pcm_hardware atmel_abdac_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_BE), + .rates = (SNDRV_PCM_RATE_KNOT), + .rate_min = RATE_MIN, + .rate_max = RATE_MAX, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 4096, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 4, + .periods_max = 64, +}; + +static int atmel_abdac_open(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + + dac->substream = substream; + atmel_abdac_hw.rate_max = dac->rates[dac->rates_num - 1]; + atmel_abdac_hw.rate_min = dac->rates[0]; + substream->runtime->hw = atmel_abdac_hw; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &dac->constraints_rates); +} + +static int atmel_abdac_close(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + dac->substream = NULL; + return 0; +} + +static int atmel_abdac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_free(dac->dma.chan); + + return retval; +} + +static int atmel_abdac_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_free(dac->dma.chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_abdac_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval; + + retval = clk_set_rate(dac->sample_clk, 256 * substream->runtime->rate); + if (retval) + return retval; + + if (!test_bit(DMA_READY, &dac->flags)) + retval = atmel_abdac_prepare_dma(dac, substream, DMA_TO_DEVICE); + + return retval; +} + +static int atmel_abdac_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + clk_enable(dac->sample_clk); + retval = dw_dma_cyclic_start(dac->dma.chan); + if (retval) + goto out; + dac_writel(dac, CTRL, DAC_BIT(EN)); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(dac->dma.chan); + dac_writel(dac, DATA, 0); + dac_writel(dac, CTRL, 0); + clk_disable(dac->sample_clk); + break; + default: + retval = -EINVAL; + break; + } +out: + return retval; +} + +static snd_pcm_uframes_t +atmel_abdac_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_src_addr(dac->dma.chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + + return frames; +} + +static irqreturn_t abdac_interrupt(int irq, void *dev_id) +{ + struct atmel_abdac *dac = dev_id; + u32 status; + + status = dac_readl(dac, INT_STATUS); + if (status & DAC_BIT(UNDERRUN)) { + dev_err(&dac->pdev->dev, "underrun detected\n"); + dac_writel(dac, INT_CLR, DAC_BIT(UNDERRUN)); + } else { + dev_err(&dac->pdev->dev, "spurious interrupt (status=0x%x)\n", + status); + dac_writel(dac, INT_CLR, status); + } + + return IRQ_HANDLED; +} + +static struct snd_pcm_ops atmel_abdac_ops = { + .open = atmel_abdac_open, + .close = atmel_abdac_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_abdac_hw_params, + .hw_free = atmel_abdac_hw_free, + .prepare = atmel_abdac_prepare, + .trigger = atmel_abdac_trigger, + .pointer = atmel_abdac_pointer, +}; + +static int __devinit atmel_abdac_pcm_new(struct atmel_abdac *dac) +{ + struct snd_pcm_hardware hw = atmel_abdac_hw; + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(dac->card, dac->card->shortname, + dac->pdev->id, 1, 0, &pcm); + if (retval) + return retval; + + strcpy(pcm->name, dac->card->shortname); + pcm->private_data = dac; + pcm->info_flags = 0; + dac->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &atmel_abdac_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &dac->pdev->dev, hw.periods_min * hw.period_bytes_min, + hw.buffer_bytes_max); + + return retval; +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct dw_dma_slave *dws = slave; + + if (dws->dma_dev == chan->device->dev) { + chan->private = dws; + return true; + } else + return false; +} + +static int set_sample_rates(struct atmel_abdac *dac) +{ + long new_rate = RATE_MAX; + int retval = -EINVAL; + int index = 0; + + /* we start at 192 kHz and work our way down to 5112 Hz */ + while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { + new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); + if (new_rate < 0) + break; + /* make sure we are below the ABDAC clock */ + if (new_rate <= clk_get_rate(dac->pclk)) { + dac->rates[index] = new_rate / 256; + index++; + } + /* divide by 256 and then by two to get next rate */ + new_rate /= 256 * 2; + } + + if (index) { + int i; + + /* reverse array, smallest go first */ + for (i = 0; i < (index / 2); i++) { + unsigned int tmp = dac->rates[index - 1 - i]; + dac->rates[index - 1 - i] = dac->rates[i]; + dac->rates[i] = tmp; + } + + dac->constraints_rates.count = index; + dac->constraints_rates.list = dac->rates; + dac->constraints_rates.mask = 0; + dac->rates_num = index; + + retval = 0; + } + + return retval; +} + +static int __devinit atmel_abdac_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct atmel_abdac *dac; + struct resource *regs; + struct atmel_abdac_pdata *pdata; + struct clk *pclk; + struct clk *sample_clk; + int retval; + int irq; + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) { + dev_dbg(&pdev->dev, "no memory resource\n"); + return -ENXIO; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_dbg(&pdev->dev, "could not get IRQ number\n"); + return irq; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_dbg(&pdev->dev, "no platform data\n"); + return -ENXIO; + } + + pclk = clk_get(&pdev->dev, "pclk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no peripheral clock\n"); + return PTR_ERR(pclk); + } + sample_clk = clk_get(&pdev->dev, "sample_clk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no sample clock\n"); + retval = PTR_ERR(pclk); + goto out_put_pclk; + } + clk_enable(pclk); + + retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct atmel_abdac), &card); + if (retval) { + dev_dbg(&pdev->dev, "could not create sound card device\n"); + goto out_put_sample_clk; + } + + dac = get_dac(card); + + dac->irq = irq; + dac->card = card; + dac->pclk = pclk; + dac->sample_clk = sample_clk; + dac->pdev = pdev; + + retval = set_sample_rates(dac); + if (retval < 0) { + dev_dbg(&pdev->dev, "could not set supported rates\n"); + goto out_free_card; + } + + dac->regs = ioremap(regs->start, regs->end - regs->start + 1); + if (!dac->regs) { + dev_dbg(&pdev->dev, "could not remap register memory\n"); + goto out_free_card; + } + + /* make sure the DAC is silent and disabled */ + dac_writel(dac, DATA, 0); + dac_writel(dac, CTRL, 0); + + retval = request_irq(irq, abdac_interrupt, 0, "abdac", dac); + if (retval) { + dev_dbg(&pdev->dev, "could not request irq\n"); + goto out_unmap_regs; + } + + snd_card_set_dev(card, &pdev->dev); + + if (pdata->dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->dws; + dma_cap_mask_t mask; + + dws->tx_reg = regs->start + DAC_DATA; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dac->dma.chan = dma_request_channel(mask, filter, dws); + } + if (!pdata->dws.dma_dev || !dac->dma.chan) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto out_unset_card_dev; + } + + strcpy(card->driver, "Atmel ABDAC"); + strcpy(card->shortname, "Atmel ABDAC"); + sprintf(card->longname, "Atmel Audio Bitstream DAC"); + + retval = atmel_abdac_pcm_new(dac); + if (retval) { + dev_dbg(&pdev->dev, "could not register ABDAC pcm device\n"); + goto out_release_dma; + } + + retval = snd_card_register(card); + if (retval) { + dev_dbg(&pdev->dev, "could not register sound card\n"); + goto out_release_dma; + } + + platform_set_drvdata(pdev, card); + + dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n", + dac->regs, dac->dma.chan->dev->device.bus_id); + + return retval; + +out_release_dma: + dma_release_channel(dac->dma.chan); + dac->dma.chan = NULL; +out_unset_card_dev: + snd_card_set_dev(card, NULL); + free_irq(irq, dac); +out_unmap_regs: + iounmap(dac->regs); +out_free_card: + snd_card_free(card); +out_put_sample_clk: + clk_put(sample_clk); + clk_disable(pclk); +out_put_pclk: + clk_put(pclk); + return retval; +} + +#ifdef CONFIG_PM +static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = card->private_data; + + dw_dma_cyclic_stop(dac->dma.chan); + clk_disable(dac->sample_clk); + clk_disable(dac->pclk); + + return 0; +} + +static int atmel_abdac_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = card->private_data; + + clk_enable(dac->pclk); + clk_enable(dac->sample_clk); + if (test_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_start(dac->dma.chan); + + return 0; +} +#else +#define atmel_abdac_suspend NULL +#define atmel_abdac_resume NULL +#endif + +static int __devexit atmel_abdac_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = get_dac(card); + + clk_put(dac->sample_clk); + clk_disable(dac->pclk); + clk_put(dac->pclk); + + dma_release_channel(dac->dma.chan); + dac->dma.chan = NULL; + snd_card_set_dev(card, NULL); + iounmap(dac->regs); + free_irq(dac->irq, dac); + snd_card_free(card); + + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver atmel_abdac_driver = { + .remove = __devexit_p(atmel_abdac_remove), + .driver = { + .name = "atmel_abdac", + }, + .suspend = atmel_abdac_suspend, + .resume = atmel_abdac_resume, +}; + +static int __init atmel_abdac_init(void) +{ + return platform_driver_probe(&atmel_abdac_driver, + atmel_abdac_probe); +} +module_init(atmel_abdac_init); + +static void __exit atmel_abdac_exit(void) +{ + platform_driver_unregister(&atmel_abdac_driver); +} +module_exit(atmel_abdac_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)"); +MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>"); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c new file mode 100644 index 000000000000..dd72e00e5ae1 --- /dev/null +++ b/sound/atmel/ac97c.c @@ -0,0 +1,932 @@ +/* + * Driver for the Atmel AC97C controller + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * 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/clk.h> +#include <linux/delay.h> +#include <linux/bitmap.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/ac97_codec.h> +#include <sound/atmel-ac97c.h> +#include <sound/memalloc.h> + +#include <linux/dw_dmac.h> + +#include "ac97c.h" + +enum { + DMA_TX_READY = 0, + DMA_RX_READY, + DMA_TX_CHAN_PRESENT, + DMA_RX_CHAN_PRESENT, +}; + +/* Serialize access to opened variable */ +static DEFINE_MUTEX(opened_mutex); + +struct atmel_ac97c_dma { + struct dma_chan *rx_chan; + struct dma_chan *tx_chan; +}; + +struct atmel_ac97c { + struct clk *pclk; + struct platform_device *pdev; + struct atmel_ac97c_dma dma; + + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_ac97 *ac97; + struct snd_ac97_bus *ac97_bus; + + u64 cur_format; + unsigned int cur_rate; + unsigned long flags; + /* Serialize access to opened variable */ + spinlock_t lock; + void __iomem *regs; + int opened; + int reset_pin; +}; + +#define get_chip(card) ((struct atmel_ac97c *)(card)->private_data) + +#define ac97c_writel(chip, reg, val) \ + __raw_writel((val), (chip)->regs + AC97C_##reg) +#define ac97c_readl(chip, reg) \ + __raw_readl((chip)->regs + AC97C_##reg) + +/* This function is called by the DMA driver. */ +static void atmel_ac97c_dma_playback_period_done(void *arg) +{ + struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->playback_substream); +} + +static void atmel_ac97c_dma_capture_period_done(void *arg) +{ + struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->capture_substream); +} + +static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip, + struct snd_pcm_substream *substream, + enum dma_data_direction direction) +{ + struct dma_chan *chan; + struct dw_cyclic_desc *cdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long buffer_len, period_len; + + /* + * We don't do DMA on "complex" transfers, i.e. with + * non-halfword-aligned buffers or lengths. + */ + if (runtime->dma_addr & 1 || runtime->buffer_size & 1) { + dev_dbg(&chip->pdev->dev, "too complex transfer\n"); + return -EINVAL; + } + + if (direction == DMA_TO_DEVICE) + chan = chip->dma.tx_chan; + else + chan = chip->dma.rx_chan; + + buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + period_len = frames_to_bytes(runtime, runtime->period_size); + + cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, + period_len, direction); + if (IS_ERR(cdesc)) { + dev_dbg(&chip->pdev->dev, "could not prepare cyclic DMA\n"); + return PTR_ERR(cdesc); + } + + if (direction == DMA_TO_DEVICE) { + cdesc->period_callback = atmel_ac97c_dma_playback_period_done; + set_bit(DMA_TX_READY, &chip->flags); + } else { + cdesc->period_callback = atmel_ac97c_dma_capture_period_done; + set_bit(DMA_RX_READY, &chip->flags); + } + + cdesc->period_callback_param = chip; + + return 0; +} + +static struct snd_pcm_hardware atmel_ac97c_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_JOINT_DUPLEX + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_BE + | SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 64 * 4096, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 4, + .periods_max = 64, +}; + +static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&opened_mutex); + chip->opened++; + runtime->hw = atmel_ac97c_hw; + if (chip->cur_rate) { + runtime->hw.rate_min = chip->cur_rate; + runtime->hw.rate_max = chip->cur_rate; + } + if (chip->cur_format) + runtime->hw.formats = (1ULL << chip->cur_format); + mutex_unlock(&opened_mutex); + chip->playback_substream = substream; + return 0; +} + +static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&opened_mutex); + chip->opened++; + runtime->hw = atmel_ac97c_hw; + if (chip->cur_rate) { + runtime->hw.rate_min = chip->cur_rate; + runtime->hw.rate_max = chip->cur_rate; + } + if (chip->cur_format) + runtime->hw.formats = (1ULL << chip->cur_format); + mutex_unlock(&opened_mutex); + chip->capture_substream = substream; + return 0; +} + +static int atmel_ac97c_playback_close(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + + mutex_lock(&opened_mutex); + chip->opened--; + if (!chip->opened) { + chip->cur_rate = 0; + chip->cur_format = 0; + } + mutex_unlock(&opened_mutex); + + chip->playback_substream = NULL; + + return 0; +} + +static int atmel_ac97c_capture_close(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + + mutex_lock(&opened_mutex); + chip->opened--; + if (!chip->opened) { + chip->cur_rate = 0; + chip->cur_format = 0; + } + mutex_unlock(&opened_mutex); + + chip->capture_substream = NULL; + + return 0; +} + +static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + + /* Set restrictions to params. */ + mutex_lock(&opened_mutex); + chip->cur_rate = params_rate(hw_params); + chip->cur_format = params_format(hw_params); + mutex_unlock(&opened_mutex); + + return retval; +} + +static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + + /* Set restrictions to params. */ + mutex_lock(&opened_mutex); + chip->cur_rate = params_rate(hw_params); + chip->cur_format = params_format(hw_params); + mutex_unlock(&opened_mutex); + + return retval; +} + +static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long word = 0; + int retval; + + /* assign channels to AC97C channel A */ + switch (runtime->channels) { + case 1: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A); + break; + case 2: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A) + | AC97C_CH_ASSIGN(PCM_RIGHT, A); + break; + default: + /* TODO: support more than two channels */ + return -EINVAL; + break; + } + ac97c_writel(chip, OCA, word); + + /* configure sample format and size */ + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + word |= AC97C_CMR_CEM_LITTLE; + break; + case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ + default: + word &= ~(AC97C_CMR_CEM_LITTLE); + break; + } + + ac97c_writel(chip, CAMR, word); + + /* set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AC97C_MR_VRA; + ac97c_writel(chip, MR, word); + } else { + word = ac97c_readl(chip, MR); + word &= ~(AC97C_MR_VRA); + ac97c_writel(chip, MR, word); + } + + retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, + runtime->rate); + if (retval) + dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", + runtime->rate); + + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + + return retval; +} + +static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long word = 0; + int retval; + + /* assign channels to AC97C channel A */ + switch (runtime->channels) { + case 1: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A); + break; + case 2: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A) + | AC97C_CH_ASSIGN(PCM_RIGHT, A); + break; + default: + /* TODO: support more than two channels */ + return -EINVAL; + break; + } + ac97c_writel(chip, ICA, word); + + /* configure sample format and size */ + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + word |= AC97C_CMR_CEM_LITTLE; + break; + case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ + default: + word &= ~(AC97C_CMR_CEM_LITTLE); + break; + } + + ac97c_writel(chip, CAMR, word); + + /* set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AC97C_MR_VRA; + ac97c_writel(chip, MR, word); + } else { + word = ac97c_readl(chip, MR); + word &= ~(AC97C_MR_VRA); + ac97c_writel(chip, MR, word); + } + + retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE, + runtime->rate); + if (retval) + dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", + runtime->rate); + + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + + return retval; +} + +static int +atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + unsigned long camr; + int retval = 0; + + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + camr |= AC97C_CMR_CENA; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(chip->dma.tx_chan); + if (chip->opened <= 1) + camr &= ~AC97C_CMR_CENA; + break; + default: + retval = -EINVAL; + goto out; + } + + ac97c_writel(chip, CAMR, camr); +out: + return retval; +} + +static int +atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + unsigned long camr; + int retval = 0; + + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + camr |= AC97C_CMR_CENA; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (chip->opened <= 1) + camr &= ~AC97C_CMR_CENA; + break; + default: + retval = -EINVAL; + break; + } + + ac97c_writel(chip, CAMR, camr); +out: + return retval; +} + +static snd_pcm_uframes_t +atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + return frames; +} + +static snd_pcm_uframes_t +atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + return frames; +} + +static struct snd_pcm_ops atmel_ac97_playback_ops = { + .open = atmel_ac97c_playback_open, + .close = atmel_ac97c_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_ac97c_playback_hw_params, + .hw_free = atmel_ac97c_playback_hw_free, + .prepare = atmel_ac97c_playback_prepare, + .trigger = atmel_ac97c_playback_trigger, + .pointer = atmel_ac97c_playback_pointer, +}; + +static struct snd_pcm_ops atmel_ac97_capture_ops = { + .open = atmel_ac97c_capture_open, + .close = atmel_ac97c_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_ac97c_capture_hw_params, + .hw_free = atmel_ac97c_capture_hw_free, + .prepare = atmel_ac97c_capture_prepare, + .trigger = atmel_ac97c_capture_trigger, + .pointer = atmel_ac97c_capture_pointer, +}; + +static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) +{ + struct snd_pcm *pcm; + struct snd_pcm_hardware hw = atmel_ac97c_hw; + int capture, playback, retval; + + capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + + retval = snd_pcm_new(chip->card, chip->card->shortname, + chip->pdev->id, playback, capture, &pcm); + if (retval) + return retval; + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &atmel_ac97_capture_ops); + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &atmel_ac97_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pdev->dev, hw.periods_min * hw.period_bytes_min, + hw.buffer_bytes_max); + if (retval) + return retval; + + pcm->private_data = chip; + pcm->info_flags = 0; + strcpy(pcm->name, chip->card->shortname); + chip->pcm = pcm; + + return 0; +} + +static int atmel_ac97c_mixer_new(struct atmel_ac97c *chip) +{ + struct snd_ac97_template template; + memset(&template, 0, sizeof(template)); + template.private_data = chip; + return snd_ac97_mixer(chip->ac97_bus, &template, &chip->ac97); +} + +static void atmel_ac97c_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct atmel_ac97c *chip = get_chip(ac97); + unsigned long word; + int timeout = 40; + + word = (reg & 0x7f) << 16 | val; + + do { + if (ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) { + ac97c_writel(chip, COTHR, word); + return; + } + udelay(1); + } while (--timeout); + + dev_dbg(&chip->pdev->dev, "codec write timeout\n"); +} + +static unsigned short atmel_ac97c_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct atmel_ac97c *chip = get_chip(ac97); + unsigned long word; + int timeout = 40; + int write = 10; + + word = (0x80 | (reg & 0x7f)) << 16; + + if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) + ac97c_readl(chip, CORHR); + +retry_write: + timeout = 40; + + do { + if ((ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) != 0) { + ac97c_writel(chip, COTHR, word); + goto read_reg; + } + udelay(10); + } while (--timeout); + + if (!--write) + goto timed_out; + goto retry_write; + +read_reg: + do { + if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) { + unsigned short val = ac97c_readl(chip, CORHR); + return val; + } + udelay(10); + } while (--timeout); + + if (!--write) + goto timed_out; + goto retry_write; + +timed_out: + dev_dbg(&chip->pdev->dev, "codec read timeout\n"); + return 0xffff; +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct dw_dma_slave *dws = slave; + + if (dws->dma_dev == chan->device->dev) { + chan->private = dws; + return true; + } else + return false; +} + +static void atmel_ac97c_reset(struct atmel_ac97c *chip) +{ + ac97c_writel(chip, MR, AC97C_MR_WRST); + + if (gpio_is_valid(chip->reset_pin)) { + gpio_set_value(chip->reset_pin, 0); + /* AC97 v2.2 specifications says minimum 1 us. */ + udelay(10); + gpio_set_value(chip->reset_pin, 1); + } + + udelay(1); + ac97c_writel(chip, MR, AC97C_MR_ENA); +} + +static int __devinit atmel_ac97c_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct atmel_ac97c *chip; + struct resource *regs; + struct ac97c_platform_data *pdata; + struct clk *pclk; + static struct snd_ac97_bus_ops ops = { + .write = atmel_ac97c_write, + .read = atmel_ac97c_read, + }; + int retval; + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) { + dev_dbg(&pdev->dev, "no memory resource\n"); + return -ENXIO; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_dbg(&pdev->dev, "no platform data\n"); + return -ENXIO; + } + + pclk = clk_get(&pdev->dev, "pclk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no peripheral clock\n"); + return PTR_ERR(pclk); + } + clk_enable(pclk); + + retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct atmel_ac97c), &card); + if (retval) { + dev_dbg(&pdev->dev, "could not create sound card device\n"); + goto err_snd_card_new; + } + + chip = get_chip(card); + + spin_lock_init(&chip->lock); + + strcpy(card->driver, "Atmel AC97C"); + strcpy(card->shortname, "Atmel AC97C"); + sprintf(card->longname, "Atmel AC97 controller"); + + chip->card = card; + chip->pclk = pclk; + chip->pdev = pdev; + chip->regs = ioremap(regs->start, regs->end - regs->start + 1); + + if (!chip->regs) { + dev_dbg(&pdev->dev, "could not remap register memory\n"); + goto err_ioremap; + } + + if (gpio_is_valid(pdata->reset_pin)) { + if (gpio_request(pdata->reset_pin, "reset_pin")) { + dev_dbg(&pdev->dev, "reset pin not available\n"); + chip->reset_pin = -ENODEV; + } else { + gpio_direction_output(pdata->reset_pin, 1); + chip->reset_pin = pdata->reset_pin; + } + } + + snd_card_set_dev(card, &pdev->dev); + + retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus); + if (retval) { + dev_dbg(&pdev->dev, "could not register on ac97 bus\n"); + goto err_ac97_bus; + } + + atmel_ac97c_reset(chip); + + retval = atmel_ac97c_mixer_new(chip); + if (retval) { + dev_dbg(&pdev->dev, "could not register ac97 mixer\n"); + goto err_ac97_bus; + } + + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; + + dws->rx_reg = regs->start + AC97C_CARHR + 2; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + chip->dma.rx_chan->dev->device.bus_id); + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } + + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; + + dws->tx_reg = regs->start + AC97C_CATHR + 2; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + chip->dma.tx_chan->dev->device.bus_id); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } + + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + + retval = atmel_ac97c_pcm_new(chip); + if (retval) { + dev_dbg(&pdev->dev, "could not register ac97 pcm device\n"); + goto err_dma; + } + + retval = snd_card_register(card); + if (retval) { + dev_dbg(&pdev->dev, "could not register sound card\n"); + goto err_ac97_bus; + } + + platform_set_drvdata(pdev, card); + + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", + chip->regs); + + return 0; + +err_dma: + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; +err_ac97_bus: + snd_card_set_dev(card, NULL); + + if (gpio_is_valid(chip->reset_pin)) + gpio_free(chip->reset_pin); + + iounmap(chip->regs); +err_ioremap: + snd_card_free(card); +err_snd_card_new: + clk_disable(pclk); + clk_put(pclk); + return retval; +} + +#ifdef CONFIG_PM +static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = card->private_data; + + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + clk_disable(chip->pclk); + + return 0; +} + +static int atmel_ac97c_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = card->private_data; + + clk_enable(chip->pclk); + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + + return 0; +} +#else +#define atmel_ac97c_suspend NULL +#define atmel_ac97c_resume NULL +#endif + +static int __devexit atmel_ac97c_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = get_chip(card); + + if (gpio_is_valid(chip->reset_pin)) + gpio_free(chip->reset_pin); + + clk_disable(chip->pclk); + clk_put(chip->pclk); + iounmap(chip->regs); + + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + + snd_card_set_dev(card, NULL); + snd_card_free(card); + + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver atmel_ac97c_driver = { + .remove = __devexit_p(atmel_ac97c_remove), + .driver = { + .name = "atmel_ac97c", + }, + .suspend = atmel_ac97c_suspend, + .resume = atmel_ac97c_resume, +}; + +static int __init atmel_ac97c_init(void) +{ + return platform_driver_probe(&atmel_ac97c_driver, + atmel_ac97c_probe); +} +module_init(atmel_ac97c_init); + +static void __exit atmel_ac97c_exit(void) +{ + platform_driver_unregister(&atmel_ac97c_driver); +} +module_exit(atmel_ac97c_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); +MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>"); diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h new file mode 100644 index 000000000000..c17bd5825980 --- /dev/null +++ b/sound/atmel/ac97c.h @@ -0,0 +1,71 @@ +/* + * Register definitions for the Atmel AC97C controller + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * 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 __SOUND_ATMEL_AC97C_H +#define __SOUND_ATMEL_AC97C_H + +#define AC97C_MR 0x08 +#define AC97C_ICA 0x10 +#define AC97C_OCA 0x14 +#define AC97C_CARHR 0x20 +#define AC97C_CATHR 0x24 +#define AC97C_CASR 0x28 +#define AC97C_CAMR 0x2c +#define AC97C_CBRHR 0x30 +#define AC97C_CBTHR 0x34 +#define AC97C_CBSR 0x38 +#define AC97C_CBMR 0x3c +#define AC97C_CORHR 0x40 +#define AC97C_COTHR 0x44 +#define AC97C_COSR 0x48 +#define AC97C_COMR 0x4c +#define AC97C_SR 0x50 +#define AC97C_IER 0x54 +#define AC97C_IDR 0x58 +#define AC97C_IMR 0x5c +#define AC97C_VERSION 0xfc + +#define AC97C_CATPR PDC_TPR +#define AC97C_CATCR PDC_TCR +#define AC97C_CATNPR PDC_TNPR +#define AC97C_CATNCR PDC_TNCR +#define AC97C_CARPR PDC_RPR +#define AC97C_CARCR PDC_RCR +#define AC97C_CARNPR PDC_RNPR +#define AC97C_CARNCR PDC_RNCR +#define AC97C_PTCR PDC_PTCR + +#define AC97C_MR_ENA (1 << 0) +#define AC97C_MR_WRST (1 << 1) +#define AC97C_MR_VRA (1 << 2) + +#define AC97C_CSR_TXRDY (1 << 0) +#define AC97C_CSR_UNRUN (1 << 2) +#define AC97C_CSR_RXRDY (1 << 4) +#define AC97C_CSR_ENDTX (1 << 10) +#define AC97C_CSR_ENDRX (1 << 14) + +#define AC97C_CMR_SIZE_20 (0 << 16) +#define AC97C_CMR_SIZE_18 (1 << 16) +#define AC97C_CMR_SIZE_16 (2 << 16) +#define AC97C_CMR_SIZE_10 (3 << 16) +#define AC97C_CMR_CEM_LITTLE (1 << 18) +#define AC97C_CMR_CEM_BIG (0 << 18) +#define AC97C_CMR_CENA (1 << 21) +#define AC97C_CMR_DMAEN (1 << 22) + +#define AC97C_SR_CAEVT (1 << 3) + +#define AC97C_CH_ASSIGN(slot, channel) \ + (AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3))) +#define AC97C_CHANNEL_NONE 0x0 +#define AC97C_CHANNEL_A 0x1 +#define AC97C_CHANNEL_B 0x2 + +#endif /* __SOUND_ATMEL_AC97C_H */ diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 195cafc5a553..a70ee7f1ed98 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -99,9 +99,6 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) if (hw == NULL) return -ENODEV; - if (!hw->ops.open) - return -ENXIO; - if (!try_module_get(hw->card->module)) return -EFAULT; @@ -113,6 +110,10 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) err = -EBUSY; break; } + if (!hw->ops.open) { + err = 0; + break; + } err = hw->ops.open(hw, file); if (err >= 0) break; @@ -151,7 +152,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) static int snd_hwdep_release(struct inode *inode, struct file * file) { - int err = -ENXIO; + int err = 0; struct snd_hwdep *hw = file->private_data; struct module *mod = hw->card->module; diff --git a/sound/core/init.c b/sound/core/init.c index dc4b80c7f311..fd56afe846ed 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -208,6 +208,7 @@ int snd_card_create(int idx, const char *xid, INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); + INIT_LIST_HEAD(&card->files_list); init_waitqueue_head(&card->shutdown_sleep); #ifdef CONFIG_PM mutex_init(&card->power_lock); @@ -274,6 +275,7 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) list_for_each_entry(_df, &shutdown_files, shutdown_list) { if (_df->file == file) { df = _df; + list_del_init(&df->shutdown_list); break; } } @@ -362,8 +364,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 2: replace file->f_op with special dummy operations */ spin_lock(&card->files_lock); - mfile = card->files; - while (mfile) { + list_for_each_entry(mfile, &card->files_list, list) { file = mfile->file; /* it's critical part, use endless loop */ @@ -376,8 +377,6 @@ int snd_card_disconnect(struct snd_card *card) mfile->file->f_op = &snd_shutdown_f_ops; fops_get(mfile->file->f_op); - - mfile = mfile->next; } spin_unlock(&card->files_lock); @@ -457,7 +456,7 @@ int snd_card_free_when_closed(struct snd_card *card) return ret; spin_lock(&card->files_lock); - if (card->files == NULL) + if (list_empty(&card->files_list)) free_now = 1; else card->free_on_last_close = 1; @@ -477,7 +476,7 @@ int snd_card_free(struct snd_card *card) return ret; /* wait, until all devices are ready for the free operation */ - wait_event(card->shutdown_sleep, card->files == NULL); + wait_event(card->shutdown_sleep, list_empty(&card->files_list)); snd_card_do_free(card); return 0; } @@ -824,15 +823,13 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return -ENOMEM; mfile->file = file; mfile->disconnected_f_op = NULL; - mfile->next = NULL; spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); kfree(mfile); return -ENODEV; } - mfile->next = card->files; - card->files = mfile; + list_add(&mfile->list, &card->files_list); spin_unlock(&card->files_lock); return 0; } @@ -854,29 +851,20 @@ EXPORT_SYMBOL(snd_card_file_add); */ int snd_card_file_remove(struct snd_card *card, struct file *file) { - struct snd_monitor_file *mfile, *pfile = NULL; + struct snd_monitor_file *mfile, *found = NULL; int last_close = 0; spin_lock(&card->files_lock); - mfile = card->files; - while (mfile) { + list_for_each_entry(mfile, &card->files_list, list) { if (mfile->file == file) { - if (pfile) - pfile->next = mfile->next; - else - card->files = mfile->next; + list_del(&mfile->list); + if (mfile->disconnected_f_op) + fops_put(mfile->disconnected_f_op); + found = mfile; break; } - pfile = mfile; - mfile = mfile->next; - } - if (mfile && mfile->disconnected_f_op) { - fops_put(mfile->disconnected_f_op); - spin_lock(&shutdown_lock); - list_del(&mfile->shutdown_list); - spin_unlock(&shutdown_lock); } - if (card->files == NULL) + if (list_empty(&card->files_list)) last_close = 1; spin_unlock(&card->files_lock); if (last_close) { @@ -884,11 +872,11 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) if (card->free_on_last_close) snd_card_do_free(card); } - if (!mfile) { + if (!found) { snd_printk(KERN_ERR "ALSA card file remove problem (%p)\n", file); return -ENOENT; } - kfree(mfile); + kfree(found); return 0; } diff --git a/sound/core/jack.c b/sound/core/jack.c index dd4a12dc09aa..c8254c667c62 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -23,6 +23,14 @@ #include <sound/jack.h> #include <sound/core.h> +static int jack_types[] = { + SW_HEADPHONE_INSERT, + SW_MICROPHONE_INSERT, + SW_LINEOUT_INSERT, + SW_JACK_PHYSICAL_INSERT, + SW_VIDEOOUT_INSERT, +}; + static int snd_jack_dev_free(struct snd_device *device) { struct snd_jack *jack = device->device_data; @@ -47,7 +55,7 @@ static int snd_jack_dev_register(struct snd_device *device) int err; snprintf(jack->name, sizeof(jack->name), "%s %s", - card->longname, jack->id); + card->shortname, jack->id); jack->input_dev->name = jack->name; /* Default to the sound card device. */ @@ -79,6 +87,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, { struct snd_jack *jack; int err; + int i; static struct snd_device_ops ops = { .dev_free = snd_jack_dev_free, .dev_register = snd_jack_dev_register, @@ -100,18 +109,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - if (type & SND_JACK_HEADPHONE) - input_set_capability(jack->input_dev, EV_SW, - SW_HEADPHONE_INSERT); - if (type & SND_JACK_LINEOUT) - input_set_capability(jack->input_dev, EV_SW, - SW_LINEOUT_INSERT); - if (type & SND_JACK_MICROPHONE) - input_set_capability(jack->input_dev, EV_SW, - SW_MICROPHONE_INSERT); - if (type & SND_JACK_MECHANICAL) - input_set_capability(jack->input_dev, EV_SW, - SW_JACK_PHYSICAL_INSERT); + for (i = 0; i < ARRAY_SIZE(jack_types); i++) + if (type & (1 << i)) + input_set_capability(jack->input_dev, EV_SW, + jack_types[i]); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -154,21 +155,17 @@ EXPORT_SYMBOL(snd_jack_set_parent); */ void snd_jack_report(struct snd_jack *jack, int status) { + int i; + if (!jack) return; - if (jack->type & SND_JACK_HEADPHONE) - input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, - status & SND_JACK_HEADPHONE); - if (jack->type & SND_JACK_LINEOUT) - input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, - status & SND_JACK_LINEOUT); - if (jack->type & SND_JACK_MICROPHONE) - input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, - status & SND_JACK_MICROPHONE); - if (jack->type & SND_JACK_MECHANICAL) - input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, - status & SND_JACK_MECHANICAL); + for (i = 0; i < ARRAY_SIZE(jack_types); i++) { + int testbit = 1 << i; + if (jack->type & testbit) + input_report_switch(jack->input_dev, jack_types[i], + status & testbit); + } input_sync(jack->input_dev); } diff --git a/sound/core/misc.c b/sound/core/misc.c index 38524f615d94..a9710e0c97af 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -95,12 +95,14 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; - for (q = list; q->subvendor; q++) - if (q->subvendor == pci->subsystem_vendor && - (!q->subdevice || q->subdevice == pci->subsystem_device)) + for (q = list; q->subvendor; q++) { + if (q->subvendor != pci->subsystem_vendor) + continue; + if (!q->subdevice || + (pci->subsystem_device & q->subdevice_mask) == q->subdevice) return q; + } return NULL; } - EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 4690b8b5681f..e570649184e2 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -692,6 +692,9 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) { + snd_mixer_oss_put_volume1_vol(fmixer, pslot, + slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e17836680f49..2864cefb773c 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1160,9 +1160,11 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: write: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: write: " + "recovering from XRUN\n"); else - printk("pcm_oss: write: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: write: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_oss_prepare(substream); if (ret < 0) @@ -1196,9 +1198,11 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: read: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: read: " + "recovering from XRUN\n"); else - printk("pcm_oss: read: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: read: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) @@ -1242,9 +1246,11 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: writev: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: writev: " + "recovering from XRUN\n"); else - printk("pcm_oss: writev: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: writev: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_oss_prepare(substream); if (ret < 0) @@ -1278,9 +1284,11 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void * runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: readv: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: readv: " + "recovering from XRUN\n"); else - printk("pcm_oss: readv: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: readv: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) @@ -1533,7 +1541,7 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); #ifdef OSS_DEBUG - printk("sync1: size = %li\n", size); + printk(KERN_DEBUG "sync1: size = %li\n", size); #endif while (1) { result = snd_pcm_oss_write2(substream, runtime->oss.buffer, size, 1); @@ -1590,7 +1598,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) mutex_lock(&runtime->oss.params_lock); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG - printk("sync: buffer_used\n"); + printk(KERN_DEBUG "sync: buffer_used\n"); #endif size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width; snd_pcm_format_set_silence(format, @@ -1603,7 +1611,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) } } else if (runtime->oss.period_ptr > 0) { #ifdef OSS_DEBUG - printk("sync: period_ptr\n"); + printk(KERN_DEBUG "sync: period_ptr\n"); #endif size = runtime->oss.period_bytes - runtime->oss.period_ptr; snd_pcm_format_set_silence(format, @@ -1767,7 +1775,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE | AFMT_S32_LE | AFMT_S32_BE | - AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_LE | AFMT_S24_BE | AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) @@ -1952,7 +1960,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr int err, cmd; #ifdef OSS_DEBUG - printk("pcm_oss: trigger = 0x%x\n", trigger); + printk(KERN_DEBUG "pcm_oss: trigger = 0x%x\n", trigger); #endif psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; @@ -2170,7 +2178,9 @@ static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stre } #ifdef OSS_DEBUG - printk("pcm_oss: space: bytes = %i, fragments = %i, fragstotal = %i, fragsize = %i\n", info.bytes, info.fragments, info.fragstotal, info.fragsize); + printk(KERN_DEBUG "pcm_oss: space: bytes = %i, fragments = %i, " + "fragstotal = %i, fragsize = %i\n", + info.bytes, info.fragments, info.fragstotal, info.fragsize); #endif if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; @@ -2473,7 +2483,7 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long if (((cmd >> 8) & 0xff) != 'P') return -EINVAL; #ifdef OSS_DEBUG - printk("pcm_oss: ioctl = 0x%x\n", cmd); + printk(KERN_DEBUG "pcm_oss: ioctl = 0x%x\n", cmd); #endif switch (cmd) { case SNDCTL_DSP_RESET: @@ -2627,7 +2637,8 @@ static ssize_t snd_pcm_oss_read(struct file *file, char __user *buf, size_t coun #else { ssize_t res = snd_pcm_oss_read1(substream, buf, count); - printk("pcm_oss: read %li bytes (returned %li bytes)\n", (long)count, (long)res); + printk(KERN_DEBUG "pcm_oss: read %li bytes " + "(returned %li bytes)\n", (long)count, (long)res); return res; } #endif @@ -2646,7 +2657,8 @@ static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size substream->f_flags = file->f_flags & O_NONBLOCK; result = snd_pcm_oss_write1(substream, buf, count); #ifdef OSS_DEBUG - printk("pcm_oss: write %li bytes (wrote %li bytes)\n", (long)count, (long)result); + printk(KERN_DEBUG "pcm_oss: write %li bytes (wrote %li bytes)\n", + (long)count, (long)result); #endif return result; } @@ -2720,7 +2732,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) int err; #ifdef OSS_DEBUG - printk("pcm_oss: mmap begin\n"); + printk(KERN_DEBUG "pcm_oss: mmap begin\n"); #endif pcm_oss_file = file->private_data; switch ((area->vm_flags & (VM_READ | VM_WRITE))) { @@ -2770,7 +2782,8 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) runtime->silence_threshold = 0; runtime->silence_size = 0; #ifdef OSS_DEBUG - printk("pcm_oss: mmap ok, bytes = 0x%x\n", runtime->oss.mmap_bytes); + printk(KERN_DEBUG "pcm_oss: mmap ok, bytes = 0x%x\n", + runtime->oss.mmap_bytes); #endif /* In mmap mode we never stop */ runtime->stop_threshold = runtime->boundary; @@ -2872,7 +2885,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, setup = kmalloc(sizeof(*setup), GFP_KERNEL); if (! setup) { buffer->error = -ENOMEM; - mutex_lock(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); return; } if (pstr->oss.setup_list == NULL) @@ -2886,7 +2899,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, if (! template.task_name) { kfree(setup); buffer->error = -ENOMEM; - mutex_lock(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); return; } } diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index ca2f4c39be46..b9afab603711 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -176,9 +176,9 @@ static inline int snd_pcm_plug_slave_format(int format, struct snd_mask *format_ #endif #ifdef PLUGIN_DEBUG -#define pdprintf( fmt, args... ) printk( "plugin: " fmt, ##args) +#define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else -#define pdprintf( fmt, args... ) +#define pdprintf(fmt, args...) #endif #endif /* __PCM_PLUGIN_H */ diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index a466443c4a26..2fa9299a440d 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, while (dst_frames1 > 0) { S1 = S2; if (src_frames1-- > 0) { - S1 = *src; + S2 = *src; src += src_step; } if (pos & ~R_MASK) { diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 192a433a2403..145931a9ff30 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -667,7 +667,6 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) spin_lock_init(&substream->self_group.lock); INIT_LIST_HEAD(&substream->self_group.substreams); list_add_tail(&substream->link_list, &substream->self_group.substreams); - spin_lock_init(&substream->timer_lock); atomic_set(&substream->mmap_count, 0); prev = substream; } @@ -692,7 +691,7 @@ EXPORT_SYMBOL(snd_pcm_new_stream); * * Returns zero if successful, or a negative error code on failure. */ -int snd_pcm_new(struct snd_card *card, char *id, int device, +int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm ** rpcm) { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 921691080f35..fbb2e391591e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -125,23 +125,32 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG +#define xrun_debug(substream) ((substream)->pstr->xrun_debug) +#else +#define xrun_debug(substream) 0 +#endif + +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream) > 1) \ + dump_stack(); \ + } while (0) + static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (substream->pstr->xrun_debug) { + if (xrun_debug(substream)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p'); - if (substream->pstr->xrun_debug > 1) - dump_stack(); + dump_stack_on_xrun(substream); } -#endif } -static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +static snd_pcm_uframes_t +snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t pos; @@ -150,17 +159,21 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) return pos; /* XRUN */ -#ifdef CONFIG_SND_DEBUG if (pos >= runtime->buffer_size) { - snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size); + if (printk_ratelimit()) { + snd_printd(KERN_ERR "BUG: stream = %i, pos = 0x%lx, " + "buffer size = 0x%lx, period size = 0x%lx\n", + substream->stream, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } -#endif pos -= pos % runtime->min_align; return pos; } -static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -182,11 +195,21 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream return 0; } -static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream)) { \ + if (printk_ratelimit()) { \ + snd_printd("PCM: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + +static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; + snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt, hw_base; snd_pcm_sframes_t delta; pos = snd_pcm_update_hw_ptr_pos(substream, runtime); @@ -194,36 +217,53 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs xrun(substream); return -EPIPE; } - if (runtime->period_size == runtime->buffer_size) - goto __next_buf; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - - delta = hw_ptr_interrupt - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 1 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif - return 0; + delta = new_hw_ptr - hw_ptr_interrupt; + if (hw_ptr_interrupt >= runtime->boundary) { + hw_ptr_interrupt -= runtime->boundary; + if (hw_base < runtime->boundary / 2) + /* hw_base was already lapped; recalc delta */ + delta = new_hw_ptr - hw_ptr_interrupt; + } + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value " + "(stream=%i, pos=%ld, intr_ptr=%ld)\n", + substream->stream, (long)pos, + (long)hw_ptr_interrupt); + /* rebase to interrupt position */ + hw_base = new_hw_ptr = hw_ptr_interrupt; + /* align hw_base to buffer_size */ + hw_base -= hw_base % runtime->buffer_size; + delta = 0; + } else { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; } - __next_buf: - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; } - + if (delta > runtime->period_size) { + hw_ptr_error(substream, + "Lost interrupts? " + "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + substream->stream, (long)delta, + (long)hw_ptr_interrupt); + /* rebase hw_ptr_interrupt */ + hw_ptr_interrupt = + new_hw_ptr - new_hw_ptr % runtime->period_size; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size; + runtime->hw_ptr_interrupt = hw_ptr_interrupt; return snd_pcm_update_hw_ptr_post(substream, runtime); } @@ -233,7 +273,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t delta; old_hw_ptr = runtime->status->hw_ptr; @@ -242,29 +282,38 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - new_hw_ptr = runtime->hw_ptr_base + pos; - - delta = old_hw_ptr - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 2 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; + + delta = new_hw_ptr - old_hw_ptr; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value [2] " + "(stream=%i, pos=%ld, old_ptr=%ld)\n", + substream->stream, (long)pos, + (long)old_hw_ptr); return 0; } - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + if (delta > runtime->period_size && runtime->periods > 1) { + hw_ptr_error(substream, + "hw_ptr skipping! " + "(pos=%ld, delta=%ld, period=%ld)\n", + (long)pos, (long)delta, + (long)runtime->period_size); + return 0; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; return snd_pcm_update_hw_ptr_post(substream, runtime); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a789efc9df39..d9b8f5379428 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -186,7 +186,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!(params->rmask & (1 << k))) continue; #ifdef RULES_DEBUG - printk("%s = ", snd_pcm_hw_param_names[k]); + printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]); printk("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); #endif changed = snd_mask_refine(m, constrs_mask(constrs, k)); @@ -206,7 +206,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!(params->rmask & (1 << k))) continue; #ifdef RULES_DEBUG - printk("%s = ", snd_pcm_hw_param_names[k]); + printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]); if (i->empty) printk("empty"); else @@ -251,7 +251,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!doit) continue; #ifdef RULES_DEBUG - printk("Rule %d [%p]: ", k, r->func); + printk(KERN_DEBUG "Rule %d [%p]: ", k, r->func); if (r->var >= 0) { printk("%s = ", snd_pcm_hw_param_names[r->var]); if (hw_is_mask(r->var)) { diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index 2c89c04f2916..ca8068b63d6c 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -85,25 +85,19 @@ static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) static int snd_pcm_timer_start(struct snd_timer * timer) { - unsigned long flags; struct snd_pcm_substream *substream; substream = snd_timer_chip(timer); - spin_lock_irqsave(&substream->timer_lock, flags); substream->timer_running = 1; - spin_unlock_irqrestore(&substream->timer_lock, flags); return 0; } static int snd_pcm_timer_stop(struct snd_timer * timer) { - unsigned long flags; struct snd_pcm_substream *substream; substream = snd_timer_chip(timer); - spin_lock_irqsave(&substream->timer_lock, flags); substream->timer_running = 0; - spin_unlock_irqrestore(&substream->timer_lock, flags); return 0; } diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 002777ba336a..473247c8e6d3 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -224,156 +224,143 @@ int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) return 0; } -int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, - int mode, struct snd_rawmidi_file * rfile) +/* look for an available substream for the given stream direction; + * if a specific subdevice is given, try to assign it + */ +static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, + int stream, int mode, + struct snd_rawmidi_substream **sub_ret) +{ + struct snd_rawmidi_substream *substream; + struct snd_rawmidi_str *s = &rmidi->streams[stream]; + static unsigned int info_flags[2] = { + [SNDRV_RAWMIDI_STREAM_OUTPUT] = SNDRV_RAWMIDI_INFO_OUTPUT, + [SNDRV_RAWMIDI_STREAM_INPUT] = SNDRV_RAWMIDI_INFO_INPUT, + }; + + if (!(rmidi->info_flags & info_flags[stream])) + return -ENXIO; + if (subdevice >= 0 && subdevice >= s->substream_count) + return -ENODEV; + if (s->substream_opened >= s->substream_count) + return -EAGAIN; + + list_for_each_entry(substream, &s->substreams, list) { + if (substream->opened) { + if (stream == SNDRV_RAWMIDI_STREAM_INPUT || + !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + continue; + } + if (subdevice < 0 || subdevice == substream->number) { + *sub_ret = substream; + return 0; + } + } + return -EAGAIN; +} + +/* open and do ref-counting for the given substream */ +static int open_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int mode) +{ + int err; + + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) + return err; + substream->opened = 1; + if (substream->use_count++ == 0) + substream->active_sensing = 1; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + rmidi->streams[substream->stream].substream_opened++; + return 0; +} + +static void close_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int cleanup); + +static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, + struct snd_rawmidi_file *rfile) { - struct snd_rawmidi *rmidi; - struct list_head *list1, *list2; struct snd_rawmidi_substream *sinput = NULL, *soutput = NULL; - struct snd_rawmidi_runtime *input = NULL, *output = NULL; int err; - if (rfile) - rfile->input = rfile->output = NULL; - mutex_lock(®ister_mutex); - rmidi = snd_rawmidi_search(card, device); - mutex_unlock(®ister_mutex); - if (rmidi == NULL) { - err = -ENODEV; - goto __error1; - } - if (!try_module_get(rmidi->card->module)) { - err = -EFAULT; - goto __error1; - } - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_lock(&rmidi->open_mutex); + rfile->input = rfile->output = NULL; if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT)) { - err = -ENXIO; - goto __error; - } - if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) { - err = -ENODEV; - goto __error; - } - if (rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened >= - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) { - err = -EAGAIN; + err = assign_substream(rmidi, subdevice, + SNDRV_RAWMIDI_STREAM_INPUT, + mode, &sinput); + if (err < 0) goto __error; - } } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT)) { - err = -ENXIO; - goto __error; - } - if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) { - err = -ENODEV; - goto __error; - } - if (rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened >= - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) { - err = -EAGAIN; + err = assign_substream(rmidi, subdevice, + SNDRV_RAWMIDI_STREAM_OUTPUT, + mode, &soutput); + if (err < 0) goto __error; - } - } - list1 = rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams.next; - while (1) { - if (list1 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { - sinput = NULL; - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - err = -EAGAIN; - goto __error; - } - break; - } - sinput = list_entry(list1, struct snd_rawmidi_substream, list); - if ((mode & SNDRV_RAWMIDI_LFLG_INPUT) && sinput->opened) - goto __nexti; - if (subdevice < 0 || (subdevice >= 0 && subdevice == sinput->number)) - break; - __nexti: - list1 = list1->next; } - list2 = rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams.next; - while (1) { - if (list2 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { - soutput = NULL; - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - err = -EAGAIN; - goto __error; - } - break; - } - soutput = list_entry(list2, struct snd_rawmidi_substream, list); - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) { - if (soutput->opened && !soutput->append) - goto __nexto; - } else { - if (soutput->opened) - goto __nexto; - } - } - if (subdevice < 0 || (subdevice >= 0 && subdevice == soutput->number)) - break; - __nexto: - list2 = list2->next; - } - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - if ((err = snd_rawmidi_runtime_create(sinput)) < 0) - goto __error; - input = sinput->runtime; - if ((err = sinput->ops->open(sinput)) < 0) + + if (sinput) { + err = open_substream(rmidi, sinput, mode); + if (err < 0) goto __error; - sinput->opened = 1; - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened++; - } else { - sinput = NULL; } - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (soutput->opened) - goto __skip_output; - if ((err = snd_rawmidi_runtime_create(soutput)) < 0) { - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) - sinput->ops->close(sinput); - goto __error; - } - output = soutput->runtime; - if ((err = soutput->ops->open(soutput)) < 0) { - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) - sinput->ops->close(sinput); + if (soutput) { + err = open_substream(rmidi, soutput, mode); + if (err < 0) { + if (sinput) + close_substream(rmidi, sinput, 0); goto __error; } - __skip_output: - soutput->opened = 1; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - soutput->append = 1; - if (soutput->use_count++ == 0) - soutput->active_sensing = 1; - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened++; - } else { - soutput = NULL; - } - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_unlock(&rmidi->open_mutex); - if (rfile) { - rfile->rmidi = rmidi; - rfile->input = sinput; - rfile->output = soutput; } + + rfile->rmidi = rmidi; + rfile->input = sinput; + rfile->output = soutput; return 0; __error: - if (input != NULL) + if (sinput && sinput->runtime) snd_rawmidi_runtime_free(sinput); - if (output != NULL) + if (soutput && soutput->runtime) snd_rawmidi_runtime_free(soutput); - module_put(rmidi->card->module); - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_unlock(&rmidi->open_mutex); - __error1: + return err; +} + +/* called from sound/core/seq/seq_midi.c */ +int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, + int mode, struct snd_rawmidi_file * rfile) +{ + struct snd_rawmidi *rmidi; + int err; + + if (snd_BUG_ON(!rfile)) + return -EINVAL; + + mutex_lock(®ister_mutex); + rmidi = snd_rawmidi_search(card, device); + if (rmidi == NULL) { + mutex_unlock(®ister_mutex); + return -ENODEV; + } + if (!try_module_get(rmidi->card->module)) { + mutex_unlock(®ister_mutex); + return -ENXIO; + } + mutex_unlock(®ister_mutex); + + mutex_lock(&rmidi->open_mutex); + err = rawmidi_open_priv(rmidi, subdevice, mode, rfile); + mutex_unlock(&rmidi->open_mutex); + if (err < 0) + module_put(rmidi->card->module); return err; } @@ -385,10 +372,13 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) unsigned short fflags; int err; struct snd_rawmidi *rmidi; - struct snd_rawmidi_file *rawmidi_file; + struct snd_rawmidi_file *rawmidi_file = NULL; wait_queue_t wait; struct snd_ctl_file *kctl; + if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) + return -EINVAL; /* invalid combination */ + if (maj == snd_major) { rmidi = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_RAWMIDI); @@ -402,24 +392,25 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) if (rmidi == NULL) return -ENODEV; - if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) - return -EINVAL; /* invalid combination */ + + if (!try_module_get(rmidi->card->module)) + return -ENXIO; + + mutex_lock(&rmidi->open_mutex); card = rmidi->card; err = snd_card_file_add(card, file); if (err < 0) - return -ENODEV; + goto __error_card; fflags = snd_rawmidi_file_flags(file); if ((file->f_flags & O_APPEND) || maj == SOUND_MAJOR) /* OSS emul? */ fflags |= SNDRV_RAWMIDI_LFLG_APPEND; - fflags |= SNDRV_RAWMIDI_LFLG_NOOPENLOCK; rawmidi_file = kmalloc(sizeof(*rawmidi_file), GFP_KERNEL); if (rawmidi_file == NULL) { - snd_card_file_remove(card, file); - return -ENOMEM; + err = -ENOMEM; + goto __error; } init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); - mutex_lock(&rmidi->open_mutex); while (1) { subdevice = -1; read_lock(&card->ctl_files_rwlock); @@ -431,8 +422,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) } } read_unlock(&card->ctl_files_rwlock); - err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device, - subdevice, fflags, rawmidi_file); + err = rawmidi_open_priv(rmidi, subdevice, fflags, rawmidi_file); if (err >= 0) break; if (err == -EAGAIN) { @@ -451,67 +441,89 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) break; } } + remove_wait_queue(&rmidi->open_wait, &wait); + if (err < 0) { + kfree(rawmidi_file); + goto __error; + } #ifdef CONFIG_SND_OSSEMUL if (rawmidi_file->input && rawmidi_file->input->runtime) rawmidi_file->input->runtime->oss = (maj == SOUND_MAJOR); if (rawmidi_file->output && rawmidi_file->output->runtime) rawmidi_file->output->runtime->oss = (maj == SOUND_MAJOR); #endif - remove_wait_queue(&rmidi->open_wait, &wait); - if (err >= 0) { - file->private_data = rawmidi_file; - } else { - snd_card_file_remove(card, file); - kfree(rawmidi_file); - } + file->private_data = rawmidi_file; + mutex_unlock(&rmidi->open_mutex); + return 0; + + __error: + snd_card_file_remove(card, file); + __error_card: mutex_unlock(&rmidi->open_mutex); + module_put(rmidi->card->module); return err; } -int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile) +static void close_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int cleanup) { - struct snd_rawmidi *rmidi; - struct snd_rawmidi_substream *substream; - struct snd_rawmidi_runtime *runtime; + rmidi->streams[substream->stream].substream_opened--; + if (--substream->use_count) + return; - if (snd_BUG_ON(!rfile)) - return -ENXIO; - rmidi = rfile->rmidi; - mutex_lock(&rmidi->open_mutex); - if (rfile->input != NULL) { - substream = rfile->input; - rfile->input = NULL; - runtime = substream->runtime; - snd_rawmidi_input_trigger(substream, 0); - substream->ops->close(substream); - if (runtime->private_free != NULL) - runtime->private_free(substream); - snd_rawmidi_runtime_free(substream); - substream->opened = 0; - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened--; - } - if (rfile->output != NULL) { - substream = rfile->output; - rfile->output = NULL; - if (--substream->use_count == 0) { - runtime = substream->runtime; + if (cleanup) { + if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT) + snd_rawmidi_input_trigger(substream, 0); + else { if (substream->active_sensing) { unsigned char buf = 0xfe; - /* sending single active sensing message to shut the device up */ + /* sending single active sensing message + * to shut the device up + */ snd_rawmidi_kernel_write(substream, &buf, 1); } if (snd_rawmidi_drain_output(substream) == -ERESTARTSYS) snd_rawmidi_output_trigger(substream, 0); - substream->ops->close(substream); - if (runtime->private_free != NULL) - runtime->private_free(substream); - snd_rawmidi_runtime_free(substream); - substream->opened = 0; - substream->append = 0; } - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened--; } + substream->ops->close(substream); + if (substream->runtime->private_free) + substream->runtime->private_free(substream); + snd_rawmidi_runtime_free(substream); + substream->opened = 0; + substream->append = 0; +} + +static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi *rmidi; + + rmidi = rfile->rmidi; + mutex_lock(&rmidi->open_mutex); + if (rfile->input) { + close_substream(rmidi, rfile->input, 1); + rfile->input = NULL; + } + if (rfile->output) { + close_substream(rmidi, rfile->output, 1); + rfile->output = NULL; + } + rfile->rmidi = NULL; mutex_unlock(&rmidi->open_mutex); + wake_up(&rmidi->open_wait); +} + +/* called from sound/core/seq/seq_midi.c */ +int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi *rmidi; + + if (snd_BUG_ON(!rfile)) + return -ENXIO; + + rmidi = rfile->rmidi; + rawmidi_release_priv(rfile); module_put(rmidi->card->module); return 0; } @@ -520,15 +532,14 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file) { struct snd_rawmidi_file *rfile; struct snd_rawmidi *rmidi; - int err; rfile = file->private_data; - err = snd_rawmidi_kernel_release(rfile); rmidi = rfile->rmidi; - wake_up(&rmidi->open_wait); + rawmidi_release_priv(rfile); kfree(rfile); snd_card_file_remove(rmidi->card, file); - return err; + module_put(rmidi->card->module); + return 0; } static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index bf8d2b4cb15e..c0154a959d55 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -181,7 +181,7 @@ char *enabled_str(int bool); /* for debug */ #ifdef SNDRV_SEQ_OSS_DEBUG extern int seq_oss_debug; -#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printk x; } while (0) +#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printd x; } while (0) #else #define debug_printk(x) /**/ #endif diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 0101a8b99b73..29896ab23403 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -321,7 +321,8 @@ void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp) freeprev = cell; } else { #if 0 - printk("type = %i, source = %i, dest = %i, client = %i\n", + printk(KERN_DEBUG "type = %i, source = %i, dest = %i, " + "client = %i\n", cell->event.type, cell->event.source.client, cell->event.dest.client, diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index d4564edd61d7..4e7ec2b49873 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -38,6 +38,10 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) if (! sgbuf) return -EINVAL; + if (dmab->area) + vunmap(dmab->area); + dmab->area = NULL; + tmpb.dev.type = SNDRV_DMA_TYPE_DEV; tmpb.dev.dev = sgbuf->dev; for (i = 0; i < sgbuf->pages; i++) { @@ -48,9 +52,6 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; snd_dma_free_pages(&tmpb); } - if (dmab->area) - vunmap(dmab->area); - dmab->area = NULL; kfree(sgbuf->table); kfree(sgbuf->page_table); diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 4cc57f902e2c..257624bd1997 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -50,18 +50,38 @@ struct link_slave { struct link_master *master; struct link_ctl_info info; int vals[2]; /* current values */ + unsigned int flags; struct snd_kcontrol slave; /* the copy of original control entry */ }; +static int slave_update(struct link_slave *slave) +{ + struct snd_ctl_elem_value *uctl; + int err, ch; + + uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return -ENOMEM; + uctl->id = slave->slave.id; + err = slave->slave.get(&slave->slave, uctl); + for (ch = 0; ch < slave->info.count; ch++) + slave->vals[ch] = uctl->value.integer.value[ch]; + kfree(uctl); + return 0; +} + /* get the slave ctl info and save the initial values */ static int slave_init(struct link_slave *slave) { struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; - int err, ch; + int err; - if (slave->info.count) - return 0; /* already initialized */ + if (slave->info.count) { + /* already initialized */ + if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE) + return slave_update(slave); + return 0; + } uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); if (!uinfo) @@ -85,15 +105,7 @@ static int slave_init(struct link_slave *slave) slave->info.max_val = uinfo->value.integer.max; kfree(uinfo); - uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); - if (!uctl) - return -ENOMEM; - uctl->id = slave->slave.id; - err = slave->slave.get(&slave->slave, uctl); - for (ch = 0; ch < slave->info.count; ch++) - slave->vals[ch] = uctl->value.integer.value[ch]; - kfree(uctl); - return 0; + return slave_update(slave); } /* initialize master volume */ @@ -229,7 +241,8 @@ static void slave_free(struct snd_kcontrol *kcontrol) * - logarithmic volume control (dB level), no linear volume * - master can only attenuate the volume, no gain */ -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, + unsigned int flags) { struct link_master *master_link = snd_kcontrol_chip(master); struct link_slave *srec; @@ -241,6 +254,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) srec->slave = *slave; memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); srec->master = master_link; + srec->flags = flags; /* override callbacks */ slave->info = slave_info; @@ -254,8 +268,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) list_add_tail(&srec->list, &master_link->slaves); return 0; } - -EXPORT_SYMBOL(snd_ctl_add_slave); +EXPORT_SYMBOL(_snd_ctl_add_slave); /* * ctl callbacks for master controls @@ -327,8 +340,20 @@ static void master_free(struct snd_kcontrol *kcontrol) } -/* - * Create a virtual master control with the given name +/** + * snd_ctl_make_virtual_master - Create a virtual master control + * @name: name string of the control element to create + * @tlv: optional TLV int array for dB information + * + * Creates a virtual matster control with the given name string. + * Returns the created control element, or NULL for errors (ENOMEM). + * + * After creating a vmaster element, you can add the slave controls + * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached(). + * + * 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. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -367,5 +392,4 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, return kctl; } - EXPORT_SYMBOL(snd_ctl_make_virtual_master); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index c3e9833dcfd9..2f8f295d6b0c 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -303,8 +303,10 @@ static void snd_mtpav_output_port_write(struct mtpav *mtp_card, snd_mtpav_send_byte(mtp_card, 0xf5); snd_mtpav_send_byte(mtp_card, portp->hwport); - //snd_printk("new outport: 0x%x\n", (unsigned int) portp->hwport); - + /* + snd_printk(KERN_DEBUG "new outport: 0x%x\n", + (unsigned int) portp->hwport); + */ if (!(outbyte & 0x80) && portp->running_status) snd_mtpav_send_byte(mtp_card, portp->running_status); } @@ -540,7 +542,7 @@ static void snd_mtpav_read_bytes(struct mtpav *mcrd) u8 sbyt = snd_mtpav_getreg(mcrd, SREG); - //printk("snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); + /* printk(KERN_DEBUG "snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); */ if (!(sbyt & SIGS_BYTE)) return; @@ -585,12 +587,12 @@ static irqreturn_t snd_mtpav_irqh(int irq, void *dev_id) static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard) { if ((mcard->res_port = request_region(port, 3, "MotuMTPAV MIDI")) == NULL) { - snd_printk("MTVAP port 0x%lx is busy\n", port); + snd_printk(KERN_ERR "MTVAP port 0x%lx is busy\n", port); return -EBUSY; } mcard->port = port; if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) { - snd_printk("MTVAP IRQ %d busy\n", irq); + snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); return -EBUSY; } mcard->irq = irq; @@ -706,7 +708,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) mtp_card->card = card; mtp_card->irq = -1; mtp_card->share_irq = 0; - mtp_card->inmidiport = 0xffffffff; mtp_card->inmidistate = 0; mtp_card->outmidihwport = 0xffffffff; init_timer(&mtp_card->timer); @@ -719,6 +720,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) if (err < 0) goto __error; + mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST; + err = snd_mtpav_get_ISA(mtp_card); if (err < 0) goto __error; diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 33d9db782e07..9284829bf927 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1015,7 +1015,7 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev) goto __err; } - snd_printk("ESI Miditerminal 4140 on 0x%lx\n", p->base); + snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base); return 0; __err: diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 780582340fef..6e31e46ca393 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -302,7 +302,7 @@ void snd_opl3_interrupt(struct snd_hwdep * hw) opl3 = hw->private_data; status = inb(opl3->l_port); #if 0 - snd_printk("AdLib IRQ status = 0x%x\n", status); + snd_printk(KERN_DEBUG "AdLib IRQ status = 0x%x\n", status); #endif if (!(status & 0x80)) return; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 16feafa2c51e..6e7d09ae0e82 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -125,7 +125,7 @@ static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) { int i; char *str = "x.24"; - printk("time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); + printk(KERN_DEBUG "time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); for (i = 0; i < opl3->max_voices; i++) printk("%c", *(str + opl3->voices[i].state + 1)); printk("\n"); @@ -218,7 +218,7 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, for (i = 0; i < END; i++) { if (best[i].voice >= 0) { #ifdef DEBUG_ALLOC - printk("%s %iop allocation on voice %i\n", + printk(KERN_DEBUG "%s %iop allocation on voice %i\n", alloc_type[i], instr_4op ? 4 : 2, best[i].voice); #endif @@ -317,7 +317,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n", + snd_printk(KERN_DEBUG "Note on, ch %i, inst %i, note %i, vel %i\n", chan->number, chan->midi_program, note, vel); #endif @@ -372,7 +372,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) return; } #ifdef DEBUG_MIDI - snd_printk(" --> OPL%i instrument: %s\n", + snd_printk(KERN_DEBUG " --> OPL%i instrument: %s\n", instr_4op ? 3 : 2, patch->name); #endif /* in SYNTH mode, application takes care of voices */ @@ -431,7 +431,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } #ifdef DEBUG_MIDI - snd_printk(" --> setting OPL3 connection: 0x%x\n", + snd_printk(KERN_DEBUG " --> setting OPL3 connection: 0x%x\n", opl3->connection_reg); #endif /* @@ -466,7 +466,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* Program the FM voice characteristics */ for (i = 0; i < (instr_4op ? 4 : 2); i++) { #ifdef DEBUG_MIDI - snd_printk(" --> programming operator %i\n", i); + snd_printk(KERN_DEBUG " --> programming operator %i\n", i); #endif op_offset = snd_opl3_regmap[voice_offset][i]; @@ -546,7 +546,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) blocknum |= OPL3_KEYON_BIT; #ifdef DEBUG_MIDI - snd_printk(" --> trigger voice %i\n", voice); + snd_printk(KERN_DEBUG " --> trigger voice %i\n", voice); #endif /* Set OPL3 KEYON_BLOCK register of requested voice */ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); @@ -602,7 +602,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) prg = extra_prg - 1; } #ifdef DEBUG_MIDI - snd_printk(" *** allocating extra program\n"); + snd_printk(KERN_DEBUG " *** allocating extra program\n"); #endif goto __extra_prg; } @@ -633,7 +633,7 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* kill voice */ #ifdef DEBUG_MIDI - snd_printk(" --> kill voice %i\n", voice); + snd_printk(KERN_DEBUG " --> kill voice %i\n", voice); #endif opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); /* clear Key ON bit */ @@ -670,7 +670,7 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Note off, ch %i, inst %i, note %i\n", + snd_printk(KERN_DEBUG "Note off, ch %i, inst %i, note %i\n", chan->number, chan->midi_program, note); #endif @@ -709,7 +709,7 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Key pressure, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -723,7 +723,7 @@ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Terminate note, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -812,7 +812,7 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Controller, TYPE = %i, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Controller, TYPE = %i, ch#: %i, inst#: %i\n", type, chan->number, chan->midi_program); #endif @@ -849,7 +849,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, opl3 = p; #ifdef DEBUG_MIDI - snd_printk("NRPN, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -864,6 +864,6 @@ void snd_opl3_sysex(void *p, unsigned char *buf, int len, opl3 = p; #ifdef DEBUG_MIDI - snd_printk("SYSEX\n"); + snd_printk(KERN_DEBUG "SYSEX\n"); #endif } diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 9a2271dc046a..a54b1dc5cc78 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -220,14 +220,14 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -EINVAL; if (count < (int)sizeof(sbi)) { - snd_printk("FM Error: Patch record too short\n"); + snd_printk(KERN_ERR "FM Error: Patch record too short\n"); return -EINVAL; } if (copy_from_user(&sbi, buf, sizeof(sbi))) return -EFAULT; if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk("FM Error: Invalid instrument number %d\n", + snd_printk(KERN_ERR "FM Error: Invalid instrument number %d\n", sbi.channel); return -EINVAL; } @@ -254,7 +254,9 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: - snd_printk("OPL3: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n"); + snd_printk(KERN_ERR "OPL3: " + "Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. " + "Fix the program.\n"); return -EINVAL; case SNDCTL_SYNTH_MEMAVL: diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 962bb9c8b9c8..6d57b6441dec 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -168,7 +168,7 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, #ifdef CONFIG_SND_DEBUG default: - snd_printk("unknown IOCTL: 0x%x\n", cmd); + snd_printk(KERN_WARNING "unknown IOCTL: 0x%x\n", cmd); #endif } return -ENOTTY; diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index aa2ae07a76d5..b60cef257b58 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -57,7 +57,7 @@ static int __devinit snd_pcsp_create(struct snd_card *card) else min_div = MAX_DIV; #if PCSP_DEBUG - printk("PCSP: lpj=%li, min_div=%i, res=%li\n", + printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%li\n", loops_per_jiffy, min_div, tp.tv_nsec); #endif diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 891d081e4825..b2b6d50c9425 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -241,7 +241,8 @@ static void snd_uart16550_io_loop(struct snd_uart16550 * uart) snd_rawmidi_receive(uart->midi_input[substream], &c, 1); if (status & UART_LSR_OE) - snd_printk("%s: Overrun on device at 0x%lx\n", + snd_printk(KERN_WARNING + "%s: Overrun on device at 0x%lx\n", uart->rmidi->name, uart->base); } @@ -636,7 +637,8 @@ static int snd_uart16550_output_byte(struct snd_uart16550 *uart, } } else { if (!snd_uart16550_write_buffer(uart, midi_byte)) { - snd_printk("%s: Buffer overrun on device at 0x%lx\n", + snd_printk(KERN_WARNING + "%s: Buffer overrun on device at 0x%lx\n", uart->rmidi->name, uart->base); return 0; } @@ -815,7 +817,8 @@ static int __devinit snd_uart16550_create(struct snd_card *card, if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (request_irq(irq, snd_uart16550_interrupt, IRQF_DISABLED, "Serial MIDI", uart)) { - snd_printk("irq %d busy. Using Polling.\n", irq); + snd_printk(KERN_WARNING + "irq %d busy. Using Polling.\n", irq); } else { uart->irq = irq; } @@ -919,19 +922,22 @@ static int __devinit snd_serial_probe(struct platform_device *devptr) case SNDRV_SERIAL_GENERIC: break; default: - snd_printk("Adaptor type is out of range 0-%d (%d)\n", + snd_printk(KERN_ERR + "Adaptor type is out of range 0-%d (%d)\n", SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]); return -ENODEV; } if (outs[dev] < 1 || outs[dev] > SNDRV_SERIAL_MAX_OUTS) { - snd_printk("Count of outputs is out of range 1-%d (%d)\n", + snd_printk(KERN_ERR + "Count of outputs is out of range 1-%d (%d)\n", SNDRV_SERIAL_MAX_OUTS, outs[dev]); return -ENODEV; } if (ins[dev] < 1 || ins[dev] > SNDRV_SERIAL_MAX_INS) { - snd_printk("Count of inputs is out of range 1-%d (%d)\n", + snd_printk(KERN_ERR + "Count of inputs is out of range 1-%d (%d)\n", SNDRV_SERIAL_MAX_INS, ins[dev]); return -ENODEV; } diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 6f48711818f3..0e631c3221e3 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -98,7 +98,9 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr) vmidi->card = card; if (midi_devs[dev] > MAX_MIDI_DEVICES) { - snd_printk("too much midi devices for virmidi %d: force to use %d\n", dev, MAX_MIDI_DEVICES); + snd_printk(KERN_WARNING + "too much midi devices for virmidi %d: " + "force to use %d\n", dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 14e3354be43a..19c6e376c7c7 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -688,7 +688,8 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) image = dsp->data + i; /* Wait DSP ready for a new read */ if ((err = vx_wait_isr_bit(chip, ISR_TX_EMPTY)) < 0) { - printk("dsp loading error at position %d\n", i); + printk(KERN_ERR + "dsp loading error at position %d\n", i); return err; } cptr = image; diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 8d6362e2d4c9..46df8817c18f 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -119,16 +119,6 @@ void snd_vx_free_firmware(struct vx_core *chip) #else /* old style firmware loading */ -static int vx_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int vx_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int vx_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -243,8 +233,6 @@ int snd_vx_setup_firmware(struct vx_core *chip) hw->iface = SNDRV_HWDEP_IFACE_VX; hw->private_data = chip; - hw->ops.open = vx_hwdep_open; - hw->ops.release = vx_hwdep_release; hw->ops.dsp_status = vx_hwdep_dsp_status; hw->ops.dsp_load = vx_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c index 0e1ba9b47904..b0560fec6bba 100644 --- a/sound/drivers/vx/vx_uer.c +++ b/sound/drivers/vx/vx_uer.c @@ -103,7 +103,7 @@ static void vx_write_one_cbit(struct vx_core *chip, int index, int val) * returns the frequency of UER, or 0 if not sync, * or a negative error code. */ -static int vx_read_uer_status(struct vx_core *chip, int *mode) +static int vx_read_uer_status(struct vx_core *chip, unsigned int *mode) { int val, freq; diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 37970666a453..36879bf88700 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -obj-$(CONFIG_L3) += l3/ - obj-$(CONFIG_SND) += other/ # Toplevel Module Dependency diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile deleted file mode 100644 index 49455b8dcc04..000000000000 --- a/sound/i2c/l3/Makefile +++ /dev/null @@ -1,8 +0,0 @@ -# -# Makefile for ALSA -# - -snd-uda1341-objs := uda1341.o - -# Module Dependency -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c deleted file mode 100644 index 9840eb43648d..000000000000 --- a/sound/i2c/l3/uda1341.c +++ /dev/null @@ -1,935 +0,0 @@ -/* - * Philips UDA1341 mixer device driver - * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz> - * - * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS - * 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble) - * 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP - * features support - * 2002-04-12 Tomas Kasparek proc interface update, code cleanup - * 2002-05-12 Tomas Kasparek another code cleanup - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/types.h> -#include <linux/slab.h> -#include <linux/errno.h> -#include <linux/ioctl.h> - -#include <asm/uaccess.h> - -#include <sound/core.h> -#include <sound/control.h> -#include <sound/initval.h> -#include <sound/info.h> - -#include <linux/l3/l3.h> - -#include <sound/uda1341.h> - -/* {{{ HW regs definition */ - -#define STAT0 0x00 -#define STAT1 0x80 -#define STAT_MASK 0x80 - -#define DATA0_0 0x00 -#define DATA0_1 0x40 -#define DATA0_2 0x80 -#define DATA_MASK 0xc0 - -#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2) -#define IS_DATA1(x) ((x) == data1) -#define IS_STATUS(x) ((x) == stat0 || (x) == stat1) -#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6) - -/* }}} */ - - -static const char *peak_names[] = { - "before", - "after", -}; - -static const char *filter_names[] = { - "flat", - "min", - "min", - "max", -}; - -static const char *mixer_names[] = { - "double differential", - "input channel 1 (line in)", - "input channel 2 (microphone)", - "digital mixer", -}; - -static const char *deemp_names[] = { - "none", - "32 kHz", - "44.1 kHz", - "48 kHz", -}; - -enum uda1341_regs_names { - stat0, - stat1, - data0_0, - data0_1, - data0_2, - data1, - ext0, - ext1, - ext2, - empty, - ext4, - ext5, - ext6, - uda1341_reg_last, -}; - -static const char *uda1341_reg_names[] = { - "stat 0 ", - "stat 1 ", - "data 00", - "data 01", - "data 02", - "data 1 ", - "ext 0", - "ext 1", - "ext 2", - "empty", - "ext 4", - "ext 5", - "ext 6", -}; - -static const int uda1341_enum_items[] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 2, //peak - before/after - 4, //deemp - none/32/44.1/48 - 0, - 4, //filter - flat/min/min/max - 0, 0, 0, - 4, //mixer - differ/line/mic/mixer - 0, 0, 0, 0, 0, -}; - -static const char ** uda1341_enum_names[] = { - NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, - peak_names, //peak - before/after - deemp_names, //deemp - none/32/44.1/48 - NULL, - filter_names, //filter - flat/min/min/max - NULL, NULL, NULL, - mixer_names, //mixer - differ/line/mic/mixer - NULL, NULL, NULL, NULL, NULL, -}; - -typedef int uda1341_cfg[CMD_LAST]; - -struct uda1341 { - int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val); - int (*read) (struct l3_client *uda1341, unsigned short reg); - unsigned char regs[uda1341_reg_last]; - int active; - spinlock_t reg_lock; - struct snd_card *card; - uda1341_cfg cfg; -#ifdef CONFIG_PM - unsigned char suspend_regs[uda1341_reg_last]; - uda1341_cfg suspend_cfg; -#endif -}; - -/* transfer 8bit integer into string with binary representation */ -static void int2str_bin8(uint8_t val, char *buf) -{ - const int size = sizeof(val) * 8; - int i; - - for (i= 0; i < size; i++){ - *(buf++) = (val >> (size - 1)) ? '1' : '0'; - val <<= 1; - } - *buf = '\0'; //end the string with zero -} - -/* {{{ HW manipulation routines */ - -static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val) -{ - struct uda1341 *uda = clnt->driver_data; - unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing - int err = 0; - - uda->regs[reg] = val; - - if (uda->active) { - if (IS_DATA0(reg)) { - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1); - } else if (IS_DATA1(reg)) { - err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1); - } else if (IS_STATUS(reg)) { - err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1); - } else if (IS_EXTEND(reg)) { - buf[0] |= (reg - ext0) & 0x7; //EXT address - buf[1] |= val; //EXT data - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2); - } - } else - printk(KERN_ERR "UDA1341 codec not active!\n"); - return err; -} - -static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg) -{ - unsigned char val; - int err; - - err = l3_read(clnt, reg, &val, 1); - if (err == 1) - // use just 6bits - the rest is address of the reg - return val & 63; - return err < 0 ? err : -EIO; -} - -static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg) -{ - return reg < uda1341_reg_last; -} - -static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg, - unsigned short mask, unsigned short shift, - unsigned short value, int flush) -{ - int change; - unsigned short old, new; - struct uda1341 *uda = clnt->driver_data; - -#if 0 - printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n", - uda1341_reg_names[reg], mask, shift, value); -#endif - - if (!snd_uda1341_valid_reg(clnt, reg)) - return -EINVAL; - spin_lock(&uda->reg_lock); - old = uda->regs[reg]; - new = (old & ~(mask << shift)) | (value << shift); - change = old != new; - if (change) { - if (flush) uda->write(clnt, reg, new); - uda->regs[reg] = new; - } - spin_unlock(&uda->reg_lock); - return change; -} - -static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what, - unsigned short value, int flush) -{ - struct uda1341 *uda = clnt->driver_data; - int ret = 0; -#ifdef CONFIG_PM - int reg; -#endif - -#if 0 - printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value); -#endif - - uda->cfg[what] = value; - - switch(what) { - case CMD_RESET: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE - uda->cfg[CMD_RESET]=0; - break; - case CMD_FS: - ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush); - break; - case CMD_FORMAT: - ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush); - break; - case CMD_OGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush); - break; - case CMD_IGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush); - break; - case CMD_DAC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush); - break; - case CMD_ADC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush); - break; - case CMD_VOLUME: - ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush); - break; - case CMD_BASS: - ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush); - break; - case CMD_TREBBLE: - ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush); - break; - case CMD_PEAK: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush); - break; - case CMD_DEEMP: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush); - break; - case CMD_MUTE: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush); - break; - case CMD_FILTER: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush); - break; - case CMD_CH1: - ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush); - break; - case CMD_CH2: - ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush); - break; - case CMD_MIC: - ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush); - break; - case CMD_MIXER: - ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush); - break; - case CMD_AGC: - ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush); - break; - case CMD_IG: - ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush); - ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush); - break; - case CMD_AGC_TIME: - ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush); - break; - case CMD_AGC_LEVEL: - ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush); - break; -#ifdef CONFIG_PM - case CMD_SUSPEND: - for (reg = stat0; reg < uda1341_reg_last; reg++) - uda->suspend_regs[reg] = uda->regs[reg]; - for (reg = 0; reg < CMD_LAST; reg++) - uda->suspend_cfg[reg] = uda->cfg[reg]; - break; - case CMD_RESUME: - for (reg = stat0; reg < uda1341_reg_last; reg++) - snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]); - for (reg = 0; reg < CMD_LAST; reg++) - uda->cfg[reg] = uda->suspend_cfg[reg]; - break; -#endif - default: - ret = -EINVAL; - break; - } - - if (!uda->active) - printk(KERN_ERR "UDA1341 codec not active!\n"); - return ret; -} - -/* }}} */ - -/* {{{ Proc interface */ -#ifdef CONFIG_PROC_FS - -static const char *format_names[] = { - "I2S-bus", - "LSB 16bits", - "LSB 18bits", - "LSB 20bits", - "MSB", - "in LSB 16bits/out MSB", - "in LSB 18bits/out MSB", - "in LSB 20bits/out MSB", -}; - -static const char *fs_names[] = { - "512*fs", - "384*fs", - "256*fs", - "Unused - bad value!", -}; - -static const char* bass_values[][16] = { - {"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", - "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB", - "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max -}; - -static const char *mic_sens_value[] = { - "-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used", -}; - -static const unsigned short AGC_atime[] = { - 11, 16, 11, 16, 21, 11, 16, 21, -}; - -static const unsigned short AGC_dtime[] = { - 100, 100, 200, 200, 200, 400, 400, 400, -}; - -static const char *AGC_level[] = { - "-9.0", "-11.5", "-15.0", "-17.5", -}; - -static const char *ig_small_value[] = { - "-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5", -}; - -/* - * this was computed as peak_value[i] = pow((63-i)*1.42,1.013) - * - * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2 - * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29 - * [61]=-2.78, [62] = -1.48, [63] = 0.0 - * I tried to compute it, but using but even using logarithm with base either 10 or 2 - * i was'n able to get values in the table from the formula. So I constructed another - * formula (see above) to interpolate the values as good as possible. If there is some - * mistake, please contact me on tomas.kasparek@seznam.cz. Thanks. - * UDA1341TS datasheet is available at: - * http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf - */ -static const char *peak_value[] = { - "-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB", - "-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB", - "-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB", - "-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB", - "-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB", - "-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB", - "-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB", - "-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB", - "-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB", -}; - -static void snd_uda1341_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int peak; - - peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1); - if (peak < 0) - peak = 0; - - snd_iprintf(buffer, "%s\n\n", uda->card->longname); - - // for information about computed values see UDA1341TS datasheet pages 15 - 21 - snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off"); - snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off"); - snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]); - snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]); - - snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]); - snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]); - snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]); - snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before"); - snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]); - - snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off"); - snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]); - - snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off"); - - if (uda->cfg[CMD_VOLUME] == 0) - snd_iprintf(buffer, "Volume : 0 dB\n"); - else if (uda->cfg[CMD_VOLUME] < 62) - snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1); - else - snd_iprintf(buffer, "Volume : -INF dB\n"); - snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]); - snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0); - snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]); - - - if(uda->cfg[CMD_CH1] < 31) - snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n", - ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1), - uda->cfg[CMD_CH1] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n"); - if(uda->cfg[CMD_CH2] < 31) - snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n", - ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1), - uda->cfg[CMD_CH2] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n"); - - if(uda->cfg[CMD_IG] > 5) - snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n", - (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]); -} - -static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int reg; - char buf[12]; - - for (reg = 0; reg < uda1341_reg_last; reg ++) { - if (reg == empty) - continue; - int2str_bin8(uda->regs[reg], buf); - snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf); - } - - int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf); - snd_iprintf(buffer, "DATA1 = %s\n", buf); -} -#endif /* CONFIG_PROC_FS */ - -static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt) -{ - struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "uda1341", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read); - if (! snd_card_proc_new(card, "uda1341-regs", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read); -} - -/* }}} */ - -/* {{{ Mixer controls setting */ - -/* {{{ UDA1341 single functions */ - -#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \ - .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 12) & 63; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - - uda->cfg[where] = val; - return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 enum functions */ - -#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \ - .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int where = kcontrol->private_value & 31; - const char **texts; - - // this register we don't handle this way - if (!uda1341_enum_items[where]) - return -EINVAL; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = uda1341_enum_items[where]; - - if (uinfo->value.enumerated.item >= uda1341_enum_items[where]) - uinfo->value.enumerated.item = uda1341_enum_items[where] - 1; - - texts = uda1341_enum_names[where]; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - - ucontrol->value.enumerated.item[0] = uda->cfg[where]; - return 0; -} - -static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - - uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask); - - return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 2regs functions */ - -#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \ - .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \ - .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \ - (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \ -} - - -static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg_1 = (kcontrol->private_value >> 5) & 15; - int reg_2 = (kcontrol->private_value >> 9) & 15; - int shift_1 = (kcontrol->private_value >> 13) & 7; - int shift_2 = (kcontrol->private_value >> 16) & 7; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - unsigned short val1, val2, val; - - val = ucontrol->value.integer.value[0]; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - val1 = val & mask_1; - val2 = (val / (mask_1 + 1)) & mask_2; - - if (invert) { - val1 = mask_1 - val1; - val2 = mask_2 - val2; - } - - uda->cfg[where] = invert ? mask - val : val; - - //FIXME - return value - snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH); - return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH); -} - -/* }}} */ - -static struct snd_kcontrol_new snd_uda1341_controls[] = { - UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1), - UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1), - - UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0), - UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0), - - UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0), - UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0), - - UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1), - UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1), - - UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0), - - UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0), - UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0), - UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0), - - UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0), - UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0), - - UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0), - UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0), - UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0), - UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0), - - UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0), -}; - -static void uda1341_free(struct l3_client *clnt) -{ - l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341) - kfree(clnt); -} - -static int uda1341_dev_free(struct snd_device *device) -{ - struct l3_client *clnt = device->device_data; - uda1341_free(clnt); - return 0; -} - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp) -{ - static struct snd_device_ops ops = { - .dev_free = uda1341_dev_free, - }; - struct l3_client *clnt; - int idx, err; - - if (snd_BUG_ON(!card)) - return -EINVAL; - - clnt = kzalloc(sizeof(*clnt), GFP_KERNEL); - if (clnt == NULL) - return -ENOMEM; - - if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) { - kfree(clnt); - return err; - } - - for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) { - uda1341_free(clnt); - return err; - } - } - - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) { - uda1341_free(clnt); - return err; - } - - *clntp = clnt; - strcpy(card->mixername, "UDA1341TS Mixer"); - ((struct uda1341 *)clnt->driver_data)->card = card; - - snd_uda1341_proc_init(card, clnt); - - return 0; -} - -/* }}} */ - -/* {{{ L3 operations */ - -static int uda1341_attach(struct l3_client *clnt) -{ - struct uda1341 *uda; - - uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL); - if (!uda) - return -ENOMEM; - - /* init fixed parts of my copy of registers */ - uda->regs[stat0] = STAT0; - uda->regs[stat1] = STAT1; - - uda->regs[data0_0] = DATA0_0; - uda->regs[data0_1] = DATA0_1; - uda->regs[data0_2] = DATA0_2; - - uda->write = snd_uda1341_codec_write; - uda->read = snd_uda1341_codec_read; - - spin_lock_init(&uda->reg_lock); - - clnt->driver_data = uda; - return 0; -} - -static void uda1341_detach(struct l3_client *clnt) -{ - kfree(clnt->driver_data); -} - -static int -uda1341_command(struct l3_client *clnt, int cmd, void *arg) -{ - if (cmd != CMD_READ_REG) - return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH); - - return snd_uda1341_codec_read(clnt, (int) arg); -} - -static int uda1341_open(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 1; - - /* init default configuration */ - snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY); - snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset - snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset - //at this moment should be QMUTED by h3600_audio_init - snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset - snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode - snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset - - return 0; -} - -static void uda1341_close(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 0; -} - -/* }}} */ - -/* {{{ Module and L3 initialization */ - -static struct l3_ops uda1341_ops = { - .open = uda1341_open, - .command = uda1341_command, - .close = uda1341_close, -}; - -static struct l3_driver uda1341_driver = { - .name = UDA1341_ALSA_NAME, - .attach_client = uda1341_attach, - .detach_client = uda1341_detach, - .ops = &uda1341_ops, - .owner = THIS_MODULE, -}; - -static int __init uda1341_init(void) -{ - return l3_add_driver(&uda1341_driver); -} - -static void __exit uda1341_exit(void) -{ - l3_del_driver(&uda1341_driver); -} - -module_init(uda1341_init); -module_exit(uda1341_exit); - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); - -EXPORT_SYMBOL(snd_chip_uda1341_mixer_new); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index a74725950b02..2df20e403f24 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -56,8 +56,8 @@ config SND_AD1848 Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. - For newer chips from Cirrus Logic, use the CS4231, CS4232 or - CS4236+ drivers. + For newer chips from Cirrus Logic, use the CS4231 or CS4232+ + drivers. To compile this driver as a module, choose M here: the module will be called snd-ad1848. @@ -94,6 +94,8 @@ config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB select SND_SB16_DSP + select SND_OPL3_LIB + select SND_MPU401_UART help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. @@ -112,26 +114,15 @@ config SND_CS4231 To compile this driver as a module, choose M here: the module will be called snd-cs4231. -config SND_CS4232 - tristate "Generic Cirrus Logic CS4232 driver" - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_WSS_LIB - help - Say Y here to include support for CS4232 chips from Cirrus - Logic - Crystal Semiconductors. - - To compile this driver as a module, choose M here: the module - will be called snd-cs4232. - config SND_CS4236 - tristate "Generic Cirrus Logic CS4236+ driver" + tristate "Generic Cirrus Logic CS4232/CS4236+ driver" select SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help - Say Y to include support for CS4235,CS4236,CS4237B,CS4238B, - CS4239 chips from Cirrus Logic - Crystal Semiconductors. + Say Y to include support for CS4232,CS4235,CS4236,CS4237B, + CS4238B,CS4239 chips from Cirrus Logic - Crystal + Semiconductors. To compile this driver as a module, choose M here: the module will be called snd-cs4236. diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 9660e598232c..bbcbf92a8ebe 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -156,6 +156,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; + struct snd_timer *timer; error = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_card_ad1816a), &card); @@ -195,6 +196,12 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard return error; } + error = snd_ad1816a_timer(chip, 0, &timer); + if (error < 0) { + snd_card_free(card); + return error; + } + if (mpu_port[dev] > 0) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, @@ -208,11 +215,8 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard OPL3_HW_AUTO, 0, &opl3) < 0) { printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2); } else { - if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 3bfca7c59baf..05aef8b97e96 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -37,7 +37,7 @@ static inline int snd_ad1816a_busy_wait(struct snd_ad1816a *chip) if (inb(AD1816A_REG(AD1816A_CHIP_STATUS)) & AD1816A_READY) return 0; - snd_printk("chip busy.\n"); + snd_printk(KERN_WARNING "chip busy.\n"); return -EBUSY; } @@ -196,7 +196,7 @@ static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what, spin_unlock(&chip->lock); break; default: - snd_printk("invalid trigger mode 0x%x.\n", what); + snd_printk(KERN_WARNING "invalid trigger mode 0x%x.\n", what); error = -EINVAL; } @@ -377,7 +377,6 @@ static struct snd_pcm_hardware snd_ad1816a_capture = { .fifo_size = 0, }; -#if 0 /* not used now */ static int snd_ad1816a_timer_close(struct snd_timer *timer) { struct snd_ad1816a *chip = snd_timer_chip(timer); @@ -442,8 +441,6 @@ static struct snd_timer_hardware snd_ad1816a_timer_table = { .start = snd_ad1816a_timer_start, .stop = snd_ad1816a_timer_stop, }; -#endif /* not used now */ - static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream) { @@ -568,7 +565,7 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip) case AD1816A_HW_AD1815: return "AD1815"; case AD1816A_HW_AD18MAX10: return "AD18max10"; default: - snd_printk("Unknown chip version %d:%d.\n", + snd_printk(KERN_WARNING "Unknown chip version %d:%d.\n", chip->version, chip->hardware); return "AD1816A - unknown"; } @@ -687,7 +684,6 @@ int __devinit snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_p return 0; } -#if 0 /* not used now */ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer) { struct snd_timer *timer; @@ -709,7 +705,6 @@ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd *rtimer = timer; return 0; } -#endif /* not used now */ /* * diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 24e60902f8ca..de83608719ea 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -31,11 +31,11 @@ * To quickly load the module, * * modprobe -a snd-cmi8330 sbport=0x220 sbirq=5 sbdma8=1 - * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 + * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 fmport=0x388 * * 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 AD1848 codec, + * The driver "magically" routes the capturing to the CMI8330 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 @@ -51,6 +51,8 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/wss.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> #include <sound/sb.h> #include <sound/initval.h> @@ -79,6 +81,9 @@ static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +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."); @@ -107,6 +112,12 @@ MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver."); module_param_array(wssdma, int, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); +module_param_array(fmport, long, NULL, 0444); +MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -149,6 +160,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; struct pnp_dev *play; + struct pnp_dev *mpu; #endif struct snd_card *card; struct snd_wss *wss; @@ -165,7 +177,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP static struct pnp_card_device_id snd_cmi8330_pnpids[] = { - { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" } } }, + { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, { .id = "" } }; @@ -219,8 +231,10 @@ WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), WSS_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), -WSS_SINGLE("FM Playback Switch", 0, - CMI8330_RECMUX, 3, 1, 1), +WSS_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0, CMI8330_RMUX3D, 7, 1, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0, @@ -323,16 +337,21 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, if (acard->play == NULL) return -EBUSY; + acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (acard->play == NULL) + return -EBUSY; + pdev = acard->cap; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D 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); /* allocate SB16 resources */ pdev = acard->play; @@ -347,6 +366,17 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + /* 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; + } + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif @@ -489,6 +519,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) { struct snd_cmi8330 *acard; int i, err; + struct snd_opl3 *opl3; acard = card->private_data; err = snd_wss_create(card, wssport[dev] + 4, -1, @@ -496,11 +527,11 @@ 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 "(AD1848) device busy??\n"); + snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); + snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); return -ENODEV; } @@ -532,6 +563,27 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) snd_printk(KERN_ERR PFX "failed to create pcms\n"); return err; } + if (fmport[dev] != SNDRV_AUTO_PORT) { + if (snd_opl3_create(card, + fmport[dev], fmport[dev] + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR PFX + "no OPL device at 0x%lx-0x%lx ?\n", + fmport[dev], fmport[dev] + 2); + } else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + return err; + } + } + + if (mpuport[dev] != SNDRV_AUTO_PORT) { + if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport[dev], 0, mpuirq[dev], + IRQF_DISABLED, NULL) < 0) + printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", + mpuport[dev]); + } strcpy(card->driver, "CMI8330/C3D"); strcpy(card->shortname, "C-Media CMI8330/C3D"); diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 5870ca21ab59..6d397e8d54ac 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -3,13 +3,11 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # -snd-cs4236-lib-objs := cs4236_lib.o snd-cs4231-objs := cs4231.o -snd-cs4232-objs := cs4232.o -snd-cs4236-objs := cs4236.o +snd-cs4236-objs := cs4236.o cs4236_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o -obj-$(CONFIG_SND_CS4232) += snd-cs4232.o -obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o +obj-$(CONFIG_SND_CS4236) += snd-cs4236.o + diff --git a/sound/isa/cs423x/cs4232.c b/sound/isa/cs423x/cs4232.c deleted file mode 100644 index 9fad2e6c0c2c..000000000000 --- a/sound/isa/cs423x/cs4232.c +++ /dev/null @@ -1,2 +0,0 @@ -#define CS4232 -#include "cs4236.c" diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index f7845986f467..a076a6ce8071 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -33,17 +33,14 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); -#ifdef CS4232 -MODULE_DESCRIPTION("Cirrus Logic CS4232"); +MODULE_DESCRIPTION("Cirrus Logic CS4232-9"); MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000}," "{Turtle Beach,Tropez Plus}," "{SIC CrystalWave 32}," "{Hewlett Packard,Omnibook 5500}," "{TerraTec,Maestro 32/96}," - "{Philips,PCA70PS}}"); -#else -MODULE_DESCRIPTION("Cirrus Logic CS4235-9"); -MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," + "{Philips,PCA70PS}}," + "{{Crystal Semiconductors,CS4235}," "{Crystal Semiconductors,CS4236}," "{Crystal Semiconductors,CS4237}," "{Crystal Semiconductors,CS4238}," @@ -70,15 +67,11 @@ MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," "{Typhoon Soundsystem,CS4236B}," "{Turtle Beach,Malibu}," "{Unknown,Digital PC 5000 Onboard}}"); -#endif -#ifdef CS4232 -#define IDENT "CS4232" -#define DEV_NAME "cs4232" -#else -#define IDENT "CS4236+" -#define DEV_NAME "cs4236" -#endif +MODULE_ALIAS("snd_cs4232"); + +#define IDENT "CS4232+" +#define DEV_NAME "cs4232+" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -128,9 +121,7 @@ MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP static int isa_registered; static int pnpc_registered; -#ifdef CS4232 static int pnp_registered; -#endif #endif /* CONFIG_PNP */ struct snd_card_cs4236 { @@ -145,11 +136,10 @@ struct snd_card_cs4236 { #ifdef CONFIG_PNP -#ifdef CS4232 /* * PNP BIOS */ -static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { +static const struct pnp_device_id snd_cs423x_pnpbiosids[] = { { .id = "CSC0100" }, { .id = "CSC0000" }, /* Guillemot Turtlebeach something appears to be cs4232 compatible @@ -157,10 +147,8 @@ static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { { .id = "GIM0100" }, { .id = "" } }; -MODULE_DEVICE_TABLE(pnp, snd_cs4232_pnpbiosids); -#endif /* CS4232 */ +MODULE_DEVICE_TABLE(pnp, snd_cs423x_pnpbiosids); -#ifdef CS4232 #define CS423X_ISAPNP_DRIVER "cs4232_isapnp" static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Philips PCA70PS */ @@ -179,12 +167,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSCf032", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Netfinity 3000 on-board soundcard */ { .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC010f" } } }, - /* --- */ - { .id = "" } /* end */ -}; -#else /* CS4236 */ -#define CS423X_ISAPNP_DRIVER "cs4236_isapnp" -static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Intel Marlin Spike Motherboard - CS4235 */ { .id = "CSC0225", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Intel Marlin Spike Motherboard (#2) - CS4235 */ @@ -266,7 +248,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* --- */ { .id = "" } /* end */ }; -#endif MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); @@ -323,17 +304,19 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) return 0; } -#ifdef CS4232 -static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, - struct pnp_dev *pdev) +static int __devinit snd_card_cs423x_pnp(int dev, struct snd_card_cs4236 *acard, + struct pnp_dev *pdev, + struct pnp_dev *cdev) { acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; - cport[dev] = -1; + if (cdev) + cport[dev] = pnp_port_start(cdev, 0); + else + cport[dev] = -1; return 0; } -#endif static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard, struct pnp_card_link *card, @@ -411,40 +394,39 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } -#ifdef CS4232 err = snd_wss_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_wss_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_wss_mixer(chip); + WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; - -#else /* CS4236 */ - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_cs4236_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_cs4236_mixer(chip); - if (err < 0) - return err; -#endif + if (chip->hardware & WSS_HW_CS4236B_MASK) { + snd_wss_free(chip); + err = snd_cs4236_create(card, + port[dev], cport[dev], + irq[dev], dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) + return err; + acard->chip = chip; + + err = snd_cs4236_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_cs4236_mixer(chip); + if (err < 0) + return err; + } else { + acard->chip = chip; + err = snd_wss_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_wss_mixer(chip); + if (err < 0) + return err; + } strcpy(card->driver, pcm->name); strcpy(card->shortname, pcm->name); sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", @@ -579,13 +561,14 @@ static struct isa_driver cs423x_isa_driver = { #ifdef CONFIG_PNP -#ifdef CS4232 -static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, +static int __devinit snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, const struct pnp_device_id *id) { static int dev; int err; struct snd_card *card; + struct pnp_dev *cdev; + char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) return -ENOENT; /* we have another procedure - card */ @@ -596,10 +579,19 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; + /* prepare second id */ + strcpy(cid, pdev->id[0].id); + cid[5] = '1'; + cdev = NULL; + list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(cdev->id[0].id, cid)) + break; + } err = snd_cs423x_card_new(dev, &card); if (err < 0) return err; - if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) { + err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); + if (err < 0) { printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n"); snd_card_free(card); return err; @@ -614,35 +606,34 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, return 0; } -static void __devexit snd_cs4232_pnp_remove(struct pnp_dev * pdev) +static void __devexit snd_cs423x_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM -static int snd_cs4232_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) +static int snd_cs423x_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) { return snd_cs423x_suspend(pnp_get_drvdata(pdev)); } -static int snd_cs4232_pnp_resume(struct pnp_dev *pdev) +static int snd_cs423x_pnp_resume(struct pnp_dev *pdev) { return snd_cs423x_resume(pnp_get_drvdata(pdev)); } #endif -static struct pnp_driver cs4232_pnp_driver = { - .name = "cs4232-pnpbios", - .id_table = snd_cs4232_pnpbiosids, - .probe = snd_cs4232_pnpbios_detect, - .remove = __devexit_p(snd_cs4232_pnp_remove), +static struct pnp_driver cs423x_pnp_driver = { + .name = "cs423x-pnpbios", + .id_table = snd_cs423x_pnpbiosids, + .probe = snd_cs423x_pnpbios_detect, + .remove = __devexit_p(snd_cs423x_pnp_remove), #ifdef CONFIG_PM - .suspend = snd_cs4232_pnp_suspend, - .resume = snd_cs4232_pnp_resume, + .suspend = snd_cs423x_pnp_suspend, + .resume = snd_cs423x_pnp_resume, #endif }; -#endif /* CS4232 */ static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard, const struct pnp_card_device_id *pid) @@ -716,18 +707,14 @@ static int __init alsa_card_cs423x_init(void) #ifdef CONFIG_PNP if (!err) isa_registered = 1; -#ifdef CS4232 - err = pnp_register_driver(&cs4232_pnp_driver); + err = pnp_register_driver(&cs423x_pnp_driver); if (!err) pnp_registered = 1; -#endif err = pnp_register_card_driver(&cs423x_pnpc_driver); if (!err) pnpc_registered = 1; -#ifdef CS4232 if (pnp_registered) err = 0; -#endif if (isa_registered) err = 0; #endif @@ -739,10 +726,8 @@ static void __exit alsa_card_cs423x_exit(void) #ifdef CONFIG_PNP if (pnpc_registered) pnp_unregister_card_driver(&cs423x_pnpc_driver); -#ifdef CS4232 if (pnp_registered) - pnp_unregister_driver(&cs4232_pnp_driver); -#endif + pnp_unregister_driver(&cs423x_pnp_driver); if (isa_registered) #endif isa_unregister_driver(&cs423x_isa_driver); diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 6a85fdc53b60..38835f31298b 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -88,10 +88,6 @@ #include <sound/wss.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); -MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); -MODULE_LICENSE("GPL"); - /* * */ @@ -286,7 +282,8 @@ int snd_cs4236_create(struct snd_card *card, if (hardware == WSS_HW_DETECT) hardware = WSS_HW_DETECT3; if (cport < 0x100) { - snd_printk("please, specify control port for CS4236+ chips\n"); + snd_printk(KERN_ERR "please, specify control port " + "for CS4236+ chips\n"); return -ENODEV; } err = snd_wss_create(card, port, cport, @@ -295,7 +292,8 @@ int snd_cs4236_create(struct snd_card *card, return err; if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { - snd_printk("CS4236+: MODE3 and extended registers not available, hardware=0x%x\n",chip->hardware); + snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers " + "not available, hardware=0x%x\n", chip->hardware); snd_device_free(card, chip); return -ENODEV; } @@ -303,16 +301,19 @@ int snd_cs4236_create(struct snd_card *card, { int idx; for (idx = 0; idx < 8; idx++) - snd_printk("CD%i = 0x%x\n", idx, inb(chip->cport + idx)); + snd_printk(KERN_DEBUG "CD%i = 0x%x\n", + idx, inb(chip->cport + idx)); for (idx = 0; idx < 9; idx++) - snd_printk("C%i = 0x%x\n", idx, snd_cs4236_ctrl_in(chip, idx)); + snd_printk(KERN_DEBUG "C%i = 0x%x\n", + idx, snd_cs4236_ctrl_in(chip, idx)); } #endif ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2); if (ver1 != ver2) { - snd_printk("CS4236+ chip detected, but control port 0x%lx is not valid\n", cport); + snd_printk(KERN_ERR "CS4236+ chip detected, but " + "control port 0x%lx is not valid\n", cport); snd_device_free(card, chip); return -ENODEV; } @@ -883,7 +884,8 @@ static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct sn spin_lock_irqsave(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0; #if 0 - printk("get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + printk(KERN_DEBUG "get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " + "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), @@ -920,7 +922,8 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn mutex_unlock(&chip->mce_mutex); #if 0 - printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + printk(KERN_DEBUG "set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " + "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), @@ -1015,23 +1018,3 @@ int snd_cs4236_mixer(struct snd_wss *chip) } return 0; } - -EXPORT_SYMBOL(snd_cs4236_create); -EXPORT_SYMBOL(snd_cs4236_pcm); -EXPORT_SYMBOL(snd_cs4236_mixer); - -/* - * INIT part - */ - -static int __init alsa_cs4236_init(void) -{ - return 0; -} - -static void __exit alsa_cs4236_exit(void) -{ -} - -module_init(alsa_cs4236_init) -module_exit(alsa_cs4236_exit) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index d746750410ea..442b081cafb7 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -49,6 +49,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ @@ -65,6 +66,8 @@ MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); module_param_array(mpu_port, long, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); module_param_array(irq, int, NULL, 0444); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver."); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); @@ -143,13 +146,19 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name, chip->port, chip->irq, chip->dma8); - if (snd_opl3_create(card, chip->port, chip->port + 2, - OPL3_HW_OPL3, 0, &opl3) < 0) - dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); - else { - error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); - if (error < 0) - goto out; + if (fm_port[n] == SNDRV_AUTO_PORT) + fm_port[n] = port[n]; /* share the same port */ + + if (fm_port[n] > 0) { + if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) + dev_warn(dev, + "opl3 not detected at 0x%lx\n", fm_port[n]); + else { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } } if (mpu_irq[n] >= 0 && mpu_irq[n] != SNDRV_AUTO_IRQ && diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 4fbb508a817f..4c6e14f87f2d 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -45,7 +45,7 @@ static int snd_es1688_dsp_command(struct snd_es1688 *chip, unsigned char val) return 1; } #ifdef CONFIG_SND_DEBUG - printk("snd_es1688_dsp_command: timeout (0x%x)\n", val); + printk(KERN_DEBUG "snd_es1688_dsp_command: timeout (0x%x)\n", val); #endif return 0; } @@ -167,13 +167,16 @@ static int snd_es1688_probe(struct snd_es1688 *chip) hw = ES1688_HW_AUTO; switch (chip->version & 0xfff0) { case 0x4880: - snd_printk("[0x%lx] ESS: AudioDrive ES488 detected, but driver is in another place\n", chip->port); + snd_printk(KERN_ERR "[0x%lx] ESS: AudioDrive ES488 detected, " + "but driver is in another place\n", chip->port); return -ENODEV; case 0x6880: hw = (chip->version & 0x0f) >= 8 ? ES1688_HW_1688 : ES1688_HW_688; break; default: - snd_printk("[0x%lx] ESS: unknown AudioDrive chip with version 0x%x (Jazz16 soundcard?)\n", chip->port, chip->version); + snd_printk(KERN_ERR "[0x%lx] ESS: unknown AudioDrive chip " + "with version 0x%x (Jazz16 soundcard?)\n", + chip->port, chip->version); return -ENODEV; } @@ -223,7 +226,7 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) } } #if 0 - snd_printk("mpu cfg = 0x%x\n", cfg); + snd_printk(KERN_DEBUG "mpu cfg = 0x%x\n", cfg); #endif spin_lock_irqsave(&chip->reg_lock, flags); snd_es1688_mixer_write(chip, 0x40, cfg); @@ -237,7 +240,9 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* enable only DMA counter interrupt */ irq_bits = irqs[chip->irq & 0x0f]; if (irq_bits < 0) { - snd_printk("[0x%lx] ESS: bad IRQ %d for ES1688 chip!!\n", chip->port, chip->irq); + snd_printk(KERN_ERR "[0x%lx] ESS: bad IRQ %d " + "for ES1688 chip!!\n", + chip->port, chip->irq); #if 0 irq_bits = 0; cfg = 0x10; @@ -250,7 +255,8 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* extended mode DMA enable */ dma = chip->dma8; if (dma > 3 || dma == 2) { - snd_printk("[0x%lx] ESS: bad DMA channel %d for ES1688 chip!!\n", chip->port, dma); + snd_printk(KERN_ERR "[0x%lx] ESS: bad DMA channel %d " + "for ES1688 chip!!\n", chip->port, dma); #if 0 dma_bits = 0; cfg = 0x00; /* disable all DMA */ @@ -341,8 +347,9 @@ static int snd_es1688_trigger(struct snd_es1688 *chip, int cmd, unsigned char va return -EINVAL; /* something is wrong */ } #if 0 - printk("trigger: val = 0x%x, value = 0x%x\n", val, value); - printk("trigger: pointer = 0x%x\n", snd_dma_pointer(chip->dma8, chip->dma_size)); + printk(KERN_DEBUG "trigger: val = 0x%x, value = 0x%x\n", val, value); + printk(KERN_DEBUG "trigger: pointer = 0x%x\n", + snd_dma_pointer(chip->dma8, chip->dma_size)); #endif snd_es1688_write(chip, 0xb8, (val & 0xf0) | value); spin_unlock(&chip->reg_lock); diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index f45f6116c77a..36c27c832360 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -45,7 +45,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, unsigned char dma_cmd; unsigned int address_high; - // snd_printk("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", addr, (long) buf, count); + snd_printdd("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", + addr, buf_addr, count); if (gus->gf1.dma1 > 3) { if (gus->gf1.enh_mode) { @@ -77,7 +78,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, snd_gf1_dma_ack(gus); snd_dma_program(gus->gf1.dma1, buf_addr, count, dma_cmd & SNDRV_GF1_DMA_READ ? DMA_MODE_READ : DMA_MODE_WRITE); #if 0 - snd_printk("address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", address << 1, count, dma_cmd); + snd_printk(KERN_DEBUG "address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", + address << 1, count, dma_cmd); #endif spin_lock_irqsave(&gus->reg_lock, flags); if (gus->gf1.enh_mode) { @@ -142,7 +144,9 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 - printk("program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", addr, (long) buffer, count, cmd); + snd_printd(KERN_DEBUG "program dma (IRQ) - " + "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, block->buf_addr, block->count, block->cmd); #endif } @@ -203,13 +207,16 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, } *block = *__block; block->next = NULL; -#if 0 - printk("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", block->addr, (long) block->buffer, block->count, block->cmd); -#endif -#if 0 - printk("gus->gf1.dma_data_pcm_last = 0x%lx\n", (long)gus->gf1.dma_data_pcm_last); - printk("gus->gf1.dma_data_pcm = 0x%lx\n", (long)gus->gf1.dma_data_pcm); -#endif + + snd_printdd("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, (long) block->buffer, block->count, + block->cmd); + + snd_printdd("gus->gf1.dma_data_pcm_last = 0x%lx\n", + (long)gus->gf1.dma_data_pcm_last); + snd_printdd("gus->gf1.dma_data_pcm = 0x%lx\n", + (long)gus->gf1.dma_data_pcm); + spin_lock_irqsave(&gus->dma_lock, flags); if (synth) { if (gus->gf1.dma_data_synth_last) { diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index 041894ddd014..2055aff71b50 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -41,7 +41,7 @@ __again: if (status == 0) return IRQ_RETVAL(handled); handled = 1; - // snd_printk("IRQ: status = 0x%x\n", status); + /* snd_printk(KERN_DEBUG "IRQ: status = 0x%x\n", status); */ if (status & 0x02) { STAT_ADD(gus->gf1.interrupt_stat_midi_in); if (gus->gf1.interrupt_handler_midi_in) @@ -65,7 +65,9 @@ __again: continue; /* multi request */ already |= _current_; /* mark request */ #if 0 - printk("voice = %i, voice_status = 0x%x, voice_verify = %i\n", voice, voice_status, inb(GUSP(gus, GF1PAGE))); + printk(KERN_DEBUG "voice = %i, voice_status = 0x%x, " + "voice_verify = %i\n", + voice, voice_status, inb(GUSP(gus, GF1PAGE))); #endif pvoice = &gus->gf1.voices[voice]; if (pvoice->use) { diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index 38510aeb21c6..edb11eefdfe3 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -82,7 +82,10 @@ static int snd_gf1_pcm_block_change(struct snd_pcm_substream *substream, count += offset & 31; offset &= ~31; - // snd_printk("block change - offset = 0x%x, count = 0x%x\n", offset, count); + /* + snd_printk(KERN_DEBUG "block change - offset = 0x%x, count = 0x%x\n", + offset, count); + */ memset(&block, 0, sizeof(block)); block.cmd = SNDRV_GF1_DMA_IRQ; if (snd_pcm_format_unsigned(runtime->format)) @@ -135,7 +138,11 @@ static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream) curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels; end = curr + (pcmp->block_size / runtime->channels); end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1; - // snd_printk("init: curr=0x%x, begin=0x%x, end=0x%x, ctrl=0x%x, ramp=0x%x, rate=0x%x\n", curr, begin, end, voice_ctrl, ramp_ctrl, rate); + /* + snd_printk(KERN_DEBUG "init: curr=0x%x, begin=0x%x, end=0x%x, " + "ctrl=0x%x, ramp=0x%x, rate=0x%x\n", + curr, begin, end, voice_ctrl, ramp_ctrl, rate); + */ pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8; vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right; spin_lock_irqsave(&gus->reg_lock, flags); @@ -205,9 +212,11 @@ static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus, ramp_ctrl = (snd_gf1_read8(gus, SNDRV_GF1_VB_VOLUME_CONTROL) & ~0xa4) | 0x03; #if 0 snd_gf1_select_voice(gus, pvoice->number); - printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + printk(KERN_DEBUG "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pcmp->pvoices[1]->number); - printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + printk(KERN_DEBUG "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pvoice->number); #endif pcmp->bpos++; @@ -299,7 +308,11 @@ static int snd_gf1_pcm_poke_block(struct snd_gus_card *gus, unsigned char *buf, unsigned int len; unsigned long flags; - // printk("poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", (int)buf, pos, count, gus->gf1.port); + /* + printk(KERN_DEBUG + "poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", + (int)buf, pos, count, gus->gf1.port); + */ while (count > 0) { len = count; if (len > 512) /* limit, to allow IRQ */ @@ -680,7 +693,8 @@ static int snd_gf1_pcm_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_gf1_pcm_playback_free; #if 0 - printk("playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); + printk(KERN_DEBUG "playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", + (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); #endif if ((err = snd_gf1_dma_init(gus)) < 0) return err; diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index f0af3f79b08b..21cc42e4c4be 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -129,8 +129,14 @@ static int snd_gf1_uart_input_open(struct snd_rawmidi_substream *substream) } spin_unlock_irqrestore(&gus->uart_cmd_lock, flags); #if 0 - snd_printk("read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); - snd_printk("[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x (page = 0x%x)\n", gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); + snd_printk(KERN_DEBUG + "read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", + gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); + snd_printk(KERN_DEBUG + "[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x " + "(page = 0x%x)\n", + gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), + inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); #endif return 0; } diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 50e429a120da..534a6eced2b8 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -170,7 +170,7 @@ static void snd_interwave_i2c_setlines(struct snd_i2c_bus *bus, int ctrl, int da unsigned long port = bus->private_value; #if 0 - printk("i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); + printk(KERN_DEBUG "i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); #endif outb((data << 1) | ctrl, port); udelay(10); @@ -183,7 +183,7 @@ static int snd_interwave_i2c_getclockline(struct snd_i2c_bus *bus) res = inb(port) & 1; #if 0 - printk("i2c_getclockline - 0x%lx -> %i\n", port, res); + printk(KERN_DEBUG "i2c_getclockline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -197,7 +197,7 @@ static int snd_interwave_i2c_getdataline(struct snd_i2c_bus *bus, int ack) udelay(10); res = (inb(port) & 2) >> 1; #if 0 - printk("i2c_getdataline - 0x%lx -> %i\n", port, res); + printk(KERN_DEBUG "i2c_getdataline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -342,7 +342,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s snd_gf1_poke(gus, local, d); snd_gf1_poke(gus, local + 1, d + 1); #if 0 - printk("d = 0x%x, local = 0x%x, local + 1 = 0x%x, idx << 22 = 0x%x\n", + printk(KERN_DEBUG "d = 0x%x, local = 0x%x, " + "local + 1 = 0x%x, idx << 22 = 0x%x\n", d, snd_gf1_peek(gus, local), snd_gf1_peek(gus, local + 1), @@ -356,7 +357,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s } } #if 0 - printk("sizes: %i %i %i %i\n", sizes[0], sizes[1], sizes[2], sizes[3]); + printk(KERN_DEBUG "sizes: %i %i %i %i\n", + sizes[0], sizes[1], sizes[2], sizes[3]); #endif } @@ -410,12 +412,12 @@ static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus) lmct = (psizes[3] << 24) | (psizes[2] << 16) | (psizes[1] << 8) | psizes[0]; #if 0 - printk("lmct = 0x%08x\n", lmct); + printk(KERN_DEBUG "lmct = 0x%08x\n", lmct); #endif for (i = 0; i < ARRAY_SIZE(lmc); i++) if (lmct == lmc[i]) { #if 0 - printk("found !!! %i\n", i); + printk(KERN_DEBUG "found !!! %i\n", i); #endif snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i); snd_interwave_bank_sizes(gus, psizes); diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 645491a53023..ef95279da7a3 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -179,12 +179,13 @@ static unsigned char __snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char unsigned char result; #if 0 outb(0x1d, port); /* password */ - printk("read [0x%lx] = 0x%x\n", port, inb(port)); + printk(KERN_DEBUG "read [0x%lx] = 0x%x\n", port, inb(port)); #endif outb(reg, chip->port); /* register */ result = inb(chip->port + 1); #if 0 - printk("read [0x%lx] = 0x%x [0x%x]\n", port, result, inb(port)); + printk(KERN_DEBUG "read [0x%lx] = 0x%x [0x%x]\n", + port, result, inb(port)); #endif return result; } @@ -233,7 +234,10 @@ static int __devinit snd_opl3sa2_detect(struct snd_card *card) snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); return -EBUSY; } - // snd_printk("REG 0A = 0x%x\n", snd_opl3sa2_read(chip, 0x0a)); + /* + snd_printk(KERN_DEBUG "REG 0A = 0x%x\n", + snd_opl3sa2_read(chip, 0x0a)); + */ chip->version = 0; tmp = snd_opl3sa2_read(chip, OPL3SA2_MISC); if (tmp == 0xff) { @@ -550,21 +554,27 @@ static int __devinit snd_opl3sa2_mixer(struct snd_card *card) #ifdef CONFIG_PM static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state) { - struct snd_opl3sa2 *chip = card->private_data; + if (card) { + struct snd_opl3sa2 *chip = card->private_data; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - chip->wss->suspend(chip->wss); - /* power down */ - snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + chip->wss->suspend(chip->wss); + /* power down */ + snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); + } return 0; } static int snd_opl3sa2_resume(struct snd_card *card) { - struct snd_opl3sa2 *chip = card->private_data; + struct snd_opl3sa2 *chip; int i; + if (!card) + return 0; + + chip = card->private_data; /* power up */ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0); @@ -613,7 +623,7 @@ static void snd_opl3sa2_free(struct snd_card *card) { struct snd_opl3sa2 *chip = card->private_data; if (chip->irq >= 0) - free_irq(chip->irq, (void *)chip); + free_irq(chip->irq, card); release_and_free_resource(chip->res_port); } @@ -628,7 +638,7 @@ static int snd_opl3sa2_card_new(int dev, struct snd_card **cardp) if (err < 0) return err; strcpy(card->driver, "OPL3SA2"); - strcpy(card->shortname, "Yamaha OPL3-SA2"); + strcpy(card->shortname, "Yamaha OPL3-SA"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index cd6e60a6a4ea..5cd555325b9d 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -252,7 +252,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", hardware); + snd_printk(KERN_ERR "chip %d not supported\n", hardware); return -ENODEV; } return 0; @@ -294,7 +294,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -336,7 +336,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -412,7 +412,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); return -EINVAL; } @@ -430,7 +430,8 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk("WSS port 0x%lx not valid\n", chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", + chip->wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -455,7 +456,7 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk("WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); goto __skip_resources; } @@ -470,13 +471,14 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk("WSS dma1 # %d not valid\n", chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", + chip->dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) if (chip->dma1 == chip->dma2) { - snd_printk("don't want to share dmas\n"); + snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } @@ -485,7 +487,8 @@ __skip_base: case 1: break; default: - snd_printk("WSS dma2 # %d not valid\n", chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", + chip->dma2); goto __skip_resources; } dma_bits |= 0x04; @@ -516,7 +519,8 @@ __skip_resources: mpu_port_bits = 0x00; break; default: - snd_printk("MPU-401 port 0x%lx not valid\n", + snd_printk(KERN_WARNING + "MPU-401 port 0x%lx not valid\n", chip->mpu_port); goto __skip_mpu; } @@ -535,7 +539,7 @@ __skip_resources: mpu_irq_bits = 0x01; break; default: - snd_printk("MPU-401 irq # %d not valid\n", + snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", chip->mpu_irq); goto __skip_mpu; } @@ -726,7 +730,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (chip->wss_base == SNDRV_AUTO_PORT) { chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); if (chip->wss_base < 0) { - snd_printk("unable to find a free WSS port\n"); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } @@ -815,14 +819,8 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) chip->fm_port, chip->fm_port + 4 - 1); } if (opl3) { -#ifdef CS4231 - const int t1dev = 1; -#else - const int t1dev = 0; -#endif - if ((error = snd_opl3_timer_new(opl3, t1dev, t1dev+1)) < 0) - return error; - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, &synth)) < 0) + error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); + if (error < 0) return error; } } @@ -900,7 +898,7 @@ static int __devinit snd_opti9xx_isa_probe(struct device *devptr, #if defined(CS4231) || defined(OPTi93X) if (dma2 == SNDRV_AUTO_DMA) { if ((dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4])) < 0) { - snd_printk("unable to find a free DMA2\n"); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); return -EBUSY; } } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 406a431af91e..475220bbcc96 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -182,7 +182,7 @@ static int snd_sbmixer_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_dt019x_input_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[5] = { + static const char *texts[5] = { "CD", "Mic", "Line", "Synth", "Master" }; @@ -269,12 +269,73 @@ static int snd_dt019x_input_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl } /* + * ALS4000 mono recording control switch + */ + +static int snd_als4k_mono_capture_route_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "L chan only", "R chan only", "L ch/2 + R ch/2" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item > 2) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_als4k_mono_capture_route_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sb *sb = snd_kcontrol_chip(kcontrol); + unsigned long flags; + unsigned char oval; + + spin_lock_irqsave(&sb->mixer_lock, flags); + oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL); + spin_unlock_irqrestore(&sb->mixer_lock, flags); + oval >>= 6; + if (oval > 2) + oval = 2; + + ucontrol->value.enumerated.item[0] = oval; + return 0; +} + +static int snd_als4k_mono_capture_route_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sb *sb = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int change; + unsigned char nval, oval; + + if (ucontrol->value.enumerated.item[0] > 2) + return -EINVAL; + spin_lock_irqsave(&sb->mixer_lock, flags); + oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL); + + nval = (oval & ~(3 << 6)) + | (ucontrol->value.enumerated.item[0] << 6); + change = nval != oval; + if (change) + snd_sbmixer_write(sb, SB_ALS4000_MONO_IO_CTRL, nval); + spin_unlock_irqrestore(&sb->mixer_lock, flags); + return change; +} + +/* * SBPRO input multiplexer */ static int snd_sb8mixer_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char *texts[3] = { "Mic", "CD", "Line" }; @@ -442,6 +503,12 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty .get = snd_dt019x_input_sw_get, .put = snd_dt019x_input_sw_put, }, + [SB_MIX_MONO_CAPTURE_ALS4K] = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_als4k_mono_capture_route_info, + .get = snd_als4k_mono_capture_route_get, + .put = snd_als4k_mono_capture_route_put, + }, }; struct snd_kcontrol *ctl; int err; @@ -636,6 +703,8 @@ static struct sbmix_elem snd_dt019x_ctl_capture_source = }; static struct sbmix_elem *snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ &snd_dt019x_ctl_master_play_vol, &snd_dt019x_ctl_pcm_play_vol, &snd_dt019x_ctl_synth_play_vol, @@ -666,18 +735,21 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -/* FIXME: SB_ALS4000_MONO_IO_CTRL needs output select ctrl! */ static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4000_ctl_master_mono_capture_route = - SB_SINGLE("Master Mono Capture Route", SB_ALS4000_MONO_IO_CTRL, 6, 0x03); -/* FIXME: mono playback switch also available on DT019X? */ +static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { + .name = "Master Mono Capture Route", + .type = SB_MIX_MONO_CAPTURE_ALS4K + }; static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_loopback = - SB_SINGLE("Analog Loopback", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); +static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); +static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + SB_SINGLE("Digital Loopback Switch", + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); /* FIXME: functionality of 3D controls might be swapped, I didn't find * a description of how to identify what is supposed to be what */ static struct sbmix_elem snd_als4000_3d_control_switch = @@ -694,6 +766,9 @@ static struct sbmix_elem snd_als4000_3d_control_delay = SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); +static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", + SB_ALS4000_FMDAC, 5, 0x01); #ifdef NOT_AVAILABLE static struct sbmix_elem snd_als4000_ctl_fmdac = SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); @@ -702,35 +777,37 @@ static struct sbmix_elem snd_als4000_ctl_qsound = #endif static struct sbmix_elem *snd_als4000_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_switch, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_dt019x_ctl_synth_play_switch, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_als4000_ctl_mic_20db_boost, - &snd_sb16_ctl_auto_mic_gain, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_als4000_ctl_master_mono_playback_switch, - &snd_als4000_ctl_master_mono_capture_route, - &snd_als4000_ctl_mono_playback_switch, - &snd_als4000_ctl_mixer_loopback, - &snd_als4000_3d_control_switch, - &snd_als4000_3d_control_ratio, - &snd_als4000_3d_control_freq, - &snd_als4000_3d_control_delay, - &snd_als4000_3d_control_poweroff_switch, + /* ALS4000a.PDF regs page */ + &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ + &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ + &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ + &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ + &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ + &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ + &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ + &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ + &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ + &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ + &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ + &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ + &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ + &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ + &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ + &snd_sb16_ctl_play_vol, /* MX41/42 15 */ + &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ + &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ + &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ + &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ + &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ + &snd_als4000_3d_control_switch, /* MX50 17 */ + &snd_als4000_3d_control_ratio, /* MX50 17 */ + &snd_als4000_3d_control_freq, /* MX50 17 */ + &snd_als4000_3d_control_delay, /* MX51 18 */ + &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ + &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ #ifdef NOT_AVAILABLE &snd_als4000_ctl_fmdac, &snd_als4000_ctl_qsound, @@ -905,13 +982,14 @@ static unsigned char dt019x_saved_regs[] = { }; static unsigned char als4000_saved_regs[] = { + /* please verify in dsheet whether regs to be added + are actually real H/W or just dummy */ SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1, SB_DSP4_OUTPUT_SW, SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1, SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1, SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1, - SB_DSP4_MIC_AGC, SB_DSP4_MIC_DEV, SB_DSP4_SPEAKER_DEV, SB_DSP4_IGAIN_DEV, SB_DSP4_IGAIN_DEV + 1, @@ -919,8 +997,10 @@ static unsigned char als4000_saved_regs[] = { SB_DT019X_OUTPUT_SW2, SB_ALS4000_MONO_IO_CTRL, SB_ALS4000_MIC_IN_GAIN, + SB_ALS4000_FMDAC, SB_ALS4000_3D_SND_FX, SB_ALS4000_3D_TIME_DELAY, + SB_ALS4000_CR3_CONFIGURATION, }; static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 7a1470376c6d..782010608ef4 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -576,10 +576,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n", 0x388, 0x388 + 2); } else { - err = snd_opl3_timer_new(opl3, 0, 1); - if (err < 0) - goto err_unmap2; - err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto err_unmap2; diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 95898b2b7b58..a34ae7b1f7d0 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -553,11 +553,11 @@ static int __devinit snd_wavefront_isa_match(struct device *pdev, return 0; #endif if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) { - snd_printk("specify CS4232 port\n"); + snd_printk(KERN_ERR "specify CS4232 port\n"); return 0; } if (ics2115_port[dev] == SNDRV_AUTO_PORT) { - snd_printk("specify ICS2115 port\n"); + snd_printk(KERN_ERR "specify ICS2115 port\n"); return 0; } return 1; diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 4c410820a994..beb312cca75b 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -633,7 +633,7 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) wbuf[1] = i >> 7; if (snd_wavefront_cmd (dev, WFC_IDENTIFY_SAMPLE_TYPE, rbuf, wbuf)) { - snd_printk("cannot identify sample " + snd_printk(KERN_WARNING "cannot identify sample " "type of slot %d\n", i); dev->sample_status[i] = WF_ST_EMPTY; continue; diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 3d6c5f2838af..ac27832b2c6f 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -219,7 +219,8 @@ void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printk(KERN_DEBUG "out: auto calibration time out " + "- reg = 0x%x, value = 0x%x\n", reg, value); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); wss_outb(chip, CS4231P(REG), value); @@ -235,7 +236,8 @@ unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); + snd_printk(KERN_DEBUG "in: auto calibration time out " + "- reg = 0x%x\n", reg); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); mb(); @@ -252,7 +254,7 @@ void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg, wss_outb(chip, CS4231P(REG), val); chip->eimage[CS4236_REG(reg)] = val; #if 0 - printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val); + printk(KERN_DEBUG "ext out : reg = 0x%x, val = 0x%x\n", reg, val); #endif } EXPORT_SYMBOL(snd_cs4236_ext_out); @@ -268,7 +270,8 @@ unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg) { unsigned char res; res = wss_inb(chip, CS4231P(REG)); - printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res); + printk(KERN_DEBUG "ext in : reg = 0x%x, val = 0x%x\n", + reg, res); return res; } #endif @@ -394,13 +397,16 @@ void snd_wss_mce_up(struct snd_wss *chip) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_up - auto calibration time out (0)\n"); + snd_printk(KERN_DEBUG + "mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= CS4231_MCE; timeout = wss_inb(chip, CS4231P(REGSEL)); if (timeout == 0x80) - snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_DEBUG "mce_up [0x%lx]: " + "serious init problem - codec still busy\n", + chip->port); if (!(timeout & CS4231_MCE)) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); @@ -419,7 +425,9 @@ void snd_wss_mce_down(struct snd_wss *chip) #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); + snd_printk(KERN_DEBUG "mce_down [0x%lx] - " + "auto calibration time out (0)\n", + (long)CS4231P(REGSEL)); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit &= ~CS4231_MCE; @@ -427,7 +435,9 @@ void snd_wss_mce_down(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_DEBUG "mce_down [0x%lx]: " + "serious init problem - codec still busy\n", + chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask)) return; @@ -565,7 +575,7 @@ static unsigned char snd_wss_get_format(struct snd_wss *chip, if (channels > 1) rformat |= CS4231_STEREO; #if 0 - snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); + snd_printk(KERN_DEBUG "get_format: 0x%x (mode=0x%x)\n", format, mode); #endif return rformat; } @@ -774,7 +784,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (1)\n"); + snd_printk(KERN_DEBUG "init: (1)\n"); #endif snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -789,7 +799,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (2)\n"); + snd_printk(KERN_DEBUG "init: (2)\n"); #endif snd_wss_mce_up(chip); @@ -800,7 +810,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (3) - afei = 0x%x\n", + snd_printk(KERN_DEBUG "init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); #endif @@ -817,7 +827,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (4)\n"); + snd_printk(KERN_DEBUG "init: (4)\n"); #endif snd_wss_mce_up(chip); @@ -829,7 +839,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (5)\n"); + snd_printk(KERN_DEBUG "init: (5)\n"); #endif } @@ -1278,7 +1288,8 @@ static int snd_wss_probe(struct snd_wss *chip) } else if (rev == 0x03) { chip->hardware = WSS_HW_CS4236B; } else { - snd_printk("unknown CS chip with version 0x%x\n", rev); + snd_printk(KERN_ERR + "unknown CS chip with version 0x%x\n", rev); return -ENODEV; /* unknown CS4231 chip? */ } } @@ -1342,7 +1353,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4235 chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */ switch (id >> 5) { @@ -1353,7 +1367,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->hardware = WSS_HW_CS4236B; break; default: - snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4236 chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x08) { /* CS4237B */ chip->hardware = WSS_HW_CS4237B; @@ -1364,7 +1381,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4237B chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x09) { /* CS4238B */ chip->hardware = WSS_HW_CS4238B; @@ -1374,7 +1394,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4238B chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ chip->hardware = WSS_HW_CS4239; @@ -1384,10 +1407,15 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4239 chip " + "(enhanced version = 0x%x)\n", + id); } } else { - snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4236/CS423xB chip " + "(enhanced version = 0x%x)\n", id); } } } @@ -1618,7 +1646,8 @@ static void snd_wss_resume(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_ERR "down [0x%lx]: serious init problem " + "- codec still busy\n", chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { return; @@ -1628,7 +1657,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -static int snd_wss_free(struct snd_wss *chip) +int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1651,6 +1680,7 @@ static int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } +EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { @@ -1820,7 +1850,8 @@ int snd_wss_create(struct snd_card *card, #if 0 if (chip->hardware & WSS_HW_CS4232_MASK) { if (chip->res_cport == NULL) - snd_printk("CS4232 control port features are not accessible\n"); + snd_printk(KERN_ERR "CS4232 control port features are " + "not accessible\n"); } #endif diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 99e1391b2eb4..3e763d6a5d67 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -679,7 +679,7 @@ au1000_init(void) return err; } - printk( KERN_INFO "ALSA AC97: Driver Initialized\n" ); + printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); au1000_card = card; return 0; } diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 4d45bd63718b..38931f2f6967 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -847,22 +847,23 @@ static int __init AtaIrqInit(void) of events. So all we need to keep the music playing is to provide the sound hardware with new data upon an interrupt from timer A. */ - mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ - mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ - mfp.tim_ct_a = 8; /* Turn on event counting. */ + st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ + st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ + st_mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ - request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", - AtaInterrupt); - mfp.int_en_a |= 0x20; /* Turn interrupt on. */ - mfp.int_mk_a |= 0x20; + if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", + AtaInterrupt)) + return 0; + st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ + st_mfp.int_mk_a |= 0x20; return 1; } #ifdef MODULE static void AtaIrqCleanUp(void) { - mfp.tim_ct_a = 0; /* stop timer */ - mfp.int_en_a &= ~0x20; /* turn interrupt off */ + st_mfp.tim_ct_a = 0; /* stop timer */ + st_mfp.int_en_a &= ~0x20; /* turn interrupt off */ free_irq(IRQ_MFP_TIMA, AtaInterrupt); } #endif /* MODULE */ @@ -1598,7 +1599,7 @@ static int __init dmasound_atari_init(void) is_falcon = 0; } else return -ENODEV; - if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0) + if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0) return dmasound_init(); else { printk("DMA sound driver: Timer A interrupt already in use\n"); diff --git a/sound/oss/dmasound/dmasound_q40.c b/sound/oss/dmasound/dmasound_q40.c index 1855b14d90c3..99bcb21c2281 100644 --- a/sound/oss/dmasound/dmasound_q40.c +++ b/sound/oss/dmasound/dmasound_q40.c @@ -371,8 +371,9 @@ static void Q40Free(void *ptr, unsigned int size) static int __init Q40IrqInit(void) { /* Register interrupt handler. */ - request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, - "DMA sound", Q40Interrupt); + if (request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "DMA sound", Q40Interrupt)) + return 0; return(1); } @@ -401,6 +402,7 @@ static void Q40PlayNextFrame(int index) u_char *start; u_long size; u_char speed; + int error; /* used by Q40Play() if all doubts whether there really is something * to be played are already wiped out. @@ -419,11 +421,13 @@ static void Q40PlayNextFrame(int index) master_outb( 0,SAMPLE_ENABLE_REG); free_irq(Q40_IRQ_SAMPLE, Q40Interrupt); if (dmasound.soft.stereo) - request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, - "Q40 sound", Q40Interrupt); + error = request_irq(Q40_IRQ_SAMPLE, Q40StereoInterrupt, 0, + "Q40 sound", Q40Interrupt); else - request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0, - "Q40 sound", Q40Interrupt); + error = request_irq(Q40_IRQ_SAMPLE, Q40MonoInterrupt, 0, + "Q40 sound", Q40Interrupt); + if (error && printk_ratelimit()) + pr_err("Couldn't register sound interrupt\n"); master_outb( speed, SAMPLE_RATE_REG); master_outb( 1,SAMPLE_CLEAR_REG); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 6e3a1848447c..9387ab00a41b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -400,6 +400,26 @@ config SND_INDIGODJ To compile this driver as a module, choose M here: the module will be called snd-indigodj +config SND_INDIGOIOX + tristate "(Echoaudio) Indigo IOx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo IOx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigoiox + +config SND_INDIGODJX + tristate "(Echoaudio) Indigo DJx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo DJx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigodjx + config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" select FW_LOADER @@ -744,8 +764,8 @@ config SND_VIRTUOSO select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and - HDAV1.3 (Deluxe). + Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X. + Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index e2b843b4f9d0..97ee127ac33d 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -143,6 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x43525970, 0xfffffff8, "CS4202", NULL, NULL }, { 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II { 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different +{ 0x43585430, 0xffffffff, "Cx20468-31", patch_conexant, NULL }, { 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, { 0x44543031, 0xfffffff0, "DT0398", NULL, NULL }, { 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028? @@ -383,7 +384,7 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho EXPORT_SYMBOL(snd_ac97_update_bits); -/* no lock version - see snd_ac97_updat_bits() */ +/* no lock version - see snd_ac97_update_bits() */ int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value) { @@ -1643,7 +1644,10 @@ static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97) { int err, idx; - //printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG)); + /* + printk(KERN_DEBUG "AC97_GPIO_CFG = %x\n", + snd_ac97_read(ac97,AC97_GPIO_CFG)); + */ snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff); diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 060ea59d5f02..73b17d526c8b 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -125,6 +125,8 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe snd_iprintf(buffer, "PCI Subsys Device: 0x%04x\n\n", ac97->subsystem_device); + snd_iprintf(buffer, "Flags: %x\n", ac97->flags); + if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) { val = snd_ac97_read(ac97, AC97_INT_PAGING); snd_ac97_update_bits(ac97, AC97_INT_PAGING, diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 0f819ddb3ebf..fd135e3d8a84 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -51,7 +51,8 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531) int idx; for (idx = 0; idx < 0x19; idx++) - printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]); + printk(KERN_DEBUG "ak4531 0x%x: 0x%x\n", + idx, ak4531->regs[idx]); } #endif diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index b36c551da566..4edf270a7809 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2142,7 +2142,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec) { int err; - snd_ali_printk("resouces allocation ...\n"); + snd_ali_printk("resources allocation ...\n"); err = pci_request_regions(codec->pci, "ALI 5451"); if (err < 0) return err; @@ -2154,7 +2154,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec) return -EBUSY; } codec->irq = codec->pci->irq; - snd_ali_printk("resouces allocated.\n"); + snd_ali_printk("resources allocated.\n"); return 0; } static int snd_ali_dev_free(struct snd_device *device) diff --git a/sound/pci/als300.c b/sound/pci/als300.c index f557c155db48..009b4c8225a5 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -91,7 +91,7 @@ #define DEBUG_PLAY_REC 0 #if DEBUG_CALLS -#define snd_als300_dbgcalls(format, args...) printk(format, ##args) +#define snd_als300_dbgcalls(format, args...) printk(KERN_DEBUG format, ##args) #define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) #define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 649849e540d3..f4aa8ff6f5f9 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -462,9 +462,10 @@ static void a3dsrc_ZeroSliceIO(a3dsrc_t * a) /* Reset Single A3D source. */ static void a3dsrc_ZeroState(a3dsrc_t * a) { - - //printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source); - + /* + printk(KERN_DEBUG "vortex: ZeroState slice: %d, source %d\n", + a->slice, a->source); + */ a3dsrc_SetAtmosState(a, 0, 0, 0, 0); a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros); a3dsrc_SetItdDline(a, A3dItdDlineZeros); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index b070e5714514..3906f5afe27a 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1135,7 +1135,10 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, snd_pcm_sgbuf_get_addr(dma->substream, 0)); break; } - //printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1); + /* + printk(KERN_DEBUG "vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", + dma->cfg0, dma->cfg1); + */ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0); hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1); @@ -1959,7 +1962,7 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[]) ADB_CODECOUT(0 + 4)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[3], ADB_CODECOUT(1 + 4)); - //printk("SDAC detected "); + /* printk(KERN_DEBUG "SDAC detected "); */ } #else // Use plain direct output to codec. @@ -2013,7 +2016,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) resmap[restype] |= (1 << i); else vortex->dma_adb[i].resources[restype] |= (1 << i); - //printk("vortex: ResManager: type %d out %d\n", restype, i); + /* + printk(KERN_DEBUG + "vortex: ResManager: type %d out %d\n", + restype, i); + */ return i; } } @@ -2024,7 +2031,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) for (i = 0; i < qty; i++) { if (resmap[restype] & (1 << i)) { resmap[restype] &= ~(1 << i); - //printk("vortex: ResManager: type %d in %d\n",restype, i); + /* + printk(KERN_DEBUG + "vortex: ResManager: type %d in %d\n", + restype, i); + */ return i; } } @@ -2789,7 +2800,7 @@ vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod) { int a, this_194; - if ((bits != 8) || (bits != 16)) + if ((bits != 8) && (bits != 16)) return -1; switch (encod) { diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 978b856f5621..2805e34bd41d 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -213,38 +213,59 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, switch (reg) { /* Voice specific parameters */ case 0: /* running */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_RUN(wt), (int)val); + */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; break; case 1: /* param 0 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,0), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; break; case 2: /* param 1 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,1), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; break; case 3: /* param 2 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,2), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; break; case 4: /* param 3 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,3), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; break; case 6: /* mute */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_MUTE(wt), (int)val); + */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; break; case 0xb: { /* delay */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); @@ -272,7 +293,9 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, return 0; break; } - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + */ hwwrite(vortex->mmio, ecx, val); return 1; } diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index eefcbf648ee1..8eea29fc42fe 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); static struct pci_device_id snd_aw2_ids[] = { - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} }; diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 1df96e76c483..e9e9b5821d41 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -211,25 +211,25 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #endif #if DEBUG_MIXER -#define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) +#define snd_azf3328_dbgmixer(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) #endif #if DEBUG_PLAY_REC -#define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgplay(format, args...) #endif #if DEBUG_MISC -#define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgtimer(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) #endif #if DEBUG_GAME -#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbggame(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbggame(format, args...) #endif diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index b116456e7707..df757575798a 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -255,6 +255,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .gpio_type = 2, .i2c_adc = 1, .spi_dac = 1 } , + /* Giga-byte GA-G1975X mobo + * Novell bnc#395807 + */ + /* FIXME: the GPIO and I2C setting aren't tested well */ + { .serial = 0x1458a006, + .name = "Giga-byte GA-G1975X", + .gpio_type = 1, + .i2c_adc = 1 }, /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". @@ -404,7 +412,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; - // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); + /* + snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value); + */ /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -422,7 +432,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, /* Wait till the transaction ends */ while (1) { status = snd_ca0106_ptr_read(emu, I2C_A, 0); - //snd_printk("I2C:status=0x%x\n", status); + /*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/ timeout++; if ((status & I2C_A_ADC_START) == 0) break; @@ -521,7 +531,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -614,7 +627,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -705,9 +721,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ @@ -799,9 +826,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) u32 reg71_set = 0; u32 reg71; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* reg71 controls ADC rate. */ switch (runtime->rate) { case 44100: @@ -846,7 +884,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) } - //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + /* + printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ snd_ca0106_ptr_write(emu, 0x13, channel, 0); snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes @@ -888,13 +933,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; - /* snd_printk("channel=%d\n",channel); */ + /* snd_printk(KERN_DEBUG "channel=%d\n", channel); */ epcm->running = running; basic |= (0x1 << channel); extended |= (0x10 << channel); snd_pcm_trigger_done(s, substream); } - /* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */ + /* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -972,8 +1017,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -995,8 +1045,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -1181,8 +1236,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) return IRQ_NONE; stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); - //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); - //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + /* + snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n", + status, stat76); + snd_printk(KERN_DEBUG "ptr=0x%08x\n", + snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + */ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ for(i = 0; i < 4; i++) { pchannel = &(chip->playback_channels[i]); @@ -1470,7 +1529,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) int size, n; size = ARRAY_SIZE(i2c_adc_init); - /* snd_printk("I2C:array size=0x%x\n", size); */ + /* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */ for (n = 0; n < size; n++) snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index b9b07f464631..f6286f84a221 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -834,7 +834,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream) struct cs4281_dma *dma = runtime->private_data; struct cs4281 *chip = snd_pcm_substream_chip(substream); - // printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies); + /* + printk(KERN_DEBUG "DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", + snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, + jiffies); + */ return runtime->buffer_size - snd_cs4281_peekBA0(chip, dma->regDCC) - 1; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 8ab07aa63652..1be96ead4244 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -194,7 +194,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, * ACSDA = Status Data Register = 474h */ #if 0 - printk("e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg, + printk(KERN_DEBUG "e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg, snd_cs46xx_peekBA0(chip, BA0_ACSDA), snd_cs46xx_peekBA0(chip, BA0_ACCAD)); #endif @@ -428,8 +428,8 @@ static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout) } if(status & SERBST_WBSY) { - snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n"); - + snd_printk(KERN_ERR "cs46xx: failure waiting for " + "FIFO command to complete\n"); return -EINVAL; } diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index 018a7de56017..4eb55aa33612 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -62,7 +62,11 @@ static inline void snd_cs46xx_poke(struct snd_cs46xx *chip, unsigned long reg, u unsigned int bank = reg >> 16; unsigned int offset = reg & 0xffff; - /*if (bank == 0) printk("snd_cs46xx_poke: %04X - %08X\n",reg >> 2,val); */ + /* + if (bank == 0) + printk(KERN_DEBUG "snd_cs46xx_poke: %04X - %08X\n", + reg >> 2,val); + */ writel(val, chip->region.idx[bank+1].remap_addr + offset); } diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index ac1d72e0a1e4..c89ed1f5bc2b 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -312,7 +312,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card, if (request_irq(pci->irq, snd_cs5535audio_interrupt, IRQF_SHARED, "CS5535 Audio", cs5535au)) { - snd_printk("unable to grab IRQ %d\n", pci->irq); + snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); err = -EBUSY; goto sndfail; } diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile index 7b576aeb3f8d..1361de77e0cd 100644 --- a/sound/pci/echoaudio/Makefile +++ b/sound/pci/echoaudio/Makefile @@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o snd-indigo-objs := indigo.o snd-indigoio-objs := indigoio.o snd-indigodj-objs := indigodj.o +snd-indigoiox-objs := indigoiox.o +snd-indigodjx-objs := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o @@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o obj-$(CONFIG_SND_INDIGO) += snd-indigo.o obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o +obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o +obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 417e25add82b..57967e580571 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -56,7 +56,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) } chip->comm_page->e3g_frq_register = - __constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); + cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 9d015a76eb69..da2065cd2c0d 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip) Control interface ******************************************************************************/ +#ifndef ECHOCARD_HAS_VMIXER + /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol, return changed; } -#ifdef ECHOCARD_HAS_VMIXER -/* On Vmixer cards this one controls the line-out volume */ -static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { - .name = "Line Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = snd_echo_output_gain_info, - .get = snd_echo_output_gain_get, - .put = snd_echo_output_gain_put, - .tlv = {.p = db_scale_output_gain}, -}; -#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; + #endif @@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, #ifdef ECHOCARD_HAS_VMIXER snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) - goto ctl_error; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; #else diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 1c88e051abf2..f9490ae36c2e 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index c3736bbd819e..e32a74897921 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -40,8 +40,7 @@ static int check_asic_status(struct echoaudio *chip) if (wait_handshake(chip)) return -EIO; - chip->comm_page->ext_box_status = - __constant_cpu_to_le32(E3G_ASIC_NOT_LOADED); + chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED); chip->asic_loaded = FALSE; clear_handshake(chip); send_vector(chip, DSP_VC_TEST_ASIC); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index be0e18192de3..4df51ef5e095 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -926,11 +926,11 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init the comm page */ chip->comm_page->comm_size = - __constant_cpu_to_le32(sizeof(struct comm_page)); + cpu_to_le32(sizeof(struct comm_page)); chip->comm_page->handshake = 0xffffffff; chip->comm_page->midi_out_free_count = - __constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); - chip->comm_page->sample_rate = __constant_cpu_to_le32(44100); + cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); + chip->comm_page->sample_rate = cpu_to_le32(44100); chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index e352f3ae292c..cb7d75a0a503 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command. #define E3G_ASIC_NOT_LOADED 0xffff #define E3G_BOX_TYPE_MASK 0xf0 -#define EXT_3GBOX_NC 0x01 -#define EXT_3GBOX_NOT_SET 0x02 +/* Indigo express control register values */ +#define INDIGO_EXPRESS_32000 0x02 +#define INDIGO_EXPRESS_44100 0x01 +#define INDIGO_EXPRESS_48000 0x00 +#define INDIGO_EXPRESS_DOUBLE_SPEED 0x10 +#define INDIGO_EXPRESS_QUAD_SPEED 0x04 +#define INDIGO_EXPRESS_CLOCK_MASK 0x17 /* diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index db6c952e9d7f..3f1e7475faea 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -208,10 +208,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index f05e39f7aad9..0b2cd9c86277 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c new file mode 100644 index 000000000000..9ab625e15652 --- /dev/null +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -0,0 +1,119 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini@shiny.it> + +*************************************************************************/ + +static int set_sample_rate(struct echoaudio *chip, u32 rate) +{ + u32 clock, control_reg, old_control_reg; + + if (wait_handshake(chip)) + return -EIO; + + old_control_reg = le32_to_cpu(chip->comm_page->control_register); + control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK; + + switch (rate) { + case 32000: + clock = INDIGO_EXPRESS_32000; + break; + case 44100: + clock = INDIGO_EXPRESS_44100; + break; + case 48000: + clock = INDIGO_EXPRESS_48000; + break; + case 64000: + clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 88200: + clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 96000: + clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + default: + return -EINVAL; + } + + control_reg |= clock; + if (control_reg != old_control_reg) { + chip->comm_page->control_register = cpu_to_le32(control_reg); + chip->sample_rate = rate; + clear_handshake(chip); + return send_vector(chip, DSP_VC_UPDATE_CLOCKS); + } + return 0; +} + + + +/* This function routes the sound from a virtual channel to a real output */ +static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, + int gain) +{ + int index; + + if (snd_BUG_ON(pipe >= num_pipes_out(chip) || + output >= num_busses_out(chip))) + return -EINVAL; + + if (wait_handshake(chip)) + return -EIO; + + chip->vmixer_gain[output][pipe] = gain; + index = output * num_pipes_out(chip) + pipe; + chip->comm_page->vmixer[index] = gain; + + DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + return 0; +} + + + +/* Tell the DSP to read and update virtual mixer levels in comm page. */ +static int update_vmixer_level(struct echoaudio *chip) +{ + if (wait_handshake(chip)) + return -EIO; + clear_handshake(chip); + return send_vector(chip, DSP_VC_SET_VMIXER_GAIN); +} + + + +static u32 detect_input_clocks(const struct echoaudio *chip) +{ + return ECHO_CLOCK_BIT_INTERNAL; +} + + + +/* The IndigoIO has no ASIC. Just do nothing */ +static int load_asic(struct echoaudio *chip) +{ + return 0; +} diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 90730a5ecb42..08392916691e 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: vchannels 0-3 and - vchannels 4-7 are routed to real channels 0-4 */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 2, 2, 0); - set_vmixer_gain(chip, 3, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c new file mode 100644 index 000000000000..3482ef69f491 --- /dev/null +++ b/sound/pci/echoaudio/indigodjx.c @@ -0,0 +1,107 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it> + * + * 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; version 2 of the License. + * + * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_DJX +#define ECHOCARD_NAME "Indigo DJx" +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 0 */ +#define PX_DIGITAL_IN 8 /* 0 */ +#define PX_NUM 8 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 4 */ +#define BX_DIGITAL_OUT 4 /* 0 */ +#define BX_ANALOG_IN 4 /* 0 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_DJX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigodjx_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c new file mode 100644 index 000000000000..f591fc2ed960 --- /dev/null +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini@shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo DJx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index a7e09ec21079..0604c8a85223 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c new file mode 100644 index 000000000000..aebee27a40ff --- /dev/null +++ b/sound/pci/echoaudio/indigoiox.c @@ -0,0 +1,109 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it> + * + * 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; version 2 of the License. + * + * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_IOX +#define ECHOCARD_NAME "Indigo IOx" +#define ECHOCARD_HAS_MONITOR +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 2 */ +#define PX_DIGITAL_IN 10 /* 0 */ +#define PX_NUM 10 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 2 */ +#define BX_DIGITAL_OUT 2 /* 0 */ +#define BX_ANALOG_IN 2 /* 2 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_IOX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigoiox_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" + diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c new file mode 100644 index 000000000000..f357521c79e6 --- /dev/null +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library 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 +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini@shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo IOx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index ede75c6ca0fb..83750e9fd7b4 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -284,10 +284,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 227386602f9b..551405114cbc 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip))) return err; - /* Default routing of the virtual channels: vchannels 0-3 go to analog - outputs and vchannels 4-7 go to S/PDIF outputs */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 2, 4, 0); - set_vmixer_gain(chip, 3, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } @@ -222,10 +210,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 77bf2a83d997..a953d142cb4b 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -44,10 +44,10 @@ static int enable_midi_input(struct echoaudio *chip, char enable) if (enable) { chip->mtc_state = MIDI_IN_STATE_NORMAL; chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT); + cpu_to_le32(DSP_FLAG_MIDI_INPUT); } else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT); + ~cpu_to_le32(DSP_FLAG_MIDI_INPUT); clear_handshake(chip); return send_vector(chip, DSP_VC_UPDATE_FLAGS); diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 0e649dcdbf64..7ef949d99a50 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -103,7 +103,10 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw) int ch; vp = &emu->voices[best[i].voice]; if ((ch = vp->ch) < 0) { - //printk("synth_get_voice: ch < 0 (%d) ??", i); + /* + printk(KERN_WARNING + "synth_get_voice: ch < 0 (%d) ??", i); + */ continue; } vp->emu->num_voices--; @@ -335,7 +338,7 @@ start_voice(struct snd_emux_voice *vp) return -EINVAL; emem->map_locked++; if (snd_emu10k1_memblk_map(hw, emem) < 0) { - // printk("emu: cannot map!\n"); + /* printk(KERN_ERR "emu: cannot map!\n"); */ return -ENOMEM; } mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7958006a1d66..f18bd6207c50 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -711,8 +711,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filena static int emu1010_firmware_thread(void *data) { struct snd_emu10k1 *emu = data; - int tmp, tmp2; - int reg; + u32 tmp, tmp2, reg; int err; for (;;) { @@ -758,7 +757,8 @@ static int emu1010_firmware_thread(void *data) snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); - snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "Audio Dock ver: %u.%u\n", + tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); @@ -804,8 +804,7 @@ static int emu1010_firmware_thread(void *data) static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) { unsigned int i; - int tmp, tmp2; - int reg; + u32 tmp, tmp2, reg; int err; const char *filename = NULL; @@ -887,7 +886,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); - snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "emu1010: Hana version: %u.%u\n", tmp, tmp2); /* Enable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); @@ -1528,6 +1527,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 7dba08f0ab8e..191e1cd9997d 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1519,7 +1519,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */ if (emu->card_capabilities->emu_model) { /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */ - snd_printk("EMU outputs on\n"); + snd_printk(KERN_INFO "EMU outputs on\n"); for (z = 0; z < 8; z++) { if (emu->card_capabilities->ca0108_chip) { A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); @@ -1567,7 +1567,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->emu_model) { if (emu->card_capabilities->ca0108_chip) { - snd_printk("EMU2 inputs on\n"); + snd_printk(KERN_INFO "EMU2 inputs on\n"); for (z = 0; z < 0x10; z++) { snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, @@ -1575,10 +1575,13 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_FXBUS2(z*2) ); } } else { - snd_printk("EMU inputs on\n"); + snd_printk(KERN_INFO "EMU inputs on\n"); /* Capture 16 (originally 8) channels of S32_LE sound */ - /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ + /* + printk(KERN_DEBUG "emufx.c: gpr=0x%x, tmp=0x%x\n", + gpr, tmp); + */ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ /* A_P16VIN(0) is delayed by one sample, * so all other A_P16VIN channels will need to also be delayed diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cf9276ddad42..78f62fd404c2 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -44,7 +44,7 @@ static void snd_emu10k1_pcm_interrupt(struct snd_emu10k1 *emu, if (epcm->substream == NULL) return; #if 0 - printk("IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n", + printk(KERN_DEBUG "IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n", epcm->substream->runtime->hw->pointer(emu, epcm->substream), snd_pcm_lib_period_bytes(epcm->substream), snd_pcm_lib_buffer_bytes(epcm->substream)); @@ -146,7 +146,11 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic 1, &epcm->extra); if (err < 0) { - /* printk("pcm_channel_alloc: failed extra: voices=%d, frame=%d\n", voices, frame); */ + /* + printk(KERN_DEBUG "pcm_channel_alloc: " + "failed extra: voices=%d, frame=%d\n", + voices, frame); + */ for (i = 0; i < voices; i++) { snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]); epcm->voices[i] = NULL; @@ -737,7 +741,10 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, struct snd_emu10k1_pcm_mixer *mix; int result = 0; - /* printk("trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", (int)emu, cmd, substream->ops->pointer(substream)); */ + /* + printk(KERN_DEBUG "trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", + (int)emu, cmd, substream->ops->pointer(substream)) + */ spin_lock(&emu->reg_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -786,7 +793,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream, /* hmm this should cause full and half full interrupt to be raised? */ outl(epcm->capture_ipr, emu->port + IPR); snd_emu10k1_intr_enable(emu, epcm->capture_inte); - /* printk("adccr = 0x%x, adcbs = 0x%x\n", epcm->adccr, epcm->adcbs); */ + /* + printk(KERN_DEBUG "adccr = 0x%x, adcbs = 0x%x\n", + epcm->adccr, epcm->adcbs); + */ switch (epcm->type) { case CAPTURE_AC97ADC: snd_emu10k1_ptr_write(emu, ADCCR, 0, epcm->capture_cr_val); @@ -857,7 +867,11 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream * ptr -= runtime->buffer_size; } #endif - /* printk("ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", ptr, runtime->buffer_size, runtime->period_size); */ + /* + printk(KERN_DEBUG + "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", + ptr, runtime->buffer_size, runtime->period_size); + */ return ptr; } @@ -1546,7 +1560,11 @@ static void snd_emu10k1_fx8010_playback_tram_poke1(unsigned short *dst_left, unsigned int count, unsigned int tram_shift) { - /* printk("tram_poke1: dst_left = 0x%p, dst_right = 0x%p, src = 0x%p, count = 0x%x\n", dst_left, dst_right, src, count); */ + /* + printk(KERN_DEBUG "tram_poke1: dst_left = 0x%p, dst_right = 0x%p, " + "src = 0x%p, count = 0x%x\n", + dst_left, dst_right, src, count); + */ if ((tram_shift & 1) == 0) { while (count--) { *dst_left-- = *src++; @@ -1623,7 +1641,12 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number]; unsigned int i; - /* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */ + /* + printk(KERN_DEBUG "prepare: etram_pages = 0x%p, dma_area = 0x%x, " + "buffer_size = 0x%x (0x%x)\n", + emu->fx8010.etram_pages, runtime->dma_area, + runtime->buffer_size, runtime->buffer_size << 2); + */ memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec)); pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */ pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index b5a802bdeb7c..4bfc31d1b281 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -226,7 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, break; if (timeout > 1000) { - snd_printk("emu10k1:I2C:timeout status=0x%x\n", status); + snd_printk(KERN_WARNING + "emu10k1:I2C:timeout status=0x%x\n", + status); break; } } diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 749a21b6bd06..e617acaf10e3 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -168,7 +168,7 @@ static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime) struct snd_emu10k1_pcm *epcm = runtime->private_data; if (epcm) { - //snd_printk("epcm free: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm free: %p\n", epcm); */ kfree(epcm); } } @@ -183,14 +183,16 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* + snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n", + substream->pcm->device, channel_id); + */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; @@ -200,10 +202,15 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; - channel->epcm=epcm; +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "p16v: open channel_id=%d, channel=%p, use=0x%x\n", + channel_id, channel, channel->use); + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); +#endif /* debug */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -224,14 +231,16 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* + snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n", + substream->pcm->device, channel_id); + */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; @@ -241,10 +250,15 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; - channel->epcm=epcm; +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "p16v: open channel_id=%d, channel=%p, use=0x%x\n", + channel_id, channel, channel->use); + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); +#endif /* debug */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -334,9 +348,19 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) int i; u32 tmp; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG "prepare:channel_number=%d, rate=%d, " + "format=0x%x, channels=%d, buffer_size=%ld, " + "period_size=%ld, periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + runtime->periods, frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->p16v_buffer.addr, emu->p16v_buffer.area, + emu->p16v_buffer.bytes); +#endif /* debug */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -379,7 +403,15 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int channel = substream->pcm->device - emu->p16v_device_offset; u32 tmp; - //printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + + /* + printk(KERN_DEBUG "prepare capture:channel_number=%d, rate=%d, " + "format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -459,13 +491,13 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; - //snd_printk("p16v channel=%d\n",channel); + /* snd_printk(KERN_DEBUG "p16v channel=%d\n", channel); */ epcm->running = running; basic |= (0x1<<channel); inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel); snd_pcm_trigger_done(s, substream); } - //snd_printk("basic=0x%x, inte=0x%x\n",basic, inte); + /* snd_printk(KERN_DEBUG "basic=0x%x, inte=0x%x\n", basic, inte); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -558,8 +590,13 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr -= runtime->buffer_size; printk(KERN_WARNING "buffer capture limited!\n"); } - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -592,7 +629,10 @@ int snd_p16v_free(struct snd_emu10k1 *chip) // release the data if (chip->p16v_buffer.area) { snd_dma_free_pages(&chip->p16v_buffer); - //snd_printk("period lables free: %p\n", &chip->p16v_buffer); + /* + snd_printk(KERN_DEBUG "period lables free: %p\n", + &chip->p16v_buffer); + */ } return 0; } @@ -604,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * int err; int capture=1; - //snd_printk("snd_p16v_pcm called. device=%d\n", device); + /* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */ emu->p16v_device_offset = device; if (rpcm) *rpcm = NULL; @@ -631,7 +671,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * snd_dma_pci_data(emu->pci), ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) return err; - //snd_printk("preallocate playback substream: err=%d\n", err); + /* + snd_printk(KERN_DEBUG + "preallocate playback substream: err=%d\n", err); + */ } for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; @@ -642,7 +685,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * snd_dma_pci_data(emu->pci), 65536 - 64, 65536 - 64)) < 0) return err; - //snd_printk("preallocate capture substream: err=%d\n", err); + /* + snd_printk(KERN_DEBUG + "preallocate capture substream: err=%d\n", err); + */ } if (rpcm) diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index d7300a1aa262..20b8da250bd0 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -53,7 +53,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, *rvoice = NULL; first_voice = last_voice = 0; for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) { - // printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice); + /* + printk(KERN_DEBUG "i %d j %d next free %d!\n", + i, j, emu->next_free_voice); + */ i %= NUM_G; /* stereo voices must be even/odd */ @@ -71,7 +74,7 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, } } if (!skip) { - // printk("allocated voice %d\n", i); + /* printk(KERN_DEBUG "allocated voice %d\n", i); */ first_voice = i; last_voice = (i + number) % NUM_G; emu->next_free_voice = last_voice; @@ -84,7 +87,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, for (i = 0; i < number; i++) { voice = &emu->voices[(first_voice + i) % NUM_G]; - // printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number); + /* + printk(kERN_DEBUG "voice alloc - %i, %i of %i\n", + voice->number, idx-first_voice+1, number); + */ voice->use = 1; switch (type) { case EMU10K1_PCM: diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index e00614cbceff..18f4d1e98c46 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -584,7 +584,8 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531, unsigned long end_time = jiffies + HZ / 10; #if 0 - printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", + printk(KERN_DEBUG + "CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC)); #endif do { diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 34a78afc26d0..dd63b132fb8e 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1673,18 +1673,22 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) status = inb(SLIO_REG(chip, IRQCONTROL)); #if 0 - printk("Es1938debug - interrupt status: =0x%x\n", status); + printk(KERN_DEBUG "Es1938debug - interrupt status: =0x%x\n", status); #endif /* AUDIO 1 */ if (status & 0x10) { #if 0 - printk("Es1938debug - AUDIO channel 1 interrupt\n"); - printk("Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 interrupt\n"); + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n", inw(SLDM_REG(chip, DMACOUNT))); - printk("Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n", inl(SLDM_REG(chip, DMAADDR))); - printk("Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n", inl(SLDM_REG(chip, DMASTATUS))); #endif /* clear irq */ @@ -1699,10 +1703,13 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) /* AUDIO 2 */ if (status & 0x20) { #if 0 - printk("Es1938debug - AUDIO channel 2 interrupt\n"); - printk("Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 interrupt\n"); + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n", inw(SLIO_REG(chip, AUDIO2DMACOUNT))); - printk("Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n", inl(SLIO_REG(chip, AUDIO2DMAADDR))); #endif diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e00421c0d8ba..4de5bacd3929 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -135,10 +135,10 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) struct hda_beep *beep = codec->beep; if (beep) { cancel_work_sync(&beep->beep_work); - flush_scheduled_work(); input_unregister_device(beep->dev); kfree(beep); + codec->beep = NULL; } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index b9679f081cae..51bf6a5daf39 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -39,7 +39,7 @@ struct hda_beep { int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) +#define snd_hda_attach_beep_device(...) 0 #define snd_hda_detach_beep_device(...) #endif #endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e16cf63821ae..a4e5e5952115 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -373,7 +373,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsol->queue[wp] = res; unsol->queue[wp + 1] = res_ex; - schedule_work(&unsol->work); + queue_work(bus->workq, &unsol->work); return 0; } @@ -437,15 +437,17 @@ static int snd_hda_bus_free(struct hda_bus *bus) if (!bus) return 0; - if (bus->unsol) { - flush_scheduled_work(); + if (bus->workq) + flush_workqueue(bus->workq); + if (bus->unsol) kfree(bus->unsol); - } list_for_each_entry_safe(codec, n, &bus->codec_list, list) { snd_hda_codec_free(codec); } if (bus->ops.private_free) bus->ops.private_free(bus); + if (bus->workq) + destroy_workqueue(bus->workq); kfree(bus); return 0; } @@ -514,6 +516,16 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); + snprintf(bus->workq_name, sizeof(bus->workq_name), + "hd-audio%d", card->number); + bus->workq = create_singlethread_workqueue(bus->workq_name); + if (!bus->workq) { + snd_printk(KERN_ERR "cannot create workqueue %s\n", + bus->workq_name); + kfree(bus); + return -ENOMEM; + } + err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops); if (err < 0) { snd_hda_bus_free(bus); @@ -635,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; @@ -670,11 +682,140 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) return 0; } +/* read all pin default configurations and save codec->init_pins */ +static int read_pin_defaults(struct hda_codec *codec) +{ + int i; + hda_nid_t nid = codec->start_nid; + + 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; + if (wid_type != AC_WID_PIN) + continue; + pin = snd_array_new(&codec->init_pins); + if (!pin) + return -ENOMEM; + pin->nid = nid; + pin->cfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + } + return 0; +} + +/* look up the given pin config list and return the item matching with NID */ +static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, + struct snd_array *array, + hda_nid_t nid) +{ + int i; + for (i = 0; i < array->used; i++) { + struct hda_pincfg *pin = snd_array_elem(array, i); + if (pin->nid == nid) + return pin; + } + return NULL; +} + +/* write a config value for the given NID */ +static void set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg) +{ + int i; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + cfg & 0xff); + cfg >>= 8; + } +} + +/* set the current pin config value for the given NID. + * the value is cached, and read via snd_hda_codec_get_pincfg() + */ +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg) +{ + struct hda_pincfg *pin; + unsigned int oldcfg; + + oldcfg = snd_hda_codec_get_pincfg(codec, nid); + pin = look_up_pincfg(codec, list, nid); + if (!pin) { + pin = snd_array_new(list); + if (!pin) + return -ENOMEM; + pin->nid = nid; + } + pin->cfg = cfg; + + /* change only when needed; e.g. if the pincfg is already present + * in user_pins[], don't write it + */ + cfg = snd_hda_codec_get_pincfg(codec, nid); + if (oldcfg != cfg) + set_pincfg(codec, nid, cfg); + return 0; +} + +int snd_hda_codec_set_pincfg(struct hda_codec *codec, + hda_nid_t nid, unsigned int cfg) +{ + return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg); +} +EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); + +/* get the current pin config value of the given pin NID */ +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_pincfg *pin; + +#ifdef CONFIG_SND_HDA_HWDEP + pin = look_up_pincfg(codec, &codec->user_pins, nid); + if (pin) + return pin->cfg; +#endif + pin = look_up_pincfg(codec, &codec->driver_pins, nid); + if (pin) + return pin->cfg; + pin = look_up_pincfg(codec, &codec->init_pins, nid); + if (pin) + return pin->cfg; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg); + +/* restore all current pin configs */ +static void restore_pincfgs(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + set_pincfg(codec, pin->nid, + snd_hda_codec_get_pincfg(codec, pin->nid)); + } +} static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); +/* restore the initial pin cfgs and release all pincfg lists */ +static void restore_init_pincfgs(struct hda_codec *codec) +{ + /* first free driver_pins and user_pins, then call restore_pincfg + * so that only the values in init_pins are restored + */ + snd_array_free(&codec->driver_pins); +#ifdef CONFIG_SND_HDA_HWDEP + snd_array_free(&codec->user_pins); +#endif + restore_pincfgs(codec); + snd_array_free(&codec->init_pins); +} + /* * codec destructor */ @@ -682,9 +823,10 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + restore_init_pincfgs(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); - flush_scheduled_work(); + flush_workqueue(codec->bus->workq); #endif list_del(&codec->list); snd_array_free(&codec->mixers); @@ -700,6 +842,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state); + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -735,9 +880,12 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr codec->bus = bus; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -774,15 +922,18 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { snd_printdd("hda_codec: no AFG or MFG node found\n"); - snd_hda_codec_free(codec); - return -ENODEV; + err = -ENODEV; + goto error; } - if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) { + err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg); + if (err < 0) { snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); - snd_hda_codec_free(codec); - return -ENOMEM; + goto error; } + err = read_pin_defaults(codec); + if (err < 0) + goto error; if (!codec->subsystem_id) { hda_nid_t nid = codec->afg ? codec->afg : codec->mfg; @@ -793,12 +944,15 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (bus->modelname) codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); + /* power-up all before initialization */ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); + if (do_init) { err = snd_hda_codec_configure(codec); - if (err < 0) { - snd_hda_codec_free(codec); - return err; - } + if (err < 0) + goto error; } snd_hda_codec_proc_new(codec); @@ -811,6 +965,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (codecp) *codecp = codec; return 0; + + error: + snd_hda_codec_free(codec); + return err; } EXPORT_SYMBOL_HDA(snd_hda_codec_new); @@ -894,6 +1052,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* FIXME: more better hash key? */ #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define INFO_AMP_CAPS (1<<0) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) @@ -984,6 +1143,21 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_amp_info *info; + + info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + if (!info) + return 0; + if (!info->head.val) { + info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + info->head.val |= INFO_AMP_CAPS; + } + return info->amp_caps; +} +EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); + /* * read the current volume to info * if the cache exists, read the cache value. @@ -1107,6 +1281,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, u16 nid = get_amp_nid(kcontrol); u8 chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps; caps = query_amp_caps(codec, nid, dir); @@ -1118,6 +1293,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, kcontrol->id.name); return -EINVAL; } + if (ofs < caps) + caps -= ofs; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = chs == 3 ? 2 : 1; uinfo->value.integer.min = 0; @@ -1126,6 +1303,32 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); + +static inline unsigned int +read_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs) +{ + unsigned int val; + val = snd_hda_codec_amp_read(codec, nid, ch, dir, idx); + val &= HDA_AMP_VOLMASK; + if (val >= ofs) + val -= ofs; + else + val = 0; + return val; +} + +static inline int +update_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs, + unsigned int val) +{ + if (val > 0) + val += ofs; + return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, + HDA_AMP_VOLMASK, val); +} + int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1134,14 +1337,13 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; if (chs & 1) - *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) - & HDA_AMP_VOLMASK; + *valp++ = read_amp_value(codec, nid, 0, dir, idx, ofs); if (chs & 2) - *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) - & HDA_AMP_VOLMASK; + *valp = read_amp_value(codec, nid, 1, dir, idx, ofs); return 0; } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); @@ -1154,18 +1356,17 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; int change = 0; snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - 0x7f, *valp); + change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x7f, *valp); + change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); snd_hda_power_down(codec); return change; } @@ -1177,6 +1378,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps, val1, val2; if (size < 4 * sizeof(unsigned int)) @@ -1185,6 +1387,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; val2 = (val2 + 1) * 25; val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); + val1 += ofs; val1 = ((int)val1) * ((int)val2); if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; @@ -1255,7 +1458,6 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); -#ifdef CONFIG_SND_HDA_RECONFIG /* Clear all controls assigned to the given codec */ void snd_hda_ctls_clear(struct hda_codec *codec) { @@ -1266,20 +1468,62 @@ void snd_hda_ctls_clear(struct hda_codec *codec) snd_array_free(&codec->mixers); } -void snd_hda_codec_reset(struct hda_codec *codec) +/* pseudo device locking + * toggle card->shutdown to allow/disallow the device access (as a hack) + */ +static int hda_lock_devices(struct snd_card *card) { - int i; + spin_lock(&card->files_lock); + if (card->shutdown) { + spin_unlock(&card->files_lock); + return -EINVAL; + } + card->shutdown = 1; + spin_unlock(&card->files_lock); + return 0; +} + +static void hda_unlock_devices(struct snd_card *card) +{ + spin_lock(&card->files_lock); + card->shutdown = 0; + spin_unlock(&card->files_lock); +} + +int snd_hda_codec_reset(struct hda_codec *codec) +{ + struct snd_card *card = codec->bus->card; + int i, pcm; + + if (hda_lock_devices(card) < 0) + return -EBUSY; + /* check whether the codec isn't used by any mixer or PCM streams */ + if (!list_empty(&card->ctl_files)) { + hda_unlock_devices(card); + return -EBUSY; + } + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + if (!cpcm->pcm) + continue; + if (cpcm->pcm->streams[0].substream_opened || + cpcm->pcm->streams[1].substream_opened) { + hda_unlock_devices(card); + return -EBUSY; + } + } + + /* OK, let it free */ #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); - flush_scheduled_work(); + flush_workqueue(codec->bus->workq); #endif snd_hda_ctls_clear(codec); /* relase PCMs */ for (i = 0; i < codec->num_pcms; i++) { if (codec->pcm_info[i].pcm) { - snd_device_free(codec->bus->card, - codec->pcm_info[i].pcm); + snd_device_free(card, codec->pcm_info[i].pcm); clear_bit(codec->pcm_info[i].device, codec->bus->pcm_dev_bits); } @@ -1292,13 +1536,22 @@ void snd_hda_codec_reset(struct hda_codec *codec) free_hda_cache(&codec->cmd_cache); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); + restore_pincfgs(codec); codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; module_put(codec->owner); codec->owner = NULL; + + /* allow device access again */ + hda_unlock_devices(card); + return 0; } -#endif /* CONFIG_SND_HDA_RECONFIG */ /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, @@ -1323,15 +1576,20 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; - - sctl = snd_hda_find_mixer_ctl(codec, *s); - if (!sctl) { - snd_printdd("Cannot find slave %s, skipped\n", *s); - continue; + int i = 0; + for (;;) { + sctl = _snd_hda_find_mixer_ctl(codec, *s, i); + if (!sctl) { + if (!i) + snd_printdd("Cannot find slave %s, " + "skipped\n", *s); + break; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + i++; } - err = snd_ctl_add_slave(kctl, sctl); - if (err < 0) - return err; } return 0; } @@ -1418,12 +1676,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, unsigned long pval; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); kcontrol->private_value = pval; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get); @@ -1435,7 +1693,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, unsigned long pval; int i, indices, err = 0, change = 0; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); pval = kcontrol->private_value; indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT; for (i = 0; i < indices; i++) { @@ -1447,7 +1705,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, change |= err; } kcontrol->private_value = pval; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err < 0 ? err : change; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put); @@ -1462,12 +1720,12 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->info(kcontrol, uinfo); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info); @@ -1479,12 +1737,12 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->get(kcontrol, ucontrol); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get); @@ -1497,7 +1755,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, unsigned long *vals; int err = 0, change = 0; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; for (vals = c->values; *vals; vals++) { kcontrol->private_value = *vals; @@ -1507,7 +1765,7 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, change |= err; } kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err < 0 ? err : change; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put); @@ -1519,12 +1777,12 @@ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_bind_ctls *c; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); c = (struct hda_bind_ctls *)kcontrol->private_value; kcontrol->private_value = *c->values; err = c->ops->tlv(kcontrol, op_flag, size, tlv); kcontrol->private_value = (long)c; - mutex_unlock(&codec->spdif_mutex); + mutex_unlock(&codec->control_mutex); return err; } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_tlv); @@ -1942,6 +2200,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + if (!kctl) + return -ENOMEM; kctl->private_value = nid; err = snd_hda_ctl_add(codec, kctl); if (err < 0) @@ -2061,8 +2321,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * don't power down the widget if it controls * eapd and EAPD_BTLENABLE is set. */ - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_EAPD) { int eapd = snd_hda_codec_read(codec, nid, 0, @@ -2131,6 +2390,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + restore_pincfgs(codec); /* restore all current pin configs */ hda_exec_init_verbs(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); @@ -2158,8 +2418,16 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { int err = snd_hda_codec_build_controls(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build controls" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } return 0; } @@ -2168,19 +2436,12 @@ EXPORT_SYMBOL_HDA(snd_hda_build_controls); int snd_hda_codec_build_controls(struct hda_codec *codec) { int err = 0; - /* fake as if already powered-on */ - hda_keep_power_on(codec); - /* then fire up */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); hda_exec_init_verbs(codec); /* continue to initialize... */ if (codec->patch_ops.init) err = codec->patch_ops.init(codec); if (!err && codec->patch_ops.build_controls) err = codec->patch_ops.build_controls(codec); - snd_hda_power_down(codec); if (err < 0) return err; return 0; @@ -2293,12 +2554,11 @@ EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp) { - int i; - unsigned int val, streams; + unsigned int i, val, wcaps; val = 0; - if (nid != codec->afg && - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { + wcaps = get_wcaps(codec, nid); + if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (val == -1) return -EIO; @@ -2312,15 +2572,20 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (val & (1 << i)) rates |= rate_bits[i].alsa_bits; } + if (rates == 0) { + snd_printk(KERN_ERR "hda_codec: rates == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0); + return -EIO; + } *ratesp = rates; } if (formatsp || bpsp) { u64 formats = 0; - unsigned int bps; - unsigned int wcaps; + unsigned int streams, bps; - wcaps = get_wcaps(codec, nid); streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (streams == -1) return -EIO; @@ -2373,6 +2638,15 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, formats |= SNDRV_PCM_FMTBIT_U8; bps = 8; } + if (formats == 0) { + snd_printk(KERN_ERR "hda_codec: formats == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i, " + "streams=0x%x)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0, + streams); + return -EIO; + } if (formatsp) *formatsp = formats; if (bpsp) @@ -2488,12 +2762,16 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, static int set_pcm_default_values(struct hda_codec *codec, struct hda_pcm_stream *info) { + int err; + /* query support PCM information from the given NID */ if (info->nid && (!info->rates || !info->formats)) { - snd_hda_query_supported_pcm(codec, info->nid, + err = snd_hda_query_supported_pcm(codec, info->nid, info->rates ? NULL : &info->rates, info->formats ? NULL : &info->formats, info->maxbps ? NULL : &info->maxbps); + if (err < 0) + return err; } if (info->ops.open == NULL) info->ops.open = hda_pcm_default_open_close; @@ -2536,13 +2814,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { dev = audio_idx[i]; if (!test_bit(dev, bus->pcm_dev_bits)) - break; + goto ok; } - if (i >= ARRAY_SIZE(audio_idx)) { - snd_printk(KERN_WARNING "Too many audio devices\n"); - return -EAGAIN; - } - break; + snd_printk(KERN_WARNING "Too many audio devices\n"); + return -EAGAIN; case HDA_PCM_TYPE_SPDIF: case HDA_PCM_TYPE_HDMI: case HDA_PCM_TYPE_MODEM: @@ -2557,6 +2832,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); return -EINVAL; } + ok: set_bit(dev, bus->pcm_dev_bits); return dev; } @@ -2593,24 +2869,36 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (!codec->patch_ops.build_pcms) return 0; err = codec->patch_ops.build_pcms(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build PCMs" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } for (pcm = 0; pcm < codec->num_pcms; pcm++) { struct hda_pcm *cpcm = &codec->pcm_info[pcm]; int dev; if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) - return 0; /* no substreams assigned */ + continue; /* no substreams assigned */ if (!cpcm->pcm) { dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); if (dev < 0) - return 0; + continue; /* no fatal error */ cpcm->device = dev; err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot attach " + "PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ + } } } return 0; @@ -2712,6 +3000,67 @@ int snd_hda_check_board_config(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** + * snd_hda_check_board_codec_sid_config - compare the current codec + subsystem ID with the + config table + + This is important for Gateway notebooks with SB450 HDA Audio + where the vendor ID of the PCI device is: + ATI Technologies Inc SB450 HDA Audio [1002:437b] + and the vendor/subvendor are found only at the codec. + + * @codec: the HDA codec + * @num_configs: number of config enums + * @models: array of model name strings + * @tbl: configuration table, terminated by null entries + * + * Compares the modelname or PCI subsystem id of the current codec with the + * given configuration table. If a matching entry is found, returns its + * config value (supposed to be 0 or positive). + * + * If no entries are matching, the function returns a negative value. + */ +int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, + int num_configs, const char **models, + const struct snd_pci_quirk *tbl) +{ + const struct snd_pci_quirk *q; + + /* Search for codec ID */ + for (q = tbl; q->subvendor; q++) { + unsigned long vendorid = (q->subdevice) | (q->subvendor << 16); + + if (vendorid == codec->subsystem_id) + break; + } + + if (!q->subvendor) + return -1; + + tbl = q; + + if (tbl->value >= 0 && tbl->value < num_configs) { +#ifdef CONFIG_SND_DEBUG_DETECT + char tmp[10]; + const char *model = NULL; + if (models) + model = models[tbl->value]; + if (!model) { + sprintf(tmp, "#%d", tbl->value); + model = tmp; + } + snd_printdd(KERN_INFO "hda_codec: model '%s' is selected " + "for config %x:%x (%s)\n", + model, tbl->subvendor, tbl->subdevice, + (tbl->name ? tbl->name : "Unknown device")); +#endif + return tbl->value; + } + return -1; +} +EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config); + +/** * snd_hda_add_new_ctls - create controls from the array * @codec: the HDA codec * @knew: the array of struct snd_kcontrol_new @@ -2803,7 +3152,7 @@ void snd_hda_power_down(struct hda_codec *codec) return; if (power_save(codec)) { codec->power_transition = 1; /* avoid reentrance */ - schedule_delayed_work(&codec->power_work, + queue_delayed_work(codec->bus->workq, &codec->power_work, msecs_to_jiffies(power_save(codec) * 1000)); } } @@ -3014,6 +3363,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + mutex_lock(&codec->spdif_mutex); + cleanup_dig_out_stream(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup); + /* * release the digital out */ @@ -3240,8 +3599,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, if (ignore_nids && is_in_nid_list(nid, ignore_nids)) continue; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf); @@ -3317,10 +3675,22 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_AUX] = nid; break; case AC_JACK_SPDIF_OUT: - cfg->dig_out_pin = nid; + case AC_JACK_DIG_OTHER_OUT: + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + continue; + cfg->dig_out_pins[cfg->dig_outs] = nid; + cfg->dig_out_type[cfg->dig_outs] = + (loc == AC_JACK_LOC_HDMI) ? + HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; + cfg->dig_outs++; break; case AC_JACK_SPDIF_IN: + case AC_JACK_DIG_OTHER_IN: cfg->dig_in_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_in_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; break; } } @@ -3426,6 +3796,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); + if (cfg->dig_outs) + snd_printd(" dig-out=0x%x/0x%x\n", + cfg->dig_out_pins[0], cfg->dig_out_pins[1]); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], @@ -3434,6 +3807,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_FRONT_LINE], cfg->input_pins[AUTO_PIN_CD], cfg->input_pins[AUTO_PIN_AUX]); + if (cfg->dig_in_pin) + snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 729fc7642d7f..2fdecf4b0eb6 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,6 +614,8 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; + char workq_name[16]; + struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -737,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; @@ -771,15 +774,19 @@ struct hda_codec { struct hda_cache_rec cmd_cache; /* cache for other commands */ struct mutex spdif_mutex; + struct mutex control_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ + struct snd_array init_pins; /* initial (BIOS) pin configurations */ + struct snd_array driver_pins; /* pin configs set by codec parser */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ struct snd_array init_verbs; /* additional init verbs */ struct snd_array hints; /* additional hints */ + struct snd_array user_pins; /* default pin configs to override */ #endif /* misc flags */ @@ -787,6 +794,9 @@ struct hda_codec { * status change * (e.g. Realtek codecs) */ + unsigned int pin_amp_workaround:1; /* pin out-amp takes index + * (e.g. Conexant codecs) + */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ @@ -852,6 +862,18 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec); #define snd_hda_sequence_write_cache snd_hda_sequence_write #endif +/* the struct for codec->pin_configs */ +struct hda_pincfg { + hda_nid_t nid; + unsigned int cfg; +}; + +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg); +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg); /* for hwdep */ + /* * Mixer */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 65745e96dc70..1d5797a96682 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -144,9 +144,9 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; if (node->type == AC_WID_PIN) { - node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); + node->pin_caps = snd_hda_query_pin_caps(codec, node->nid); node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid); } if (node->wid_caps & AC_WCAP_OUT_AMP) { diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 300ab407cf42..1c57505c2874 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -30,6 +30,12 @@ #include <sound/hda_hwdep.h> #include <sound/minors.h> +/* hint string pair */ +struct hda_hint { + const char *key; + const char *val; /* contained in the same alloc as key */ +}; + /* * write/read an out-of-bound verb */ @@ -99,16 +105,17 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) static void clear_hwdep_elements(struct hda_codec *codec) { - char **head; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - head = codec->hints.list; - for (i = 0; i < codec->hints.used; i++, head++) - kfree(*head); + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + kfree(hint->key); /* we don't need to free hint->val */ + } snd_array_free(&codec->hints); + snd_array_free(&codec->user_pins); } static void hwdep_free(struct snd_hwdep *hwdep) @@ -140,7 +147,8 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) #endif snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); - snd_array_init(&codec->hints, sizeof(char *), 32); + snd_array_init(&codec->hints, sizeof(struct hda_hint), 32); + snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16); return 0; } @@ -153,7 +161,13 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) static int clear_codec(struct hda_codec *codec) { - snd_hda_codec_reset(codec); + int err; + + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR "The codec is being used, can't free.\n"); + return err; + } clear_hwdep_elements(codec); return 0; } @@ -162,20 +176,29 @@ static int reconfig_codec(struct hda_codec *codec) { int err; + snd_hda_power_up(codec); snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); - snd_hda_codec_reset(codec); + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR + "The codec is being used, can't reconfigure.\n"); + goto error; + } err = snd_hda_codec_configure(codec); if (err < 0) - return err; + goto error; /* rebuild PCMs */ err = snd_hda_codec_build_pcms(codec); if (err < 0) - return err; + goto error; /* rebuild mixers */ err = snd_hda_codec_build_controls(codec); if (err < 0) - return err; - return 0; + goto error; + err = snd_card_register(codec->bus->card); + error: + snd_hda_power_down(codec); + return err; } /* @@ -271,47 +294,195 @@ static ssize_t type##_store(struct device *dev, \ CODEC_ACTION_STORE(reconfig); CODEC_ACTION_STORE(clear); +static ssize_t init_verbs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->init_verbs.used; i++) { + struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "0x%02x 0x%03x 0x%04x\n", + v->nid, v->verb, v->param); + } + return len; +} + 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; - char *p; - struct hda_verb verb, *v; + struct hda_verb *v; + int nid, verb, param; - verb.nid = simple_strtoul(buf, &p, 0); - verb.verb = simple_strtoul(p, &p, 0); - verb.param = simple_strtoul(p, &p, 0); - if (!verb.nid || !verb.verb || !verb.param) + if (sscanf(buf, "%i %i %i", &nid, &verb, ¶m) != 3) + return -EINVAL; + if (!nid || !verb) return -EINVAL; v = snd_array_new(&codec->init_verbs); if (!v) return -ENOMEM; - *v = verb; + v->nid = nid; + v->verb = verb; + v->param = param; return count; } +static ssize_t hints_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "%s = %s\n", hint->key, hint->val); + } + return len; +} + +static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) +{ + int i; + + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + if (!strcmp(hint->key, key)) + return hint; + } + return NULL; +} + +static void remove_trail_spaces(char *str) +{ + char *p; + if (!*str) + return; + p = str + strlen(str) - 1; + for (; isspace(*p); p--) { + *p = 0; + if (p == str) + return; + } +} + +#define MAX_HINTS 1024 + 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; - char *p; - char **hint; + char *key, *val; + struct hda_hint *hint; - if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n') + while (isspace(*buf)) + buf++; + if (!*buf || *buf == '#' || *buf == '\n') return count; - p = kstrndup_noeol(buf, 1024); - if (!p) + if (*buf == '=') + return -EINVAL; + key = kstrndup_noeol(buf, 1024); + if (!key) return -ENOMEM; - hint = snd_array_new(&codec->hints); + /* extract key and val */ + val = strchr(key, '='); + if (!val) { + kfree(key); + return -EINVAL; + } + *val++ = 0; + while (isspace(*val)) + val++; + remove_trail_spaces(key); + remove_trail_spaces(val); + hint = get_hint(codec, key); + if (hint) { + /* replace */ + kfree(hint->key); + hint->key = key; + hint->val = val; + return count; + } + /* allocate a new hint entry */ + if (codec->hints.used >= MAX_HINTS) + hint = NULL; + else + hint = snd_array_new(&codec->hints); if (!hint) { - kfree(p); + kfree(key); return -ENOMEM; } - *hint = p; + hint->key = key; + hint->val = val; + return count; +} + +static ssize_t pin_configs_show(struct hda_codec *codec, + struct snd_array *list, + char *buf) +{ + int i, len = 0; + for (i = 0; i < list->used; i++) { + struct hda_pincfg *pin = snd_array_elem(list, i); + len += sprintf(buf + len, "0x%02x 0x%08x\n", + pin->nid, pin->cfg); + } + return len; +} + +static ssize_t init_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->init_pins, buf); +} + +static ssize_t user_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->user_pins, buf); +} + +static ssize_t driver_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->driver_pins, buf); +} + +#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) +{ + 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); + if (err < 0) + return err; return count; } @@ -330,8 +501,11 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RO(mfg), CODEC_ATTR_RW(name), CODEC_ATTR_RW(modelname), - CODEC_ATTR_WO(init_verbs), - CODEC_ATTR_WO(hints), + CODEC_ATTR_RW(init_verbs), + CODEC_ATTR_RW(hints), + CODEC_ATTR_RO(init_pin_configs), + CODEC_ATTR_RW(user_pin_configs), + CODEC_ATTR_RO(driver_pin_configs), CODEC_ATTR_WO(reconfig), CODEC_ATTR_WO(clear), }; @@ -350,4 +524,29 @@ int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) return 0; } +/* + * Look for hint string + */ +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + struct hda_hint *hint = get_hint(codec, key); + return hint ? hint->val : NULL; +} +EXPORT_SYMBOL_HDA(snd_hda_get_hint); + +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + const char *p = snd_hda_get_hint(codec, key); + if (!p || !*p) + return -ENOENT; + switch (toupper(*p)) { + case 'T': /* true */ + case 'Y': /* yes */ + case '1': + return 1; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint); + #endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ad5df2ae6f7d..30829ee920c3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -381,6 +381,7 @@ struct azx { /* HD codec */ unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ struct hda_bus *bus; /* CORB/RIRB */ @@ -858,13 +859,18 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) SD_CTL_DMA_START | SD_INT_MASK); } -/* stop a stream */ -static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +/* stop DMA */ +static void azx_stream_clear(struct azx *chip, struct azx_dev *azx_dev) { - /* stop DMA */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~(SD_CTL_DMA_START | SD_INT_MASK)); azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ +} + +/* stop a stream */ +static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +{ + azx_stream_clear(chip, azx_dev); /* disable SIE */ azx_writeb(chip, INTCTL, azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); @@ -996,10 +1002,11 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); - } else { + } else if (chip->bus && chip->bus->workq) { /* bogus IRQ, process it later */ azx_dev->irq_pending = 1; - schedule_work(&chip->irq_pending_work); + queue_work(chip->bus->workq, + &chip->irq_pending_work); } } } @@ -1074,8 +1081,7 @@ static int azx_setup_periods(struct azx *chip, azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); - period_bytes = snd_pcm_lib_period_bytes(substream); - azx_dev->period_bytes = period_bytes; + period_bytes = azx_dev->period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ @@ -1122,24 +1128,17 @@ static int azx_setup_periods(struct azx *chip, error: snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); return -EINVAL; } -/* - * set up the SD for streaming - */ -static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +/* reset stream */ +static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev) { unsigned char val; int timeout; - /* make sure the run bit is zero for SD */ - azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & - ~SD_CTL_DMA_START); - /* reset stream */ + azx_stream_clear(chip, azx_dev); + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET); udelay(3); @@ -1156,7 +1155,15 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) && --timeout) ; +} +/* + * set up the SD for streaming + */ +static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +{ + /* make sure the run bit is zero for SD */ + azx_stream_clear(chip, azx_dev); /* program the stream_tag */ azx_sd_writel(azx_dev, SD_CTL, (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)| @@ -1227,7 +1234,6 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { }; static int __devinit azx_codec_create(struct azx *chip, const char *model, - unsigned int codec_probe_mask, int no_init) { struct hda_bus_template bus_temp; @@ -1260,7 +1266,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { if (probe_codec(chip, c) < 0) { /* Some BIOSen give you wrong codec addresses * that don't exist @@ -1284,7 +1290,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* Then create codec instances */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + 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); if (err < 0) @@ -1402,6 +1408,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + + azx_stream_reset(chip, azx_dev); return 0; } @@ -1428,6 +1436,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -1442,6 +1455,9 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; hinfo->ops.cleanup(hinfo, apcm->codec, substream); @@ -1455,23 +1471,37 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int bufsize, period_bytes, format_val; + int err; - azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); - azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hinfo->maxbps); - if (!azx_dev->format_val) { + format_val = snd_hda_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hinfo->maxbps); + if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", runtime->rate, runtime->channels, runtime->format); return -EINVAL; } + bufsize = snd_pcm_lib_buffer_bytes(substream); + period_bytes = snd_pcm_lib_period_bytes(substream); + snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(chip, substream, azx_dev) < 0) - return -EINVAL; + bufsize, format_val); + + if (bufsize != azx_dev->bufsize || + period_bytes != azx_dev->period_bytes || + format_val != azx_dev->format_val) { + azx_dev->bufsize = bufsize; + azx_dev->period_bytes = period_bytes; + azx_dev->format_val = format_val; + err = azx_setup_periods(chip, substream, azx_dev); + if (err < 0) + return err; + } + azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; @@ -1741,7 +1771,6 @@ static void azx_clear_irq_pending(struct azx *chip) for (i = 0; i < chip->num_streams; i++) chip->azx_dev[i].irq_pending = 0; spin_unlock_irq(&chip->reg_lock); - flush_scheduled_work(); } static struct snd_pcm_ops azx_pcm_ops = { @@ -1947,16 +1976,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) return 0; } -static int azx_resume_early(struct pci_dev *pci) -{ - return pci_restore_state(pci); -} - static int azx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); if (pci_enable_device(pci) < 0) { printk(KERN_ERR "hda-intel: pci_enable_device failed, " "disabling device\n"); @@ -2062,26 +2088,31 @@ static int __devinit check_position_fix(struct azx *chip, int fix) { const struct snd_pci_quirk *q; - /* Check VIA HD Audio Controller exist */ - if (chip->pci->vendor == PCI_VENDOR_ID_VIA && - chip->pci->device == VIA_HDAC_DEVICE_ID) { + switch (fix) { + case POS_FIX_LPIB: + case POS_FIX_POSBUF: + return fix; + } + + /* Check VIA/ATI HD Audio Controller exist */ + switch (chip->driver_type) { + case AZX_DRIVER_VIA: + case AZX_DRIVER_ATI: chip->via_dmapos_patch = 1; /* Use link position directly, avoid any transfer problem. */ return POS_FIX_LPIB; } chip->via_dmapos_patch = 0; - if (fix == POS_FIX_AUTO) { - q = snd_pci_quirk_lookup(chip->pci, position_fix_list); - if (q) { - printk(KERN_INFO - "hda_intel: position_fix set to %d " - "for device %04x:%04x\n", - q->value, q->subvendor, q->subdevice); - return q->value; - } + q = snd_pci_quirk_lookup(chip->pci, position_fix_list); + if (q) { + printk(KERN_INFO + "hda_intel: position_fix set to %d " + "for device %04x:%04x\n", + q->value, q->subvendor, q->subdevice); + return q->value; } - return fix; + return POS_FIX_AUTO; } /* @@ -2098,23 +2129,36 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), + /* forced codec slots */ + SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} }; +#define AZX_FORCE_CODEC_MASK 0x100 + static void __devinit check_probe_mask(struct azx *chip, int dev) { const struct snd_pci_quirk *q; - if (probe_mask[dev] == -1) { + chip->codec_probe_mask = probe_mask[dev]; + if (chip->codec_probe_mask == -1) { q = snd_pci_quirk_lookup(chip->pci, probe_mask_list); if (q) { printk(KERN_INFO "hda_intel: probe_mask set to 0x%x " "for device %04x:%04x\n", q->value, q->subvendor, q->subdevice); - probe_mask[dev] = q->value; + chip->codec_probe_mask = q->value; } } + + /* check forced option */ + if (chip->codec_probe_mask != -1 && + (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) { + chip->codec_mask = chip->codec_probe_mask & 0xff; + printk(KERN_INFO "hda_intel: codec_mask forced to 0x%x\n", + chip->codec_mask); + } } @@ -2211,9 +2255,17 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd("chipset global capabilities = 0x%x\n", gcap); + /* ATI chips seems buggy about 64bit DMA addresses */ + if (chip->driver_type == AZX_DRIVER_ATI) + gcap &= ~0x01; + /* allow 64bit DMA address if supported by H/W */ if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK)) pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK); + else { + pci_set_dma_mask(pci, DMA_32BIT_MASK); + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK); + } /* read number of streams from GCAP register instead of using * hardcoded value @@ -2347,8 +2399,7 @@ static int __devinit azx_probe(struct pci_dev *pci, card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_mask[dev], - probe_only[dev]); + err = azx_codec_create(chip, model[dev], probe_only[dev]); if (err < 0) goto out_free; @@ -2445,10 +2496,10 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* AMD Generic, PCI class code and Vendor ID for HD Audio */ @@ -2468,7 +2519,6 @@ static struct pci_driver driver = { .remove = __devexit_p(azx_remove), #ifdef CONFIG_PM .suspend = azx_suspend, - .resume_early = azx_resume_early, .resume = azx_resume, #endif }; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 6f2fe0f9fdd8..83349013b4df 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -26,8 +26,10 @@ /* * for mixer controls */ +#define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs) \ + ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23)) #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \ - ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19)) + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0) /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ @@ -96,7 +98,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves); -void snd_hda_codec_reset(struct hda_codec *codec); +int snd_hda_codec_reset(struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ @@ -134,7 +136,7 @@ extern struct hda_ctl_ops snd_hda_bind_sw; /* for bind-switch */ struct hda_bind_ctls { struct hda_ctl_ops *ops; - long values[]; + unsigned long values[]; }; int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, @@ -227,6 +229,7 @@ struct hda_multi_out { hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ hda_nid_t dig_out_nid; /* digital out audio widget */ + hda_nid_t *slave_dig_outs; int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ @@ -251,6 +254,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream, @@ -296,6 +301,9 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen); int snd_hda_check_board_config(struct hda_codec *codec, int num_configs, const char **modelnames, const struct snd_pci_quirk *pci_list); +int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, + int num_configs, const char **models, + const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); @@ -349,9 +357,12 @@ struct auto_pin_cfg { int line_out_type; /* AUTO_PIN_XXX_OUT */ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; hda_nid_t input_pins[AUTO_PIN_LAST]; - hda_nid_t dig_out_pin; + int dig_outs; + hda_nid_t dig_out_pins[2]; hda_nid_t dig_in_pin; hda_nid_t mono_out_pin; + int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ + int dig_in_type; /* HDA_PCM_TYPE_XXX */ }; #define get_defcfg_connect(cfg) \ @@ -400,6 +411,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) 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); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); void snd_hda_ctls_clear(struct hda_codec *codec); @@ -422,6 +434,23 @@ static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) } #endif +#ifdef CONFIG_SND_HDA_RECONFIG +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key); +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key); +#else +static inline +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + return NULL; +} + +static inline +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + return -ENOENT; +} +#endif + /* * power-management */ @@ -453,6 +482,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) +#define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f) /* * CEA Short Audio Descriptor data diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7ca66d654148..93d7499350c6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -399,7 +399,10 @@ static void print_conn_list(struct snd_info_buffer *buffer, { int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX) + if (conn_len > 1 && + wid_type != AC_WID_AUD_MIX && + wid_type != AC_WID_VOL_KNB && + wid_type != AC_WID_POWER) curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); @@ -466,8 +469,9 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg) @@ -553,8 +557,14 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-Out caps: "); print_amp_caps(buffer, codec, nid, HDA_OUTPUT); snd_iprintf(buffer, " Amp-Out vals: "); - print_amp_vals(buffer, codec, nid, HDA_OUTPUT, - wid_caps & AC_WCAP_STEREO, 1); + if (wid_type == AC_WID_PIN && + codec->pin_amp_workaround) + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); + else + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, 1); } switch (wid_type) { diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 26247cfe749d..5bb48ee8b6c6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -27,11 +27,12 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; - + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL termination! */ @@ -154,6 +155,16 @@ static const char *ad_slave_sws[] = { static void ad198x_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new ad_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT), + { } /* end */ +}; + +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ + static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -181,6 +192,21 @@ static int ad198x_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = ad_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; @@ -275,6 +301,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, format, substream); } +static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture */ @@ -333,7 +367,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = { .ops = { .open = ad198x_dig_playback_pcm_open, .close = ad198x_dig_playback_pcm_close, - .prepare = ad198x_dig_playback_pcm_prepare + .prepare = ad198x_dig_playback_pcm_prepare, + .cleanup = ad198x_dig_playback_pcm_cleanup }, }; @@ -397,7 +432,8 @@ static void ad198x_free(struct hda_codec *codec) return; ad198x_free_kctls(codec); - kfree(codec->spec); + kfree(spec); + snd_hda_detach_beep_device(codec); } static struct hda_codec_ops ad198x_patch_ops = { @@ -536,8 +572,6 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -601,8 +635,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -800,8 +833,6 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), { @@ -993,10 +1024,8 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), @@ -1018,15 +1047,14 @@ static struct hda_amp_list ad1986a_loopbacks[] = { static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) { - unsigned int conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + unsigned int conf = snd_hda_codec_get_pincfg(codec, nid); return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; } static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1034,6 +1062,13 @@ static int patch_ad1986a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x19); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); spec->multiout.dac_nids = ad1986a_dac_nids; @@ -1213,8 +1248,6 @@ static struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), @@ -1285,6 +1318,7 @@ static struct hda_amp_list ad1983_loopbacks[] = { static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1292,6 +1326,13 @@ static int patch_ad1983(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids); spec->multiout.dac_nids = ad1983_dac_nids; @@ -1361,8 +1402,6 @@ static struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), @@ -1407,8 +1446,8 @@ static struct hda_verb ad1981_init_verbs[] = { {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, /* Mic boost: 0dB */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Record selector: Front mic */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, @@ -1673,10 +1712,10 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ - SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), + SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), /* Lenovo Thinkpad T60/X60/Z6xx */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD), /* HP nx6320 (reversed SSID, H/W bug) */ SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), {} @@ -1685,7 +1724,7 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1693,6 +1732,13 @@ static int patch_ad1981(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids); spec->multiout.dac_nids = ad1981_dac_nids; @@ -1885,8 +1931,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = { #define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 -static hda_nid_t ad1989b_slave_dig_outs[2] = { - AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +static hda_nid_t ad1989b_slave_dig_outs[] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 }; static struct hda_input_mux ad1988_6stack_capture_source = { @@ -1979,9 +2025,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2025,9 +2068,6 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2057,9 +2097,6 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2288,10 +2325,6 @@ static struct hda_verb ad1988_capture_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, { } }; @@ -2399,10 +2432,6 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } @@ -2474,10 +2503,6 @@ static struct hda_verb ad1988_laptop_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } @@ -2881,7 +2906,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = AD1988_SPDIF_IN; @@ -2931,7 +2956,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { static int patch_ad1988(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2951,7 +2976,7 @@ static int patch_ad1988(struct hda_codec *codec) if (board_config == AD1988_AUTO) { /* automatic parse from the BIOS config */ - int err = ad1988_parse_auto_config(codec); + err = ad1988_parse_auto_config(codec); if (err < 0) { ad198x_free(codec); return err; @@ -2961,6 +2986,13 @@ static int patch_ad1988(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: @@ -3117,12 +3149,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - */ HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3195,10 +3221,10 @@ static struct hda_verb ad1884_init_verbs[] = { {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear mic) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Analog mixer; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -3231,7 +3257,7 @@ static const char *ad1884_slave_vols[] = { "CD Playback Volume", "Internal Mic Playback Volume", "Docking Mic Playback Volume" - "Beep Playback Volume", + /* "Beep Playback Volume", */ "IEC958 Playback Volume", NULL }; @@ -3239,6 +3265,7 @@ static const char *ad1884_slave_vols[] = { static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3246,6 +3273,13 @@ static int patch_ad1884(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); spec->multiout.dac_nids = ad1884_dac_nids; @@ -3312,8 +3346,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), 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), @@ -3349,7 +3381,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* docking mic boost */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Analog mixer - docking mic; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ @@ -3370,10 +3402,6 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - */ HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3459,7 +3487,7 @@ static const char *ad1984_models[AD1984_MODELS] = { static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), {} }; @@ -3552,8 +3580,6 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3613,10 +3639,10 @@ static struct hda_verb ad1884a_init_verbs[] = { {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear line-in) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-E (rear mic) pin */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -3686,8 +3712,6 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3715,8 +3739,6 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3827,8 +3849,6 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3900,10 +3920,11 @@ static const char *ad1884a_models[AD1884A_MODELS] = { static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), + 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(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; @@ -3911,7 +3932,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { static int patch_ad1884a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3919,6 +3940,13 @@ static int patch_ad1884a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); spec->multiout.dac_nids = ad1884a_dac_nids; @@ -3956,6 +3984,14 @@ static int patch_ad1884a(struct hda_codec *codec) spec->multiout.dig_out_nid = 0; codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1884A_THINKPAD: spec->mixers[0] = ad1984a_thinkpad_mixers; @@ -4073,8 +4109,6 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), { } /* end */ }; @@ -4087,8 +4121,6 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT), { } /* end */ }; @@ -4247,7 +4279,7 @@ static const char *ad1882_models[AD1986A_MODELS] = { static int patch_ad1882(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4255,6 +4287,13 @@ static int patch_ad1882(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = 3; spec->multiout.dac_nids = ad1882_dac_nids; @@ -4262,13 +4301,13 @@ static int patch_ad1882(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); spec->adc_nids = ad1882_adc_nids; spec->capsrc_nids = ad1882_capsrc_nids; - if (codec->vendor_id == 0x11d1882) + if (codec->vendor_id == 0x11d41882) spec->input_mux = &ad1882_capture_source; else spec->input_mux = &ad1882a_capture_source; spec->num_mixers = 2; spec->mixers[0] = ad1882_base_mixers; - if (codec->vendor_id == 0x11d1882) + if (codec->vendor_id == 0x11d41882) spec->mixers[1] = ad1882_loopback_mixers; else spec->mixers[1] = ad1882a_loopback_mixers; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index f3ebe837f2d5..c921264bbd71 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -680,13 +680,13 @@ static int patch_cmi9880(struct hda_codec *codec) struct auto_pin_cfg cfg; /* collect pin default configuration */ - port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_e = snd_hda_codec_get_pincfg(codec, 0x0f); + port_f = snd_hda_codec_get_pincfg(codec, 0x10); spec->front_panel = 1; if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE || get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) { - port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_g = snd_hda_codec_get_pincfg(codec, 0x1f); + port_h = snd_hda_codec_get_pincfg(codec, 0x20); spec->channel_modes = cmi9880_channel_modes; /* no front panel */ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE || @@ -703,8 +703,8 @@ static int patch_cmi9880(struct hda_codec *codec) spec->multiout.max_channels = cmi9880_channel_modes[0].channels; } else { spec->input_mux = &cmi9880_basic_mux; - port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_spdifi = snd_hda_codec_get_pincfg(codec, 0x13); + port_spdifo = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE) spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75de40aaab0a..1f2ad76ca94b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -58,6 +58,7 @@ struct conexant_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; + hda_nid_t vmaster_nid; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -72,6 +73,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; + unsigned int no_auto_mic; unsigned int need_dac_fix; /* capture */ @@ -347,6 +349,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } +#ifdef CONFIG_SND_JACK static int conexant_add_jack(struct hda_codec *codec, hda_nid_t nid, int type) { @@ -394,7 +397,6 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) static int conexant_init_jacks(struct hda_codec *codec) { -#ifdef CONFIG_SND_JACK struct conexant_spec *spec = codec->spec; int i; @@ -422,10 +424,19 @@ static int conexant_init_jacks(struct hda_codec *codec) ++hv; } } -#endif return 0; } +#else +static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ +} + +static inline int conexant_init_jacks(struct hda_codec *codec) +{ + return 0; +} +#endif static int conexant_init(struct hda_codec *codec) { @@ -452,6 +463,29 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } +static struct snd_kcontrol_new cxt_capture_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + {} +}; + +static const char *slave_vols[] = { + "Headphone Playback Volume", + "Speaker Playback Volume", + NULL +}; + +static const char *slave_sws[] = { + "Headphone Playback Switch", + "Speaker Playback Switch", + NULL +}; + static int conexant_build_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -479,6 +513,32 @@ static int conexant_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; + snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, + HDA_OUTPUT, vmaster_tlv); + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + vmaster_tlv, slave_vols); + if (err < 0) + return err; + } + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_sws); + if (err < 0) + return err; + } + + if (spec->input_mux) { + err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); + if (err < 0) + return err; + } + return 0; } @@ -710,13 +770,6 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static struct snd_kcontrol_new cxt5045_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), @@ -750,13 +803,6 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { }; static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), @@ -993,15 +1039,9 @@ static const char *cxt5045_models[CXT5045_MODELS] = { }; static struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), - SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -1011,8 +1051,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), {} }; @@ -1026,6 +1066,7 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -1125,7 +1166,7 @@ static int patch_cxt5045(struct hda_codec *codec) /* Conexant 5047 specific */ #define CXT5047_SPDIF_OUT 0x11 -static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c }; +static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ static hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; @@ -1133,20 +1174,6 @@ static struct hda_channel_mode cxt5047_modes[1] = { { 2, NULL }, }; -static struct hda_input_mux cxt5047_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x2 }, - } -}; - -static struct hda_input_mux cxt5047_hp_capture_source = { - .num_items = 1, - .items = { - { "ExtMic", 0x2 }, - } -}; - static struct hda_input_mux cxt5047_toshiba_capture_source = { .num_items = 2, .items = { @@ -1170,7 +1197,11 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, * the headphone jack */ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, + /* NOTE: Conexat codec needs the index for *OUTPUT* amp of + * pin widgets unlike other codecs. In this case, we need to + * set index 0x01 for the volume from the mixer amp 0x19. + */ + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, @@ -1178,16 +1209,6 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */ -static struct hda_bind_ctls cxt5047_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), - 0 - }, -}; - /* mute internal speaker if HP is plugged */ static void cxt5047_hp_automute(struct hda_codec *codec) { @@ -1198,27 +1219,8 @@ static void cxt5047_hp_automute(struct hda_codec *codec) AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp2_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_codec_read(codec, 0x13, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - - bits = spec->hp_present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, + /* See the note in cxt5047_hp_master_sw_put */ + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); } @@ -1259,55 +1261,14 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec, } } -/* unsolicited event for HP jack sensing - non-EAPD systems */ -static void cxt5047_hp2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5047_hp2_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static struct snd_kcontrol_new cxt5047_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_base_mixers[] = { + HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT), - - {} -}; - -static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -1320,29 +1281,15 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5047_hp_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { + /* See the note in cxt5047_hp_master_sw_put */ + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, { } /* end */ }; @@ -1353,8 +1300,8 @@ static struct hda_verb cxt5047_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, /* HP, Speaker */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1d, AC_VERB_SET_CONNECT_SEL,0x0}, + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ /* Record selector: Mic */ {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1374,30 +1321,7 @@ static struct hda_verb cxt5047_init_verbs[] = { /* configuration for Toshiba Laptops */ static struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */ - /* pin sensing on HP and Mic jacks */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, - {} -}; - -/* configuration for HP Laptops */ -static struct hda_verb cxt5047_hp_init_verbs[] = { - /* pin sensing on HP jack */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - /* 0x13 is actually shared by both HP and speaker; - * setting the connection to 0 (=0x19) makes the master volume control - * working mysteriouslly... - */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Record selector: Ext Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ {} }; @@ -1562,10 +1486,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = { }; static struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5047_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; @@ -1579,6 +1502,7 @@ static int patch_cxt5047(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); @@ -1587,9 +1511,8 @@ static int patch_cxt5047(struct hda_codec *codec) spec->num_adc_nids = 1; spec->adc_nids = cxt5047_adc_nids; spec->capsrc_nids = cxt5047_capsrc_nids; - spec->input_mux = &cxt5047_capture_source; spec->num_mixers = 1; - spec->mixers[0] = cxt5047_mixers; + spec->mixers[0] = cxt5047_base_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5047_init_verbs; spec->spdif_route = 0; @@ -1603,21 +1526,22 @@ static int patch_cxt5047(struct hda_codec *codec) cxt5047_cfg_tbl); switch (board_config) { case CXT5047_LAPTOP: - codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; + codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->input_mux = &cxt5047_hp_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_hp_init_verbs; - spec->mixers[0] = cxt5047_hp_mixers; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_only_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; codec->patch_ops.init = cxt5047_hp_init; break; case CXT5047_LAPTOP_EAPD: spec->input_mux = &cxt5047_toshiba_capture_source; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - spec->mixers[0] = cxt5047_toshiba_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; #ifdef CONFIG_SND_DEBUG @@ -1628,6 +1552,7 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; #endif } + spec->vmaster_nid = 0x13; return 0; } @@ -1663,8 +1588,11 @@ static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5051_portb_automic(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1680,6 +1608,8 @@ static void cxt5051_portc_automic(struct hda_codec *codec) unsigned int present; hda_nid_t new_adc; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1766,6 +1696,22 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1796,6 +1742,66 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + +static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Docking HP */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1813,18 +1819,24 @@ static int cxt5051_init(struct hda_codec *codec) enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_HP_DV6736, /* HP without mic switch */ + CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_HP_DV6736] = "hp-dv6736", + [CXT5051_LENOVO_X200] = "lenovo-x200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), {} }; @@ -1837,6 +1849,7 @@ static int patch_cxt5051(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; codec->patch_ops = conexant_patch_ops; codec->patch_ops.init = cxt5051_init; @@ -1857,17 +1870,22 @@ static int patch_cxt5051(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); switch (board_config) { case CXT5051_HP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; - default: - case CXT5051_LAPTOP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + case CXT5051_HP_DV6736: + spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; + spec->mixers[0] = cxt5051_hp_dv6736_mixers; + spec->no_auto_mic = 1; + break; + case CXT5051_LENOVO_X200: + spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; } diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 3564f4e4b74c..fcc77fec4487 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = { {} /* terminator */ }; -static struct hda_verb pinout_disable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00}, - {} -}; - static struct hda_verb unsolicited_response_verb[] = { {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | INTEL_HDMI_EVENT_TAG}, @@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, static void hdmi_enable_output(struct hda_codec *codec) { - /* Enable Audio InfoFrame Transmission */ - 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); /* Unmute */ if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, PIN_NID, 0, @@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec) snd_hda_sequence_write(codec, pinout_enable_verb); } -static void hdmi_disable_output(struct hda_codec *codec) +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - snd_hda_sequence_write(codec, pinout_disable_verb); - 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_MUTE); + 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); +} - /* - * FIXME: noises may arise when playing music after reloading the - * kernel module, until the next X restart or monitor repower. - */ +/* + * Disable Audio InfoFrame Transmission + */ +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, + AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) @@ -368,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { u8 *params = (u8 *)ai; + u8 sum = 0; int i; hdmi_debug_dip_size(codec); hdmi_clear_dip_buffers(codec); /* be paranoid */ + for (i = 0; i < sizeof(ai); i++) + sum += params[i]; + ai->checksum = - sum; + 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]); @@ -419,14 +422,18 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, /* * CA defaults to 0 for basic stereo audio */ - if (!eld->eld_ver) - return 0; - if (!eld->spk_alloc) - return 0; if (channels <= 2) return 0; /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!eld->spk_alloc) + eld->spk_alloc = 0xffff; + + /* * expand ELD's speaker allocation mask * * ELD tells the speaker mask in a compact(paired) form, @@ -485,6 +492,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hdmi_setup_channel_mapping(codec, &ai); hdmi_fill_audio_infoframe(codec, &ai); + hdmi_start_infoframe_trans(codec); } @@ -562,7 +570,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, { struct intel_hdmi_spec *spec = codec->spec; - hdmi_disable_output(codec); + hdmi_stop_infoframe_trans(codec); return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -582,8 +590,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, substream); - hdmi_enable_output(codec); - return 0; } @@ -628,8 +634,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) static int intel_hdmi_init(struct hda_codec *codec) { - /* disable audio output as early as possible */ - hdmi_disable_output(codec); + hdmi_enable_output(codec); snd_hda_sequence_write(codec, unsolicited_response_verb); @@ -679,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .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 = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; @@ -687,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb"); 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:10951392"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 0270fda0bda5..d57d8132a06e 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -160,14 +160,18 @@ static int patch_nvhdmi(struct hda_codec *codec) */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10de0002"); +MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Nvidia HDMI HD-audio codec"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9065ebf9c065..82097790f6f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -77,6 +78,7 @@ enum { ALC260_ACER, ALC260_WILL, ALC260_REPLACER_672V, + ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -103,6 +105,7 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, + ALC262_TYAN, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -238,6 +241,13 @@ enum { ALC883_MODEL_LAST, }; +/* styles of capture selection */ +enum { + CAPT_MUX = 0, /* only mux based */ + CAPT_MIX, /* only mixer based */ + CAPT_1MUX_MIX, /* first mux and other mixers */ +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -246,6 +256,7 @@ struct alc_spec { struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -269,13 +280,15 @@ struct alc_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; + hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */ + int dig_out_type; /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ + int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -293,7 +306,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ @@ -305,6 +318,9 @@ struct alc_spec { unsigned int jack_present: 1; unsigned int master_sw: 1; + /* other flags */ + unsigned int no_analog :1; /* digital I/O only */ + /* for virtual master */ hda_nid_t vmaster_nid; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -314,13 +330,6 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; - -#ifdef SND_HDA_NEEDS_RESUME -#define ALC_MAX_PINS 16 - unsigned int num_pins; - hda_nid_t pin_nids[ALC_MAX_PINS]; - unsigned int pin_cfgs[ALC_MAX_PINS]; -#endif }; /* @@ -336,6 +345,7 @@ struct alc_config_preset { hda_nid_t *dac_nids; hda_nid_t dig_out_nid; /* optional */ hda_nid_t hp_nid; /* optional */ + hda_nid_t *slave_dig_outs; unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; @@ -392,7 +402,8 @@ 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]; - if (spec->is_mix_capture) { + if (spec->capture_style && + !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -750,6 +761,24 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, #endif /* CONFIG_SND_DEBUG */ /* + * set up the input pin config (depending on the given auto-pin type) + */ +static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, + int auto_pin_type) +{ + unsigned int val = PIN_IN; + + if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { + unsigned int pincap; + pincap = snd_hda_query_pin_caps(codec, nid); + pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + if (pincap & AC_PINCAP_VREF_80) + val = PIN_VREF80; + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); +} + +/* */ static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) { @@ -810,6 +839,7 @@ static void setup_preset(struct alc_spec *spec, spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; spec->multiout.dig_out_nid = preset->dig_out_nid; + spec->multiout.slave_dig_outs = preset->slave_dig_outs; spec->multiout.hp_nid = preset->hp_nid; spec->num_mux_defs = preset->num_mux_defs; @@ -921,7 +951,7 @@ static void alc_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } #else -#define alc_mic_automute(codec) /* NOP */ +#define alc_mic_automute(codec) do {} while(0) /* NOP */ #endif /* disabled */ /* unsolicited event for HP jack sensing */ @@ -952,7 +982,7 @@ static void alc888_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); 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); - if ((tmp & 0xf0) == 2) + if ((tmp & 0xf0) == 0x20) /* alc888S-VC */ snd_hda_codec_read(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0x830); @@ -991,8 +1021,7 @@ static void alc_subsystem_id(struct hda_codec *codec, nid = 0x1d; if (codec->vendor_id == 0x10ec0260) nid = 0x17; - ass = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + ass = snd_hda_codec_get_pincfg(codec, nid); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ @@ -1037,6 +1066,7 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: @@ -1065,6 +1095,7 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: + case 0x10ec0887: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -1164,16 +1195,8 @@ static void alc_fix_pincfg(struct hda_codec *codec, return; cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) { - int i; - u32 val = cfg->val; - for (i = 0; i < 4; i++) { - snd_hda_codec_write(codec, cfg->nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, - val & 0xff); - val >>= 8; - } - } + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } /* @@ -1373,8 +1396,6 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -1481,8 +1502,6 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1502,11 +1521,11 @@ static int alc_cap_vol_info(struct snd_kcontrol *kcontrol, struct alc_spec *spec = codec->spec; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT); err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1517,11 +1536,11 @@ static int alc_cap_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct alc_spec *spec = codec->spec; int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[0], 3, 0, HDA_INPUT); err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1537,11 +1556,11 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); int err; - mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->control_mutex); kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], 3, 0, HDA_INPUT); err = func(kcontrol, ucontrol); - mutex_unlock(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_unlock(&codec->control_mutex); return err; } @@ -1576,8 +1595,7 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, snd_hda_mixer_amp_switch_put); } -#define DEFINE_CAPMIX(num) \ -static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ +#define _DEFINE_CAPMIX(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Capture Switch", \ @@ -1598,7 +1616,9 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .get = alc_cap_vol_get, \ .put = alc_cap_vol_put, \ .tlv = { .c = alc_cap_vol_tlv }, \ - }, \ + } + +#define _DEFINE_CAPSRC(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ /* .name = "Capture Source", */ \ @@ -1607,15 +1627,28 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .info = alc_mux_enum_info, \ .get = alc_mux_enum_get, \ .put = alc_mux_enum_put, \ - }, \ - { } /* end */ \ + } + +#define DEFINE_CAPMIX(num) \ +static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ + _DEFINE_CAPMIX(num), \ + _DEFINE_CAPSRC(num), \ + { } /* end */ \ +} + +#define DEFINE_CAPMIX_NOSRC(num) \ +static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \ + _DEFINE_CAPMIX(num), \ + { } /* end */ \ } /* up to three ADCs */ DEFINE_CAPMIX(1); DEFINE_CAPMIX(2); DEFINE_CAPMIX(3); - +DEFINE_CAPMIX_NOSRC(1); +DEFINE_CAPMIX_NOSRC(2); +DEFINE_CAPMIX_NOSRC(3); /* * ALC880 5-stack model @@ -1704,8 +1737,6 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -1882,13 +1913,6 @@ static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* additional mixers to alc880_asus_mixer */ -static struct snd_kcontrol_new alc880_pcbeep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ -}; - /* TCL S700 */ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -1921,8 +1945,6 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -1997,6 +2019,13 @@ static const char *alc_slave_sws[] = { static void alc_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new alc_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT), + { } /* end */ +}; + static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2018,11 +2047,13 @@ static int alc_build_controls(struct hda_codec *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; - spec->multiout.share_spdif = 1; + if (!spec->no_analog) { + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -2030,8 +2061,24 @@ static int alc_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = alc_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); @@ -2040,7 +2087,8 @@ static int alc_build_controls(struct hda_codec *codec) if (err < 0) return err; } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, alc_slave_sws); if (err < 0) @@ -2949,6 +2997,14 @@ static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -3032,7 +3088,8 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = { .ops = { .open = alc880_dig_playback_pcm_open, .close = alc880_dig_playback_pcm_close, - .prepare = alc880_dig_playback_pcm_prepare + .prepare = alc880_dig_playback_pcm_prepare, + .cleanup = alc880_dig_playback_pcm_cleanup }, }; @@ -3059,6 +3116,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; + if (spec->no_analog) + goto skip_analog; + info->name = spec->stream_name_analog; if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) @@ -3082,12 +3142,17 @@ static int alc_build_pcms(struct hda_codec *codec) } } + skip_analog: /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms = 2; + codec->slave_dig_outs = spec->multiout.slave_dig_outs; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out_type) + info->pcm_type = spec->dig_out_type; + else + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid && spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); @@ -3102,6 +3167,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->spdif_status_reset = 1; } + if (spec->no_analog) + return 0; + /* If the use of more than one ADC is requested for the current * model, configure a second analog capture-only PCM. */ @@ -3160,65 +3228,17 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); - codec->spec = NULL; /* to be sure */ + snd_hda_detach_beep_device(codec); } #ifdef SND_HDA_NEEDS_RESUME -static void store_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid, end_nid; - - 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; - if (wid_type != AC_WID_PIN) - continue; - if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids)) - break; - spec->pin_nids[spec->num_pins] = nid; - spec->pin_cfgs[spec->num_pins] = - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); - spec->num_pins++; - } -} - -static void resume_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t pin_nid = spec->pin_nids[i]; - unsigned int pin_config = spec->pin_cfgs[i]; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - } -} - static int alc_resume(struct hda_codec *codec) { - resume_pin_configs(codec); codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); return 0; } -#else -#define store_pin_configs(codec) #endif /* @@ -3557,7 +3577,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), @@ -3600,7 +3620,8 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */ + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), {} @@ -3780,7 +3801,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { - .mixers = { alc880_asus_mixer, alc880_pcbeep_mixer }, + .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), @@ -3818,8 +3839,7 @@ static struct alc_config_preset alc880_presets[] = { .init_hook = alc880_uniwill_p53_hp_automute, }, [ALC880_FUJITSU] = { - .mixers = { alc880_fujitsu_mixer, - alc880_pcbeep_mixer, }, + .mixers = { alc880_fujitsu_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_uniwill_p53_init_verbs, alc880_beep_init_verbs }, @@ -4112,7 +4132,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -4200,11 +4220,9 @@ 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)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC880_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC880_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -4219,7 +4237,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; + int i, err; static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -4250,8 +4268,23 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + /* 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; @@ -4261,9 +4294,8 @@ static int alc880_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -4278,21 +4310,33 @@ static void alc880_auto_init(struct hda_codec *codec) alc_inithook(codec); } -/* - * OK, here we have finally the patch for ALC880 - */ - static void set_capture_mixer(struct alc_spec *spec) { - static struct snd_kcontrol_new *caps[3] = { - alc_capture_mixer1, - alc_capture_mixer2, - alc_capture_mixer3, + static struct snd_kcontrol_new *caps[2][3] = { + { alc_capture_mixer_nosrc1, + alc_capture_mixer_nosrc2, + alc_capture_mixer_nosrc3 }, + { alc_capture_mixer1, + alc_capture_mixer2, + alc_capture_mixer3 }, }; - if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) - spec->cap_mixer = caps[spec->num_adc_nids - 1]; + if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { + int mux; + if (spec->input_mux && spec->input_mux->num_items > 1) + mux = 1; + else + mux = 0; + spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; + } } +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +/* + * OK, here we have finally the patch for ALC880 + */ + static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; @@ -4328,6 +4372,12 @@ static int patch_alc880(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); @@ -4354,6 +4404,7 @@ static int patch_alc880(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -4461,6 +4512,26 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = { }, }, }; + +/* Maxdata Favorit 100XS */ +static struct hda_input_mux alc260_favorit100_capture_sources[2] = { + { + .num_items = 2, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + }, + }, + { + .num_items = 3, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + { "Mixer", 0x5 }, + }, + }, +}; + /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -4503,12 +4574,6 @@ static struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc260_pc_beep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), - { } /* end */ -}; - /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec, hda_nid_t hp, hda_nid_t line, @@ -4700,8 +4765,6 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), { } /* end */ @@ -4746,8 +4809,18 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + { } /* end */ +}; + +/* Maxdata Favorit 100XS: one output and one input (0x12) jack + */ +static struct snd_kcontrol_new alc260_favorit100_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), { } /* end */ }; @@ -4765,8 +4838,6 @@ static struct snd_kcontrol_new alc260_will_mixer[] = { ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), { } /* end */ }; @@ -5124,6 +5195,89 @@ static struct hda_verb alc260_acer_init_verbs[] = { { } }; +/* Initialisation sequence for Maxdata Favorit 100XS + * (adapted from Acer init verbs). + */ +static struct hda_verb alc260_favorit100_init_verbs[] = { + /* GPIO 0 enables the output jack. + * Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Line/Mic input jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right + * (no equiv mixer ctrl) + */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to mic to match ALSA's default state. + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + static struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -5270,8 +5424,6 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), @@ -5469,7 +5621,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -5544,11 +5696,9 @@ static void alc260_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 (nid >= 0x12) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC260_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC260_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -5621,7 +5771,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC260_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -5629,9 +5779,8 @@ static int alc260_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -5668,6 +5817,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { [ALC260_ACER] = "acer", [ALC260_WILL] = "will", [ALC260_REPLACER_672V] = "replacer", + [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -5677,6 +5827,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), @@ -5699,8 +5850,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { static struct alc_config_preset alc260_presets[] = { [ALC260_BASIC] = { .mixers = { alc260_base_output_mixer, - alc260_input_mixer, - alc260_pc_beep_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5779,6 +5929,18 @@ static struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), .input_mux = alc260_acer_capture_sources, }, + [ALC260_FAVORIT100] = { + .mixers = { alc260_favorit100_mixer }, + .init_verbs = { alc260_favorit100_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), + .input_mux = alc260_favorit100_capture_sources, + }, [ALC260_WILL] = { .mixers = { alc260_will_mixer }, .init_verbs = { alc260_init_verbs, alc260_will_verbs }, @@ -5855,6 +6017,12 @@ static int patch_alc260(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); @@ -5880,6 +6048,7 @@ static int patch_alc260(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); spec->vmaster_nid = 0x08; @@ -6051,8 +6220,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { 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), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6079,8 +6246,6 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6132,8 +6297,6 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { 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("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6242,8 +6405,10 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), + /* FIXME: this looks suspicious... HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + */ { } /* end */ }; @@ -6875,19 +7040,9 @@ static void alc882_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]; - unsigned int vref; if (!nid) continue; - vref = PIN_IN; - if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - unsigned int pincap; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if ((pincap >> AC_PINCAP_VREF_SHIFT) & - AC_PINCAP_VREF_80) - vref = PIN_VREF80; - } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, vref); + 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, @@ -6898,18 +7053,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) static void alc882_auto_init_input_src(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; 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 @@ -6922,8 +7080,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) break; } } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute); + /* 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); } } } @@ -7012,12 +7182,15 @@ static int patch_alc882(struct hda_codec *codec) 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 */ case 0x106b3600: /* Macbook 3.1 */ + case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; default: @@ -7049,6 +7222,12 @@ static int patch_alc882(struct hda_codec *codec) } } + 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]); @@ -7069,7 +7248,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->is_mix_capture = 1; /* matrix-style capture */ + spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -7086,6 +7265,7 @@ static int patch_alc882(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -7137,10 +7317,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = { 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 */ @@ -7358,8 +7542,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { 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), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7422,8 +7604,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { 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), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7447,8 +7627,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { 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), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7473,8 +7651,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { 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), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7498,8 +7674,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { 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), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7907,36 +8081,83 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_6st_dell_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc888_3st_hp_front_automute(struct hda_codec *codec) +{ + unsigned int present, bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_3st_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc888_3st_hp_front_automute(codec); + break; + } +} + static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ - { } -}; - -static struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - { } + { } /* end */ }; +/* + * 2ch mode + */ static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } + { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc888_3st_hp_4ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 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, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ static struct hda_verb alc888_3st_hp_6ch_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 }, { 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, 0x01 }, + { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[2] = { +static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, + { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, }; @@ -8197,7 +8418,7 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_HP_EVENT: - printk("hp_event\n"); + /* printk(KERN_DEBUG "hp_event\n"); */ alc888_6st_dell_front_automute(codec); break; } @@ -8456,24 +8677,37 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_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), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x013e, "Acer Aspire 4930G", ALC888_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ + SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", + ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), + SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), + SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", + ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", + ALC888_ACER_ASPIRE_4930G), + /* default Acer */ + 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, 0x1873, "Asus M90V", ALC888_ASUS_M90V), 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(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_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), @@ -8505,9 +8739,11 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { 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, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), + 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, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550", + ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), @@ -8522,10 +8758,16 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { 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, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; +static hda_nid_t alc1200_slave_dig_outs[] = { + ALC883_DIGOUT_NID, 0, +}; + static struct alc_config_preset alc883_presets[] = { [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, @@ -8761,6 +9003,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, + .unsol_event = alc888_3st_hp_unsol_event, + .init_hook = alc888_3st_hp_front_automute, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -8866,6 +9110,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC1200_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc1200_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -8933,11 +9178,9 @@ static void alc883_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 (alc883_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); - if (nid != ALC883_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC883_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -8952,6 +9195,8 @@ static int alc883_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; if (err < 0) return err; @@ -8965,6 +9210,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* 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; + } + } + return 1; /* config found */ } @@ -9016,6 +9281,12 @@ static int patch_alc883(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); @@ -9028,14 +9299,36 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC888 Analog"; spec->stream_name_digital = "ALC888 Digital"; } + 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->capture_style = CAPT_MIX; /* matrix-style capture */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; spec->stream_name_digital = "ALC889 Digital"; + 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; + spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style + capture */ break; default: spec->stream_name_analog = "ALC883 Analog"; spec->stream_name_digital = "ALC883 Digital"; + 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->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -9046,15 +9339,9 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - 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->is_mix_capture = 1; /* matrix-style capture */ if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -9107,8 +9394,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), @@ -9129,8 +9414,6 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), { } /* end */ @@ -9239,8 +9522,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT), HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT), { } /* end */ @@ -9269,8 +9550,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -9418,6 +9697,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_tyan_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, 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_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_tyan_verbs[] = { + /* Headphone automute */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* P11 AUX_IN, white 4-pin connector */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, + + {} +}; + +/* unsolicited event for HP jack sensing */ +static void alc262_tyan_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute line output on ATX panel */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute line output if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_tyan_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_tyan_automute(codec); +} + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -9884,8 +10224,6 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -10457,8 +10795,14 @@ static int alc262_parse_auto_config(struct hda_codec *codec) alc262_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ + } err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -10468,8 +10812,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + dig_only: + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; @@ -10478,13 +10825,12 @@ static int alc262_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -10526,20 +10872,19 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", + [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series", + ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", + ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", + ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), @@ -10557,17 +10902,18 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN", - ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ + SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), - SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), - SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", + ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), @@ -10783,6 +11129,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_TYAN] = { + .mixers = { alc262_tyan_mixer }, + .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x02, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_tyan_unsol_event, + .init_hook = alc262_tyan_automute, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -10835,6 +11194,14 @@ static int patch_alc262(struct hda_codec *codec) } } + if (!spec->no_analog) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + } + if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); @@ -10846,7 +11213,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -10863,8 +11230,10 @@ static int patch_alc262(struct hda_codec *codec) spec->capsrc_nids = alc262_capsrc_nids; } } - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + if (!spec->no_analog) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -11244,19 +11613,13 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, static struct hda_verb alc268_base_init_verbs[] = { /* Unmute DAC0-1 and set vol = 0 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* * Set up output mixers (0x0c - 0x0e) */ /* set vol=0 to output mixers */ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11275,9 +11638,7 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11299,10 +11660,8 @@ static struct hda_verb alc268_base_init_verbs[] = { */ static struct hda_verb alc268_volume_init_verbs[] = { /* set output DAC */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, @@ -11310,16 +11669,12 @@ static struct hda_verb alc268_volume_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {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)}, {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11518,7 +11873,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -11612,9 +11967,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec) alc268_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ - + } err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -11624,25 +11984,26 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; + dig_only: /* digital only support output */ - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - if (spec->autocfg.speaker_pins[0] != 0x1d) + if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -11689,6 +12050,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), + SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), @@ -11703,7 +12065,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11785,7 +12147,8 @@ static struct alc_config_preset alc268_presets[] = { }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, - alc268_capture_alt_mixer }, + alc268_beep_mixer, + alc268_capture_alt_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11854,7 +12217,7 @@ static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; int board_config; - int err; + int i, has_beep, err; spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -11903,15 +12266,30 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_digital_playback = &alc268_pcm_digital_playback; - if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) - /* override the amp caps for beep generator */ - snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, + has_beep = 0; + for (i = 0; i < spec->num_mixers; i++) { + if (spec->mixers[i] == alc268_beep_mixer) { + has_beep = 1; + break; + } + } + + if (has_beep) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) + /* override the amp caps for beep generator */ + snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, (0x0c << AC_AMPCAP_OFFSET_SHIFT) | (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) | (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) | (0 << AC_AMPCAP_MUTE_SHIFT)); + } - if (!spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); int i; @@ -11992,8 +12370,6 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), @@ -12020,8 +12396,6 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12045,8 +12419,6 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12083,13 +12455,6 @@ static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { { } /* end */ }; -/* beep control */ -static struct snd_kcontrol_new alc269_beep_mixer[] = { - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), - { } /* end */ -}; - static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -12489,7 +12854,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, */ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; imux->items[imux->num_items].label = "Int Mic"; imux->items[imux->num_items].index = 0x05; imux->num_items++; @@ -12507,13 +12872,34 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, #define alc269_pcm_digital_playback alc880_pcm_digital_playback #define alc269_pcm_digital_capture alc880_pcm_digital_capture +static struct hda_pcm_stream alc269_44k_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ + .ops = { + .open = alc880_playback_pcm_open, + .prepare = alc880_playback_pcm_prepare, + .cleanup = alc880_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream alc269_44k_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ +}; + /* * BIOS auto configuration */ static int alc269_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -12530,22 +12916,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - /* create a beep mixer control if the pin 0x1d isn't assigned */ - for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++) - if (spec->autocfg.input_pins[i] == 0x1d) - break; - if (i >= ARRAY_SIZE(spec->autocfg.input_pins)) - add_mixer(spec, alc269_beep_mixer); - add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; /* set default input source */ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, @@ -12555,10 +12934,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -12655,7 +13033,7 @@ static struct alc_config_preset alc269_presets[] = { .init_hook = alc269_eeepc_dmic_inithook, }, [ALC269_FUJITSU] = { - .mixers = { alc269_fujitsu_mixer, alc269_beep_mixer }, + .mixers = { alc269_fujitsu_mixer }, .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, alc269_eeepc_dmic_init_verbs }, @@ -12720,13 +13098,26 @@ static int patch_alc269(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); spec->stream_name_analog = "ALC269 Analog"; - spec->stream_analog_playback = &alc269_pcm_analog_playback; - spec->stream_analog_capture = &alc269_pcm_analog_capture; - + if (codec->subsystem_id == 0x17aa3bf8) { + /* Due to a hardware problem on Lenovo Ideadpad, we need to + * fix the sample rate of analog I/O to 44.1kHz + */ + spec->stream_analog_playback = &alc269_44k_pcm_analog_playback; + spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; + } else { + spec->stream_analog_playback = &alc269_pcm_analog_playback; + spec->stream_analog_capture = &alc269_pcm_analog_capture; + } spec->stream_name_digital = "ALC269 Digital"; spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; @@ -12736,6 +13127,7 @@ static int patch_alc269(struct hda_codec *codec) spec->capsrc_nids = alc269_capsrc_nids; if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -12986,8 +13378,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = { static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT), { } }; @@ -13461,7 +13851,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -13548,12 +13938,8 @@ static void alc861_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 (nid >= 0x0c && nid <= 0x11) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - } + if (nid >= 0x0c && nid <= 0x11) + alc_set_input_pin(codec, nid, i); } } @@ -13589,7 +13975,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; if (spec->kctls.list) @@ -13598,13 +13984,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -13813,6 +14198,12 @@ static int patch_alc861(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); @@ -13824,6 +14215,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; @@ -13980,9 +14373,6 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14006,9 +14396,6 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14047,8 +14434,6 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -14359,9 +14744,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; @@ -14523,11 +14906,9 @@ 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)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC861VD_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC861VD_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -14693,7 +15074,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; if (spec->kctls.list) @@ -14702,13 +15083,12 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -14759,6 +15139,12 @@ static int patch_alc861vd(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); @@ -14781,9 +15167,10 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -14972,8 +15359,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -14995,8 +15380,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15972,56 +16355,55 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), - SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), - SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), - SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), + SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), + /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/ + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), - SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), + SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), + SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), - SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), - SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), + SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", + ALC663_ASUS_H13), {} }; @@ -16341,7 +16723,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - /* printk("DAC nid=%x\n",nid); */ + /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */ /* specify the DAC as the extra output */ if (!spec->multiout.hp_nid) spec->multiout.hp_nid = nid; @@ -16371,26 +16753,58 @@ 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 alc_spec *spec, +static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + 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; - 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->num_items++; + 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; @@ -16440,7 +16854,6 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc662_is_input_pin(nid) alc880_is_input_pin(nid) #define ALC662_PIN_CD_NID ALC880_PIN_CD_NID static void alc662_auto_init_analog_input(struct hda_codec *codec) @@ -16450,12 +16863,10 @@ 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(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); - if (nid != ALC662_PIN_CD_NID) + if (alc662_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)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -16493,20 +16904,20 @@ static int alc662_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) @@ -16516,7 +16927,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -16568,6 +16978,12 @@ static int patch_alc662(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); @@ -16591,10 +17007,14 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); + if (codec->vendor_id == 0x10ec0662) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + else + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 35b83dc6e19e..b5e108aa8f63 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -43,6 +43,7 @@ enum { }; enum { + STAC_AUTO, STAC_REF, STAC_9200_OQO, STAC_9200_DELL_D21, @@ -55,20 +56,24 @@ enum { STAC_9200_DELL_M25, STAC_9200_DELL_M26, STAC_9200_DELL_M27, - STAC_9200_GATEWAY, + STAC_9200_M4, + STAC_9200_M4_2, STAC_9200_PANASONIC, STAC_9200_MODELS }; enum { + STAC_9205_AUTO, STAC_9205_REF, STAC_9205_DELL_M42, STAC_9205_DELL_M43, STAC_9205_DELL_M44, + STAC_9205_EAPD, STAC_9205_MODELS }; enum { + STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, STAC_DELL_M6_AMIC, @@ -79,28 +84,40 @@ enum { }; enum { + STAC_92HD83XXX_AUTO, STAC_92HD83XXX_REF, + STAC_92HD83XXX_PWR_REF, + STAC_DELL_S14, STAC_92HD83XXX_MODELS }; enum { + STAC_92HD71BXX_AUTO, STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, STAC_DELL_M4_3, STAC_HP_M4, + STAC_HP_DV5, + STAC_HP_HDX, STAC_92HD71BXX_MODELS }; enum { + STAC_925x_AUTO, STAC_925x_REF, + STAC_M1, + STAC_M1_2, + STAC_M2, STAC_M2_2, - STAC_MA6, - STAC_PA6, + STAC_M3, + STAC_M5, + STAC_M6, STAC_925x_MODELS }; enum { + STAC_922X_AUTO, STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, @@ -128,6 +145,7 @@ enum { }; enum { + STAC_927X_AUTO, STAC_D965_REF_NO_JD, /* no jack-detection */ STAC_D965_REF, STAC_D965_3ST, @@ -137,6 +155,12 @@ enum { STAC_927X_MODELS }; +enum { + STAC_9872_AUTO, + STAC_9872_VAIO, + STAC_9872_MODELS +}; + struct sigmatel_event { hda_nid_t nid; unsigned char type; @@ -160,6 +184,7 @@ struct sigmatel_spec { unsigned int alt_switch: 1; unsigned int hp_detect: 1; unsigned int spdif_mute: 1; + unsigned int check_volume_offset:1; /* gpio lines */ unsigned int eapd_mask; @@ -172,6 +197,7 @@ struct sigmatel_spec { unsigned int stream_delay; /* analog loopback */ + struct snd_kcontrol_new *aloopback_ctl; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -196,6 +222,8 @@ struct sigmatel_spec { hda_nid_t hp_dacs[5]; hda_nid_t speaker_dacs[5]; + int volume_offset; + /* capture */ hda_nid_t *adc_nids; unsigned int num_adcs; @@ -217,7 +245,6 @@ struct sigmatel_spec { /* pin widgets */ hda_nid_t *pin_nids; unsigned int num_pins; - unsigned int *pin_configs; /* codec specific stuff */ struct hda_verb *init; @@ -328,7 +355,11 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x10, 0x40, + 0x03, 0x0c, 0x20, 0x40, +}; + +static hda_nid_t stac92hd83xxx_amp_nids[1] = { + 0xc, }; static hda_nid_t stac92hd71bxx_pwr_nids[3] = { @@ -389,6 +420,10 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +static hda_nid_t stac927x_slave_dig_outs[2] = { + 0x1f, 0, +}; + static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -461,15 +496,21 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd83xxx_pin_nids[14] = { +static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x1d, 0x1e, 0x1f, 0x20 + 0x0f, 0x10, 0x11, 0x1f, 0x20, }; -static hda_nid_t stac92hd71bxx_pin_nids[11] = { + +#define STAC92HD71BXX_NUM_PINS 13 +static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x00, + 0x00, 0x14, 0x18, 0x19, 0x1e, + 0x1f, 0x20, 0x27 +}; +static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, - 0x1f, + 0x1f, 0x20, 0x27 }; static hda_nid_t stac927x_pin_nids[14] = { @@ -831,13 +872,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - /* start of config #1 */ - { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3}, - - /* start of config #2 */ - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, @@ -847,26 +884,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#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}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +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)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -875,6 +911,8 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* mute the master volume */ + { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, {} }; @@ -945,16 +983,6 @@ static struct hda_verb stac9205_core_init[] = { .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ } -#define STAC_INPUT_SOURCE(cnt) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Input Source", \ - .count = cnt, \ - .info = stac92xx_mux_enum_info, \ - .get = stac92xx_mux_enum_get, \ - .put = stac92xx_mux_enum_put, \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -969,7 +997,6 @@ static struct hda_verb stac9205_core_init[] = { static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), - STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), { } /* end */ @@ -994,8 +1021,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1005,9 +1030,22 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { +static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + {} +}; +static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + {} +}; + +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), @@ -1032,8 +1070,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1085,9 +1121,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1113,10 +1146,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), +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), @@ -1126,16 +1160,14 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { - STAC_INPUT_SOURCE(1), + 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[] = { - STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -1144,9 +1176,13 @@ static struct snd_kcontrol_new stac9205_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac9205_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), + {} +}; + /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { - STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), @@ -1157,9 +1193,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - STAC_INPUT_SOURCE(3), - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -1171,6 +1204,11 @@ static struct snd_kcontrol_new stac927x_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac927x_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), + {} +}; + static struct snd_kcontrol_new stac_dmux_mixer = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Input Source", @@ -1196,10 +1234,7 @@ static const char *slave_vols[] = { "LFE Playback Volume", "Side Playback Volume", "Headphone Playback Volume", - "Headphone Playback Volume", "Speaker Playback Volume", - "External Speaker Playback Volume", - "Speaker2 Playback Volume", NULL }; @@ -1210,10 +1245,7 @@ static const char *slave_sws[] = { "LFE Playback Switch", "Side Playback Switch", "Headphone Playback Switch", - "Headphone Playback Switch", "Speaker Playback Switch", - "External Speaker Playback Switch", - "Speaker2 Playback Switch", "IEC958 Playback Switch", NULL }; @@ -1283,6 +1315,8 @@ static int stac92xx_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], HDA_OUTPUT, vmaster_tlv); + /* correct volume offset */ + vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, slave_vols); if (err < 0) @@ -1295,6 +1329,13 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } + if (spec->aloopback_ctl && + snd_hda_get_bool_hint(codec, "loopback") == 1) { + err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl); + if (err < 0) + return err; + } + stac92xx_free_kctls(codec); /* no longer needed */ /* create jack input elements */ @@ -1334,7 +1375,16 @@ static unsigned int ref9200_pin_configs[8] = { 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -/* +static unsigned int gateway9200_m4_pin_configs[8] = { + 0x400000fe, 0x404500f4, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; +static unsigned int gateway9200_m4_2_pin_configs[8] = { + 0x400000fe, 0x404500f4, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; + +/* STAC 9200 pin configs for 102801A8 102801DE @@ -1464,10 +1514,13 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_9200_DELL_M25] = dell9200_m25_pin_configs, [STAC_9200_DELL_M26] = dell9200_m26_pin_configs, [STAC_9200_DELL_M27] = dell9200_m27_pin_configs, + [STAC_9200_M4] = gateway9200_m4_pin_configs, + [STAC_9200_M4_2] = gateway9200_m4_2_pin_configs, [STAC_9200_PANASONIC] = ref9200_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { + [STAC_AUTO] = "auto", [STAC_REF] = "ref", [STAC_9200_OQO] = "oqo", [STAC_9200_DELL_D21] = "dell-d21", @@ -1480,7 +1533,8 @@ static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_9200_DELL_M25] = "dell-m25", [STAC_9200_DELL_M26] = "dell-m26", [STAC_9200_DELL_M27] = "dell-m27", - [STAC_9200_GATEWAY] = "gateway", + [STAC_9200_M4] = "gateway-m4", + [STAC_9200_M4_2] = "gateway-m4-2", [STAC_9200_PANASONIC] = "panasonic", }; @@ -1488,6 +1542,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8, "unknown Dell", STAC_9200_DELL_D21), @@ -1550,11 +1606,9 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC), /* Gateway machines needs EAPD to be set on resume */ - SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY), - SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*", - STAC_9200_GATEWAY), - SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707", - STAC_9200_GATEWAY), + SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_M4), + SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*", STAC_9200_M4_2), + SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707", STAC_9200_M4_2), /* OQO Mobile */ SND_PCI_QUIRK(0x1106, 0x3288, "OQO Model 2", STAC_9200_OQO), {} /* terminator */ @@ -1565,44 +1619,87 @@ static unsigned int ref925x_pin_configs[8] = { 0x90a70320, 0x02214210, 0x01019020, 0x9033032e, }; -static unsigned int stac925x_MA6_pin_configs[8] = { - 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x90a70320, 0x90100211, 0x400003f1, 0x9033032e, +static unsigned int stac925xM1_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925x_PA6_pin_configs[8] = { - 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x50a103f0, 0x90100211, 0x400003f1, 0x9033032e, +static unsigned int stac925xM1_2_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, +}; + +static unsigned int stac925xM2_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; static unsigned int stac925xM2_2_pin_configs[8] = { - 0x40c003f3, 0x424503f2, 0x04180011, 0x02a19020, - 0x50a103f0, 0x90100212, 0x400003f1, 0x9033032e, + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, +}; + +static unsigned int stac925xM3_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x503303f3, +}; + +static unsigned int stac925xM5_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, +}; + +static unsigned int stac925xM6_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x90330320, }; static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { [STAC_REF] = ref925x_pin_configs, + [STAC_M1] = stac925xM1_pin_configs, + [STAC_M1_2] = stac925xM1_2_pin_configs, + [STAC_M2] = stac925xM2_pin_configs, [STAC_M2_2] = stac925xM2_2_pin_configs, - [STAC_MA6] = stac925x_MA6_pin_configs, - [STAC_PA6] = stac925x_PA6_pin_configs, + [STAC_M3] = stac925xM3_pin_configs, + [STAC_M5] = stac925xM5_pin_configs, + [STAC_M6] = stac925xM6_pin_configs, }; static const char *stac925x_models[STAC_925x_MODELS] = { + [STAC_925x_AUTO] = "auto", [STAC_REF] = "ref", + [STAC_M1] = "m1", + [STAC_M1_2] = "m1-2", + [STAC_M2] = "m2", [STAC_M2_2] = "m2-2", - [STAC_MA6] = "m6", - [STAC_PA6] = "pa6", + [STAC_M3] = "m3", + [STAC_M5] = "m5", + [STAC_M6] = "m6", +}; + +static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { + SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_M2), + SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_M5), + SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_M1), + SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_M2), + SND_PCI_QUIRK(0x107b, 0x0367, "Gateway MX6453", STAC_M1_2), + /* Not sure about the brand name for those */ + SND_PCI_QUIRK(0x107b, 0x0281, "Gateway mobile", STAC_M1), + SND_PCI_QUIRK(0x107b, 0x0507, "Gateway mobile", STAC_M3), + SND_PCI_QUIRK(0x107b, 0x0281, "Gateway mobile", STAC_M6), + SND_PCI_QUIRK(0x107b, 0x0685, "Gateway mobile", STAC_M2_2), + {} /* terminator */ }; static struct snd_pci_quirk stac925x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF), - SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_REF), - SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_REF), - SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6), - SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6), - SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2), + + /* Default table for unknown ID */ + SND_PCI_QUIRK(0x1002, 0x437b, "Gateway mobile", STAC_M2_2), + {} /* terminator */ }; @@ -1629,6 +1726,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", [STAC_DELL_M6_AMIC] = "dell-m6-amic", @@ -1641,6 +1739,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, @@ -1664,50 +1764,68 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd83xxx_pin_configs[14] = { +static unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x01451160, 0x98560170, }; +static unsigned int dell_s14_pin_configs[10] = { + 0x02214030, 0x02211010, 0x02a19020, 0x01014050, + 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160, + 0x40f000f0, 0x40f000f0, +}; + static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, + [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, + [STAC_DELL_S14] = dell_s14_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_AUTO] = "auto", [STAC_92HD83XXX_REF] = "ref", + [STAC_92HD83XXX_PWR_REF] = "mic-ref", + [STAC_DELL_S14] = "dell-s14", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD71BXX_REF), + "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, + "unknown Dell", STAC_DELL_S14), {} /* terminator */ }; -static unsigned int ref92hd71bxx_pin_configs[11] = { +static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_1_pin_configs[11] = { +static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_2_pin_configs[11] = { +static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_3_pin_configs[11] = { +static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -1716,28 +1834,39 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_2] = dell_m4_2_pin_configs, [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, + [STAC_HP_DV5] = NULL, + [STAC_HP_HDX] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_AUTO] = "auto", [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", + [STAC_HP_DV5] = "hp-dv5", + [STAC_HP_HDX] = "hp-hdx", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, - "HP dv5", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, - "HP dv7", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, - "HP dv7", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, + "HP", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, + "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, + "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, + "HP HDX", STAC_HP_HDX), /* HDX18 */ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, - "unknown HP", STAC_HP_M4), + "HP mini 1000", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, + "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -1889,6 +2018,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { }; static const char *stac922x_models[STAC_922X_MODELS] = { + [STAC_922X_AUTO] = "auto", [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", @@ -1916,6 +2046,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D945_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D945_REF), /* Intel 945G based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101, "Intel D945G", STAC_D945GTP3), @@ -1969,6 +2101,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "Intel D945P", STAC_D945GTP3), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), + /* other intel */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0204, + "Intel D945", STAC_D945_REF), /* other systems */ /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, @@ -1993,31 +2128,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7, "Dell XPS M1210", STAC_922X_DELL_M82), /* ECS/PC Chips boards */ - SND_PCI_QUIRK(0x1019, 0x2144, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2608, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2633, - "ECS/PC chips P17G/1333", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2811, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2812, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2813, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2814, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2815, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2816, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2817, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2818, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2819, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2820, + SND_PCI_QUIRK_MASK(0x1019, 0xf000, 0x2000, "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; @@ -2060,6 +2171,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { }; static const char *stac927x_models[STAC_927X_MODELS] = { + [STAC_927X_AUTO] = "auto", [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", [STAC_D965_3ST] = "3stack", @@ -2072,26 +2184,16 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D965_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D965_REF), /* Intel 946 based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), /* 965 based 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2100, + "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000, + "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), @@ -2107,15 +2209,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2300, + "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, + "Intel D965", STAC_D965_5ST), {} /* terminator */ }; @@ -2168,19 +2265,25 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs, [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs, [STAC_9205_DELL_M44] = dell_9205_m44_pin_configs, + [STAC_9205_EAPD] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { + [STAC_9205_AUTO] = "auto", [STAC_9205_REF] = "ref", [STAC_9205_DELL_M42] = "dell-m42", [STAC_9205_DELL_M43] = "dell-m43", [STAC_9205_DELL_M44] = "dell-m44", + [STAC_9205_EAPD] = "eapd", }; static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_9205_REF), + /* Dell */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, @@ -2211,101 +2314,24 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, "Dell Vostro 1500", STAC_9205_DELL_M42), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), {} /* terminator */ }; -static int stac92xx_save_bios_config_regs(struct hda_codec *codec) +static void stac92xx_set_config_regs(struct hda_codec *codec, + unsigned int *pincfgs) { int i; struct sigmatel_spec *spec = codec->spec; - - kfree(spec->pin_configs); - spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t nid = spec->pin_nids[i]; - unsigned int pin_cfg; - - pin_cfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", - nid, pin_cfg); - spec->pin_configs[i] = pin_cfg; - } - - return 0; -} - -static void stac92xx_set_config_reg(struct hda_codec *codec, - hda_nid_t pin_nid, unsigned int pin_config) -{ - int i; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - i = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", - pin_nid, i); -} -static void stac92xx_set_config_regs(struct hda_codec *codec) -{ - int i; - struct sigmatel_spec *spec = codec->spec; - - if (!spec->pin_configs) - return; + if (!pincfgs) + return; for (i = 0; i < spec->num_pins; i++) - stac92xx_set_config_reg(codec, spec->pin_nids[i], - spec->pin_configs[i]); -} - -static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!pins) - return stac92xx_save_bios_config_regs(codec); - - kfree(spec->pin_configs); - spec->pin_configs = kmemdup(pins, - spec->num_pins * sizeof(*pins), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - stac92xx_set_config_regs(codec); - return 0; -} - -static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid, - unsigned int cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_nids[i] == nid) { - spec->pin_configs[i] = cfg; - stac92xx_set_config_reg(codec, nid, cfg); - break; - } - } + if (spec->pin_nids[i] && pincfgs[i]) + snd_hda_codec_set_pincfg(codec, spec->pin_nids[i], + pincfgs[i]); } /* @@ -2370,6 +2396,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct sigmatel_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture callbacks @@ -2414,7 +2448,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = { .ops = { .open = stac92xx_dig_playback_pcm_open, .close = stac92xx_dig_playback_pcm_close, - .prepare = stac92xx_dig_playback_pcm_prepare + .prepare = stac92xx_dig_playback_pcm_prepare, + .cleanup = stac92xx_dig_playback_pcm_cleanup }, }; @@ -2469,6 +2504,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec) info->name = "STAC92xx Analog"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs; @@ -2484,7 +2521,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + info->pcm_type = spec->autocfg.dig_out_type[0]; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; @@ -2500,8 +2537,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_100) return AC_PINCTL_VREF_100; @@ -2676,22 +2712,37 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) +static struct snd_kcontrol_new * +stac_control_new(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + const char *name) { struct snd_kcontrol_new *knew; snd_array_init(&spec->kctls, sizeof(*knew), 32); knew = snd_array_new(&spec->kctls); if (!knew) - return -ENOMEM; + return NULL; *knew = *ktemp; - knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) + if (!knew->name) { + /* roolback */ + memset(knew, 0, sizeof(*knew)); + spec->kctls.alloced--; + return NULL; + } + return knew; +} + +static int stac92xx_add_control_temp(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + int idx, const char *name, + unsigned long val) +{ + struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name); + if (!knew) return -ENOMEM; + knew->index = idx; knew->private_value = val; return 0; } @@ -2713,6 +2764,29 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, return stac92xx_add_control_idx(spec, type, 0, name, val); } +static struct snd_kcontrol_new stac_input_src_temp = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, +}; + +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->num_adcs || imux->num_items <= 1) + return 0; /* no need for input source control */ + knew = stac_control_new(spec, &stac_input_src_temp, + stac_input_src_temp.name); + if (!knew) + return -ENOMEM; + knew->count = spec->num_adcs; + return 0; +} + /* check whether the line-input can be used as line-out */ static hda_nid_t check_line_out_switch(struct hda_codec *codec) { @@ -2724,7 +2798,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec) if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; nid = cfg->input_pins[AUTO_PIN_LINE]; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; return 0; @@ -2743,12 +2817,11 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec) mic_pin = AUTO_PIN_MIC; for (;;) { hda_nid_t nid = cfg->input_pins[mic_pin]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; } @@ -2796,8 +2869,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); for (j = 0; j < conn_len; j++) { - wcaps = snd_hda_param_read(codec, conn[j], - AC_PAR_AUDIO_WIDGET_CAP); + wcaps = get_wcaps(codec, conn[j]); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* we check only analog outputs */ if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) @@ -2812,6 +2884,16 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } + /* if all DACs are already assigned, connect to the primary DAC */ + if (conn_len > 1) { + for (j = 0; j < conn_len; j++) { + if (conn[j] == spec->multiout.dac_nids[0]) { + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, j); + break; + } + } + } return 0; } @@ -2852,6 +2934,26 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) add_spec_dacs(spec, dac); } + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) { + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = dac; + else + add_spec_extra_dacs(spec, dac); + } + spec->hp_dacs[i] = dac; + } + + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) + add_spec_extra_dacs(spec, dac); + spec->speaker_dacs[i] = dac; + } + /* add line-in as output */ nid = check_line_out_switch(codec); if (nid) { @@ -2879,26 +2981,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } } - for (i = 0; i < cfg->hp_outs; i++) { - nid = cfg->hp_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = dac; - else - add_spec_extra_dacs(spec, dac); - } - spec->hp_dacs[i] = dac; - } - - for (i = 0; i < cfg->speaker_outs; i++) { - nid = cfg->speaker_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) - add_spec_extra_dacs(spec, dac); - spec->speaker_dacs[i] = dac; - } - snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", spec->multiout.num_dacs, spec->multiout.dac_nids[0], @@ -2911,24 +2993,47 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } /* create volume control/switch for the given prefx type */ -static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs) +static int create_controls_idx(struct hda_codec *codec, const char *pfx, + int idx, hda_nid_t nid, int chs) { + struct sigmatel_spec *spec = codec->spec; char name[32]; int err; + if (!spec->check_volume_offset) { + unsigned int caps, step, nums, db_scale; + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + step = (caps & AC_AMPCAP_STEP_SIZE) >> + AC_AMPCAP_STEP_SIZE_SHIFT; + step = (step + 1) * 25; /* in .01dB unit */ + nums = (caps & AC_AMPCAP_NUM_STEPS) >> + AC_AMPCAP_NUM_STEPS_SHIFT; + db_scale = nums * step; + /* if dB scale is over -64dB, and finer enough, + * let's reduce it to half + */ + if (db_scale > 6400 && nums >= 0x1f) + spec->volume_offset = nums / 2; + spec->check_volume_offset = 1; + } + sprintf(name, "%s Playback Volume", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name, + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, + spec->volume_offset)); if (err < 0) return err; sprintf(name, "%s Playback Switch", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); if (err < 0) return err; return 0; } +#define create_controls(codec, pfx, nid, chs) \ + create_controls_idx(codec, pfx, 0, nid, chs) + static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) { if (spec->multiout.num_dacs > 4) { @@ -2954,40 +3059,32 @@ static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) return 1; } -static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - if (spec->autocfg.line_outs != 1) - return 0; - if (spec->multiout.hp_nid == nid) - return 0; - for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) - if (spec->multiout.extra_out_nid[i] == nid) - return 0; - return 1; -} - -/* 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) +/* Create output controls + * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT) + */ +static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, + const hda_nid_t *dac_nids, + int type) { struct sigmatel_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid; int i, err; unsigned int wid_caps; - for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) { - nid = spec->multiout.dac_nids[i]; - if (i == 2) { + for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + nid = dac_nids[i]; + if (!nid) + continue; + if (type != AUTO_PIN_HP_OUT && i == 2) { /* Center/LFE */ - err = create_controls(spec, "Center", nid, 1); + err = create_controls(codec, "Center", nid, 1); if (err < 0) return err; - err = create_controls(spec, "LFE", nid, 2); + err = create_controls(codec, "LFE", nid, 2); if (err < 0) return err; @@ -3003,23 +3100,47 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } else { - const char *name = chname[i]; - /* if it's a single DAC, assign a better name */ - if (!i && is_unique_dac(spec, nid)) { - switch (cfg->line_out_type) { - case AUTO_PIN_HP_OUT: - name = "Headphone"; - break; - case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - break; - } + const char *name; + int idx; + switch (type) { + case AUTO_PIN_HP_OUT: + name = "Headphone"; + idx = i; + break; + case AUTO_PIN_SPEAKER_OUT: + name = "Speaker"; + idx = i; + break; + default: + name = chname[i]; + idx = 0; + break; } - err = create_controls(spec, name, nid, 3); + err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } } } + 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) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + + err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, + spec->multiout.dac_nids, + cfg->line_out_type); + if (err < 0) + return err; if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { err = stac92xx_add_control(spec, @@ -3054,40 +3175,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid; - int i, err, nums; + int err; + + err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins, + spec->hp_dacs, AUTO_PIN_HP_OUT); + if (err < 0) + return err; + + err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, + spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT); + if (err < 0) + return err; - nums = 0; - for (i = 0; i < cfg->hp_outs; i++) { - static const char *pfxs[] = { - "Headphone", "Headphone2", "Headphone3", - }; - unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->hp_dacs[i]; - if (!nid) - continue; - err = create_controls(spec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } - nums = 0; - for (i = 0; i < cfg->speaker_outs; i++) { - static const char *pfxs[] = { - "Speaker", "External Speaker", "Speaker2", - }; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->speaker_dacs[i]; - if (!nid) - continue; - err = create_controls(spec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } return 0; } @@ -3296,11 +3395,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, unsigned int wcaps; unsigned int def_conf; - def_conf = snd_hda_codec_read(codec, - spec->dmic_nids[i], - 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0); + def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; @@ -3424,6 +3519,7 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec) static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) { struct sigmatel_spec *spec = codec->spec; + int hp_swap = 0; int err; if ((err = snd_hda_parse_pin_def_config(codec, @@ -3451,6 +3547,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_outs = spec->autocfg.hp_outs; spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; spec->autocfg.hp_outs = 0; + hp_swap = 1; } if (spec->autocfg.mono_out_pin) { int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & @@ -3506,13 +3603,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = stac92xx_auto_fill_dac_nids(codec); if (err < 0) return err; + err = stac92xx_auto_create_multi_out_ctls(codec, + &spec->autocfg); + if (err < 0) + return err; } - err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg); - - if (err < 0) - return err; - /* setup analog beep controls */ if (spec->anabeep_nid > 0) { err = stac92xx_auto_create_beep_ctls(codec, @@ -3545,12 +3641,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out #endif err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); - if (err < 0) return err; - err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); + /* All output parsing done, now restore the swapped hp pins */ + if (hp_swap) { + memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, + sizeof(spec->autocfg.hp_pins)); + spec->autocfg.hp_outs = spec->autocfg.line_outs; + spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; + spec->autocfg.line_outs = 0; + } + err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -3579,11 +3682,15 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return err; } + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = dig_out; if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; @@ -3646,9 +3753,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; unsigned int defcfg; - defcfg = snd_hda_codec_read(codec, pin, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); + defcfg = snd_hda_codec_get_pincfg(codec, pin); if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); @@ -3661,7 +3766,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, } if (lfe_pin) { - err = create_controls(spec, "LFE", lfe_pin, 1); + err = create_controls(codec, "LFE", lfe_pin, 1); if (err < 0) return err; } @@ -3692,7 +3797,11 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return err; } - if (spec->autocfg.dig_out_pin) + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; @@ -3748,8 +3857,7 @@ static int stac92xx_add_jack(struct hda_codec *codec, #ifdef CONFIG_SND_JACK struct sigmatel_spec *spec = codec->spec; struct sigmatel_jack *jack; - int def_conf = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); char name[32]; @@ -3864,6 +3972,36 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +/* override some hints from the hwdep entry */ +static void stac_store_hints(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + const char *p; + int val; + + val = snd_hda_get_bool_hint(codec, "hp_detect"); + if (val >= 0) + spec->hp_detect = val; + p = snd_hda_get_hint(codec, "gpio_mask"); + if (p) { + spec->gpio_mask = simple_strtoul(p, NULL, 0); + spec->eapd_mask = spec->gpio_dir = spec->gpio_data = + spec->gpio_mask; + } + p = snd_hda_get_hint(codec, "gpio_dir"); + if (p) + spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_data"); + if (p) + spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "eapd_mask"); + if (p) + spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + val = snd_hda_get_bool_hint(codec, "eapd_switch"); + if (val >= 0) + spec->eapd_switch = val; +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3880,6 +4018,9 @@ static int stac92xx_init(struct hda_codec *codec) spec->adc_nids[i], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* override some hints */ + stac_store_hints(codec); + /* set up GPIO */ gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. @@ -3929,8 +4070,7 @@ static int stac92xx_init(struct hda_codec *codec) pinctl); } } - conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + 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); @@ -3942,8 +4082,8 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], AC_PINCTL_IN_EN); - if (cfg->dig_out_pin) - stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, + if (cfg->dig_out_pins[0]) + stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0], AC_PINCTL_OUT_EN); if (cfg->dig_in_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, @@ -3971,8 +4111,7 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); def_conf = get_defcfg_connect(def_conf); /* skip any ports that don't have jacks since presence * detection is useless */ @@ -4026,7 +4165,6 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - kfree(spec->pin_configs); stac92xx_free_jacks(codec); snd_array_free(&spec->events); @@ -4037,7 +4175,9 @@ static void stac92xx_free(struct hda_codec *codec) static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int flag) { - unsigned int pin_ctl = snd_hda_codec_read(codec, nid, + unsigned int old_ctl, pin_ctl; + + pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); if (pin_ctl & AC_PINCTL_IN_EN) { @@ -4051,14 +4191,17 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, return; } + old_ctl = pin_ctl; /* if setting pin direction bits, clear the current direction bits first */ if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl | flag); + pin_ctl |= flag; + if (old_ctl != pin_ctl) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl); } static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -4066,9 +4209,10 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, { unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl & ~flag); + if (pin_ctl & flag) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl & ~flag); } static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) @@ -4163,8 +4307,19 @@ static void stac92xx_hp_detect(struct hda_codec *codec) continue; if (presence) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); +#if 0 /* FIXME */ +/* Resetting the pinctl like below may lead to (a sort of) regressions + * on some devices since they use the HP pin actually for line/speaker + * outs although the default pin config shows a different pin (that is + * wrong and useless). + * + * So, it's basically a problem of default pin configs, likely a BIOS issue. + * But, disabling the code below just works around it, and I'm too tired of + * bug reports with such devices... + */ else stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val); +#endif /* FIXME */ } } @@ -4320,7 +4475,6 @@ static int stac92xx_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - stac92xx_set_config_regs(codec); stac92xx_init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -4331,6 +4485,37 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } + +/* + * using power check for controlling mute led of HP HDX notebooks + * check for mute state only on Speakers (nid = 0x10) + * + * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise + * the LED is NOT working properly ! + */ + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x08; /* orange */ + else + spec->gpio_data |= 0x08; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} +#endif + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -4369,16 +4554,11 @@ static int patch_stac9200(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, stac9200_cfg_tbl); - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9200_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4390,7 +4570,8 @@ static int patch_stac9200(struct hda_codec *codec) spec->num_adcs = 1; spec->num_pwrs = 0; - if (spec->board_config == STAC_9200_GATEWAY || + if (spec->board_config == STAC_9200_M4 || + spec->board_config == STAC_9200_M4_2 || spec->board_config == STAC_9200_OQO) spec->init = stac9200_eapd_init; else @@ -4408,6 +4589,12 @@ static int patch_stac9200(struct hda_codec *codec) return err; } + /* CF-74 has no headphone detection, and the driver should *NOT* + * do detection and HP/speaker toggle because the hardware does it. + */ + if (spec->board_config == STAC_9200_PANASONIC) + spec->hp_detect = 0; + codec->patch_ops = stac92xx_patch_ops; return 0; @@ -4425,21 +4612,26 @@ static int patch_stac925x(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; - spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS, + + /* Check first for codec ID */ + spec->board_config = snd_hda_check_board_codec_sid_config(codec, + STAC_925x_MODELS, + stac925x_models, + stac925x_codec_id_cfg_tbl); + + /* Now checks for PCI ID, if codec ID is not found */ + if (spec->board_config < 0) + spec->board_config = snd_hda_check_board_config(codec, + STAC_925x_MODELS, stac925x_models, stac925x_cfg_tbl); again: - if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac925x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4517,17 +4709,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) stac92hd73xx_models, stac92hd73xx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD73XX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd73xx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } num_dacs = snd_hda_get_connections(codec, 0x0a, conn, STAC92HD73_DAC_COUNT + 2) - 1; @@ -4541,14 +4728,18 @@ again: 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; } spec->multiout.dac_nids = spec->dac_nids; @@ -4587,18 +4778,18 @@ again: spec->init = dell_m6_core_init; switch (spec->board_config) { case STAC_DELL_M6_AMIC: /* Analog Mics */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + 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 */ - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + 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 */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + 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; @@ -4658,7 +4849,10 @@ static struct hda_input_mux stac92hd83xxx_dmux = { static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; + int num_dacs; + hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4672,23 +4866,17 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->dmux_nids = stac92hd83xxx_dmux_nids; spec->adc_nids = 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; - switch (codec->vendor_id) { - case 0x111d7605: - break; - default: - spec->num_pwrs--; - spec->init++; /* switch to config #2 */ - } - 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; @@ -4697,16 +4885,20 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD83XXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd83xxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; + + switch (codec->vendor_id) { + case 0x111d7604: + case 0x111d7605: + if (spec->board_config == STAC_92HD83XXX_PWR_REF) + break; + spec->num_pwrs = 0; + break; } err = stac92xx_parse_auto_config(codec, 0x1d, 0); @@ -4725,6 +4917,23 @@ again: return err; } + switch (spec->board_config) { + case STAC_DELL_S14: + nid = 0xf; + break; + default: + nid = 0xe; + break; + } + + num_dacs = snd_hda_get_connections(codec, nid, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + /* set port X to select the last DAC + */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + codec->patch_ops = stac92xx_patch_ops; codec->proc_widget_hook = stac92hd_proc_hook; @@ -4732,7 +4941,16 @@ again: return 0; } -static struct hda_input_mux stac92hd71bxx_dmux = { +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 }, @@ -4742,10 +4960,67 @@ static struct hda_input_mux stac92hd71bxx_dmux = { } }; +/* get the pin connection (fixed, none, etc) */ +static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int cfg; + + cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); + return get_defcfg_connect(cfg); +} + +static int stac92hd71bxx_connected_ports(struct hda_codec *codec, + hda_nid_t *nids, int num_nids) +{ + struct sigmatel_spec *spec = codec->spec; + int idx, num; + unsigned int def_conf; + + for (num = 0; num < num_nids; num++) { + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == nids[num]) + break; + if (idx >= spec->num_pins) + break; + def_conf = stac_get_defcfg_connect(codec, idx); + if (def_conf == AC_JACK_PORT_NONE) + break; + } + return num; +} + +static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, + hda_nid_t dig0pin) +{ + struct sigmatel_spec *spec = codec->spec; + int idx; + + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == dig0pin) + break; + if ((idx + 2) >= spec->num_pins) + return 0; + + /* dig1pin case */ + if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE) + return 2; + + /* dig0pin + dig2pin case */ + if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE) + return 2; + if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE) + return 1; + else + return 0; +} + 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) @@ -4753,27 +5028,32 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = stac92xx_patch_ops; - spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pins = STAC92HD71BXX_NUM_PINS; + switch (codec->vendor_id) { + case 0x111d76b6: + case 0x111d76b7: + spec->pin_nids = stac92hd71bxx_pin_nids_4port; + break; + case 0x111d7603: + case 0x111d7608: + /* On 92HD75Bx 0x27 isn't a pin nid */ + spec->num_pins--; + /* fallthrough */ + default: + spec->pin_nids = stac92hd71bxx_pin_nids_6port; + } spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pin_nids = stac92hd71bxx_pin_nids; - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, - sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd71bxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } if (spec->board_config > STAC_92HD71BXX_REF) { /* GPIO0 = EAPD */ @@ -4782,16 +5062,34 @@ again: spec->gpio_data = 0x01; } + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nids = stac92hd71bxx_dmux_nids; + switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: + unmute_init++; + /* 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 */ @@ -4818,7 +5116,15 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; - stac_change_pin_config(codec, 0xf, 0x40f000f0); + unmute_init++; + snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); + snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); + stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; + 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) @@ -4828,12 +5134,23 @@ 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; 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; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write_cache(codec, unmute_init); + + spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; @@ -4841,18 +5158,17 @@ again: spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; - spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->smux_nids = stac92hd71bxx_smux_nids; spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ - stac_change_pin_config(codec, 0x0e, 0x01813040); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); /* fallthru */ @@ -4865,21 +5181,38 @@ again: case STAC_DELL_M4_3: spec->num_dmics = 1; spec->num_smuxes = 0; - spec->num_dmuxes = 0; + spec->num_dmuxes = 1; + break; + case STAC_HP_DV5: + snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); + stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + break; + case STAC_HP_HDX: + spec->num_dmics = 1; + spec->num_dmuxes = 1; + spec->num_smuxes = 1; + /* + * For controlling MUTE LED on HP HDX16/HDX18 notebooks, + * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* orange/white mute led on GPIO3, orange=0, white=1 */ + spec->gpio_mask |= 0x08; + spec->gpio_dir |= 0x08; + spec->gpio_data |= 0x08; /* set to white */ + + /* register check_power_status callback. */ + codec->patch_ops.check_power_status = + stac92xx_hp_hdx_check_power_status; +#endif break; - default: - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); }; spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) - spec->private_dimux.num_items += - spec->num_dmics - - (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); + spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; - err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + err = stac92xx_parse_auto_config(codec, 0x21, 0); if (!err) { if (spec->board_config < 0) { printk(KERN_WARNING "hda_codec: No auto-config is " @@ -4954,17 +5287,12 @@ static int patch_stac922x(struct hda_codec *codec) } again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac922x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->adc_nids = stac922x_adc_nids; spec->mux_nids = stac922x_mux_nids; @@ -5015,24 +5343,19 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) { - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - "STAC927x, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); + else + stac92xx_set_config_regs(codec, stac927x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac927x_adc_nids; @@ -5061,15 +5384,15 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: case 0x1028022e: /* correct the device field to SPDIF out */ - stac_change_pin_config(codec, 0x21, 0x01442070); + snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070); break; }; /* configure the analog microphone on some laptops */ - stac_change_pin_config(codec, 0x0c, 0x90a79130); + snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ - stac_change_pin_config(codec, 0x0f, 0x0227011f); + snd_hda_codec_set_pincfg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ - stac_change_pin_config(codec, 0x0e, 0x02a79130); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ @@ -5096,6 +5419,7 @@ static int patch_stac927x(struct hda_codec *codec) } spec->num_pwrs = 0; + spec->aloopback_ctl = stac927x_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->eapd_switch = 1; @@ -5154,16 +5478,11 @@ static int patch_stac9205(struct hda_codec *codec) stac9205_models, stac9205_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9205_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac9205_adc_nids; @@ -5180,17 +5499,20 @@ static int patch_stac9205(struct hda_codec *codec) spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_ctl = stac9205_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - spec->eapd_switch = 1; + /* Turn on/off EAPD per HP plugging */ + if (spec->board_config != STAC_9205_EAPD) + spec->eapd_switch = 1; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ case STAC_9205_DELL_M43: /* Enable SPDIF in/out */ - stac_change_pin_config(codec, 0x1f, 0x01441030); - stac_change_pin_config(codec, 0x20, 0x1c410030); + snd_hda_codec_set_pincfg(codec, 0x1f, 0x01441030); + snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030); /* Enable unsol response for GPIO4/Dock HP connection */ err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01); @@ -5247,223 +5569,87 @@ static int patch_stac9205(struct hda_codec *codec) * STAC9872 hack */ -/* static config for Sony VAIO FE550G and Sony VAIO AR */ -static hda_nid_t vaio_dacs[] = { 0x2 }; -#define VAIO_HP_DAC 0x5 -static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; -static hda_nid_t vaio_mux_nids[] = { 0x15 }; - -static struct hda_input_mux vaio_mux = { - .num_items = 3, - .items = { - /* { "HP", 0x0 }, */ - { "Mic Jack", 0x1 }, - { "Internal Mic", 0x2 }, - { "PCM", 0x3 }, - } -}; - -static struct hda_verb vaio_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ +static struct hda_verb stac9872_core_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; -static struct hda_verb vaio_ar_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ -/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ -/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ - {} -}; - -static struct snd_kcontrol_new vaio_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ +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), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} + { } /* end */ }; -static struct snd_kcontrol_new vaio_ar_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} +static hda_nid_t stac9872_pin_nids[] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, + 0x11, 0x13, 0x14, }; -static struct hda_codec_ops stac9872_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, -#ifdef SND_HDA_NEEDS_RESUME - .resume = stac92xx_resume, -#endif +static hda_nid_t stac9872_adc_nids[] = { + 0x8 /*,0x6*/ }; -static int stac9872_vaio_init(struct hda_codec *codec) -{ - int err; - - err = stac92xx_init(codec); - if (err < 0) - return err; - if (codec->patch_ops.unsol_event) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); - return 0; -} - -static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) -{ - if (get_pin_presence(codec, 0x0a)) { - stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - } else { - stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - } -} - -static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case STAC_HP_EVENT: - stac9872_vaio_hp_detect(codec, res); - break; - } -} - -static struct hda_codec_ops stac9872_vaio_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac9872_vaio_init, - .free = stac92xx_free, - .unsol_event = stac9872_vaio_unsol_event, -#ifdef CONFIG_PM - .resume = stac92xx_resume, -#endif +static hda_nid_t stac9872_mux_nids[] = { + 0x15 }; -enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */ - CXD9872RD_VAIO, - /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */ - STAC9872AK_VAIO, - /* Unknown. id=0x83847661 and subsys=0x104D1200. */ - STAC9872K_VAIO, - /* AR Series. id=0x83847664 and subsys=104D1300 */ - CXD9872AKD_VAIO, - STAC_9872_MODELS, +static unsigned int stac9872_vaio_pin_configs[9] = { + 0x03211020, 0x411111f0, 0x411111f0, 0x03a15030, + 0x411111f0, 0x90170110, 0x411111f0, 0x411111f0, + 0x90a7013e }; static const char *stac9872_models[STAC_9872_MODELS] = { - [CXD9872RD_VAIO] = "vaio", - [CXD9872AKD_VAIO] = "vaio-ar", + [STAC_9872_AUTO] = "auto", + [STAC_9872_VAIO] = "vaio", +}; + +static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { + [STAC_9872_VAIO] = stac9872_vaio_pin_configs, }; static struct snd_pci_quirk stac9872_cfg_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO), - SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO), - {} + {} /* terminator */ }; static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; - int board_config; + int err; - board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, - stac9872_models, - stac9872_cfg_tbl); - if (board_config < 0) - /* unknown config, let generic-parser do its job... */ - return snd_hda_parse_generic_codec(codec); - spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - switch (board_config) { - case CXD9872RD_VAIO: - case STAC9872AK_VAIO: - case STAC9872K_VAIO: - spec->mixer = vaio_mixer; - spec->init = vaio_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->adc_nids = vaio_adcs; - spec->num_pwrs = 0; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_vaio_patch_ops; - break; - - case CXD9872AKD_VAIO: - spec->mixer = vaio_ar_mixer; - spec->init = vaio_ar_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->num_pwrs = 0; - spec->adc_nids = vaio_adcs; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_patch_ops; - break; - } + spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, + stac9872_models, + stac9872_cfg_tbl); + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, " + "using BIOS defaults\n"); + else + stac92xx_set_config_regs(codec, + stac9872_brd_tbl[spec->board_config]); + + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; + spec->multiout.dac_nids = spec->dac_nids; + spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); + 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; + + err = stac92xx_parse_auto_config(codec, 0x10, 0x12); + if (err < 0) { + stac92xx_free(codec); + return -EINVAL; + } + spec->input_mux = &spec->private_imux; + codec->patch_ops = stac92xx_patch_ops; return 0; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c761394cbe84..b25a5cc637d6 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1308,16 +1308,13 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) unsigned int def_conf; unsigned char seqassoc; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { if (seqassoc == 0xff) { def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - def_conf >> 24); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } } @@ -1354,7 +1351,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; @@ -1827,7 +1824,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1709_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; @@ -2371,7 +2368,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; @@ -2836,7 +2833,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; @@ -3155,7 +3152,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1702_DIGOUT_NID; spec->extra_dig_out_nid = 0x1B; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index bab1c700f497..3dd63f1cda53 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -458,7 +458,7 @@ static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id) u16 pbkstatus; struct snd_pcm_substream *substream; pbkstatus = inw(ICEDS(ice, INTSTAT)); - /* printk("pbkstatus = 0x%x\n", pbkstatus); */ + /* printk(KERN_DEBUG "pbkstatus = 0x%x\n", pbkstatus); */ for (idx = 0; idx < 6; idx++) { if ((pbkstatus & (3 << (idx * 2))) == 0) continue; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 7ff36d3f0f44..128510e77a78 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -241,6 +241,8 @@ get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream) struct snd_rawmidi_substream, list); } +static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable); + static void vt1724_midi_write(struct snd_ice1712 *ice) { struct snd_rawmidi_substream *s; @@ -254,6 +256,11 @@ static void vt1724_midi_write(struct snd_ice1712 *ice) for (i = 0; i < count; ++i) outb(buffer[i], ICEREG1724(ice, MPU_DATA)); } + /* mask irq when all bytes have been transmitted. + * enabled again in output_trigger when the new data comes in. + */ + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, + !snd_rawmidi_transmit_empty(s)); } static void vt1724_midi_read(struct snd_ice1712 *ice) @@ -272,31 +279,34 @@ static void vt1724_midi_read(struct snd_ice1712 *ice) } } -static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, - u8 flag, int enable) +/* call with ice->reg_lock */ +static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable) { - struct snd_ice1712 *ice = substream->rmidi->private_data; - u8 mask; - - spin_lock_irq(&ice->reg_lock); - mask = inb(ICEREG1724(ice, IRQMASK)); + u8 mask = inb(ICEREG1724(ice, IRQMASK)); if (enable) mask &= ~flag; else mask |= flag; outb(mask, ICEREG1724(ice, IRQMASK)); +} + +static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, + u8 flag, int enable) +{ + struct snd_ice1712 *ice = substream->rmidi->private_data; + + spin_lock_irq(&ice->reg_lock); + enable_midi_irq(ice, flag, enable); spin_unlock_irq(&ice->reg_lock); } static int vt1724_midi_output_open(struct snd_rawmidi_substream *s) { - vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1); return 0; } static int vt1724_midi_output_close(struct snd_rawmidi_substream *s) { - vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); return 0; } @@ -311,6 +321,7 @@ static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up) vt1724_midi_write(ice); } else { ice->midi_output = 0; + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); } spin_unlock_irqrestore(&ice->reg_lock, flags); } @@ -320,6 +331,7 @@ static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s) struct snd_ice1712 *ice = s->rmidi->private_data; unsigned long timeout; + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); /* 32 bytes should be transmitted in less than about 12 ms */ timeout = jiffies + msecs_to_jiffies(15); do { @@ -389,24 +401,24 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status &= status_mask; if (status == 0) break; + spin_lock(&ice->reg_lock); if (++timeout > 10) { status = inb(ICEREG1724(ice, IRQSTAT)); printk(KERN_ERR "ice1724: Too long irq loop, " "status = 0x%x\n", status); if (status & VT1724_IRQ_MPU_TX) { printk(KERN_ERR "ice1724: Disabling MPU_TX\n"); - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_TX, - ICEREG1724(ice, IRQMASK)); + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); } + spin_unlock(&ice->reg_lock); break; } handled = 1; if (status & VT1724_IRQ_MPU_TX) { - spin_lock(&ice->reg_lock); if (ice->midi_output) vt1724_midi_write(ice); - spin_unlock(&ice->reg_lock); + else + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); /* Due to mysterical reasons, MPU_TX is always * generated (and can't be cleared) when a PCM * playback is going. So let's ignore at the @@ -415,15 +427,14 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status_mask &= ~VT1724_IRQ_MPU_TX; } if (status & VT1724_IRQ_MPU_RX) { - spin_lock(&ice->reg_lock); if (ice->midi_input) vt1724_midi_read(ice); else vt1724_midi_clear_rx(ice); - spin_unlock(&ice->reg_lock); } /* ack MPU irq */ outb(status, ICEREG1724(ice, IRQSTAT)); + spin_unlock(&ice->reg_lock); if (status & VT1724_IRQ_MTPCM) { /* * Multi-track PCM @@ -745,7 +756,14 @@ static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream) spin_unlock_irq(&ice->reg_lock); - /* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */ + /* + printk(KERN_DEBUG "pro prepare: ch = %d, addr = 0x%x, " + "buffer = 0x%x, period = 0x%x\n", + substream->runtime->channels, + (unsigned int)substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream)); + */ return 0; } @@ -2122,7 +2140,9 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, wait_i2c_busy(ice); val = inb(ICEREG1724(ice, I2C_DATA)); mutex_unlock(&ice->i2c_mutex); - /* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */ + /* + printk(KERN_DEBUG "i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); + */ return val; } @@ -2131,7 +2151,9 @@ void snd_vt1724_write_i2c(struct snd_ice1712 *ice, { mutex_lock(&ice->i2c_mutex); wait_i2c_busy(ice); - /* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */ + /* + printk(KERN_DEBUG "i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); + */ outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(data, ICEREG1724(ice, I2C_DATA)); outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index c51659b9caf6..fd948bfd9aef 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -345,8 +345,9 @@ static int juli_mute_put(struct snd_kcontrol *kcontrol, new_gpio = old_gpio & ~((unsigned int) kcontrol->private_value); } - /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \ - new_gpio 0x%x\n", + /* printk(KERN_DEBUG + "JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, " + "new_gpio 0x%x\n", (unsigned int)ucontrol->value.integer.value[0], old_gpio, new_gpio); */ if (old_gpio != new_gpio) { diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 48d3679292a7..2a8e5cd8f2d8 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -133,8 +133,10 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME; /* due to possible conflicts with stac9460_set_rate_val, mutexing */ mutex_lock(&spec->mute_mutex); - /*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, - ucontrol->value.integer.value[0]);*/ + /* + printk(KERN_DEBUG "Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, + ucontrol->value.integer.value[0]); + */ change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]); mutex_unlock(&spec->mute_mutex); return change; @@ -185,7 +187,10 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el change = (ovol != nvol); if (change) { ovol = (0x7f - nvol) | (tmp & 0x80); - /*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/ + /* + printk(KERN_DEBUG "DAC Volume: reg 0x%02x: 0x%02x\n", + idx, ovol); + */ stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); } return change; @@ -344,7 +349,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) for (idx = 0; idx < 7 ; ++idx) changed[idx] = stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 0); - /*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); udelay(10); /* unmuting - only originally unmuted dacs - diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 671ff65db029..57648810eaf1 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip) int time = 100; if (chip->buggy_semaphore) return 0; /* just ignore ... */ - while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) + while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) udelay(1); if (! time && ! chip->in_ac97_init) snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n"); @@ -689,7 +689,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> ichdev->pos_shift); #if 0 - printk("bdbar[%i] = 0x%x [0x%x]\n", + printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); #endif } @@ -701,8 +701,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags; ichdev->position = 0; #if 0 - printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n", - ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1); + printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, " + "period_size1 = 0x%x\n", + ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, + ichdev->fragsize1); #endif /* clear interrupts */ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); @@ -768,7 +770,8 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich ichdev->lvi_frag %= ichdev->frags; ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf + ichdev->lvi_frag * ichdev->fragsize1); #if 0 - printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n", + printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, " + "all = 0x%x, 0x%x\n", ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2], ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port), inl(port + 4), inb(port + ICH_REG_OFF_CR)); @@ -2287,23 +2290,23 @@ static void do_ali_reset(struct intel8x0 *chip) iputdword(chip, ICHREG(ALI_INTERRUPTSR), 0x00000000); } -static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) -{ - unsigned long end_time; - unsigned int cnt, status, nstatus; - - /* put logic to right state */ - /* first clear status bits */ - status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; - if (chip->device_type == DEVICE_NFORCE) - status |= ICH_NVSPINT; - cnt = igetdword(chip, ICHREG(GLOB_STA)); - iputdword(chip, ICHREG(GLOB_STA), cnt & status); +#ifdef CONFIG_SND_AC97_POWER_SAVE +static struct snd_pci_quirk ich_chip_reset_mode[] = { + SND_PCI_QUIRK(0x1014, 0x051f, "Thinkpad R32", 1), + { } /* end */ +}; +static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip) +{ + unsigned int cnt; /* ACLink on, 2 channels */ + + if (snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) + return -EIO; + cnt = igetdword(chip, ICHREG(GLOB_CNT)); cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); -#ifdef CONFIG_SND_AC97_POWER_SAVE + /* do cold reset - the full ac97 powerdown may leave the controller * in a warm state but actually it cannot communicate with the codec. */ @@ -2312,22 +2315,58 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) udelay(10); iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD); msleep(1); + return 0; +} +#define snd_intel8x0_ich_chip_can_cold_reset(chip) \ + (!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) #else +#define snd_intel8x0_ich_chip_cold_reset(chip) 0 +#define snd_intel8x0_ich_chip_can_cold_reset(chip) (0) +#endif + +static int snd_intel8x0_ich_chip_reset(struct intel8x0 *chip) +{ + unsigned long end_time; + unsigned int cnt; + /* ACLink on, 2 channels */ + cnt = igetdword(chip, ICHREG(GLOB_CNT)); + cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); end_time = (jiffies + (HZ / 4)) + 1; do { if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) - goto __ok; + return 0; schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n", igetdword(chip, ICHREG(GLOB_CNT))); return -EIO; +} + +static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) +{ + unsigned long end_time; + unsigned int status, nstatus; + unsigned int cnt; + int err; + + /* put logic to right state */ + /* first clear status bits */ + status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; + if (chip->device_type == DEVICE_NFORCE) + status |= ICH_NVSPINT; + cnt = igetdword(chip, ICHREG(GLOB_STA)); + iputdword(chip, ICHREG(GLOB_STA), cnt & status); + + if (snd_intel8x0_ich_chip_can_cold_reset(chip)) + err = snd_intel8x0_ich_chip_cold_reset(chip); + else + err = snd_intel8x0_ich_chip_reset(chip); + if (err < 0) + return err; - __ok: -#endif if (probing) { /* wait for any codec ready status. * Once it becomes ready it should remain ready diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 33a843c19316..6ec0fc50d6be 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -411,7 +411,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size)); bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> chip->pcm_pos_shift); - // printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + /* + printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n", + idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + */ } ichdev->frags = ichdev->size / ichdev->fragsize; } @@ -421,8 +424,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags; ichdev->position = 0; #if 0 - printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n", - ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1); + printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, " + "period_size1 = 0x%x\n", + ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, + ichdev->fragsize1); #endif /* clear interrupts */ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); @@ -465,7 +470,8 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic ichdev->lvi_frag * ichdev->fragsize1); #if 0 - printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n", + printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], " + "prefetch = %i, all = 0x%x, 0x%x\n", ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2], ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port), inl(port + 4), inb(port + ICH_REG_OFF_CR)); diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index bfc19e36c4b6..c1eb84a14c42 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, /* set the format to the board */ err = mixart_set_format(stream, format); if(err < 0) { + mutex_unlock(&mgr->setup_mutex); return err; } diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 3782b52bc0e8..4cf4cd8c939c 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -345,8 +345,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET )); /* motherboard xilinx status 5 will say that the board is performing a reset */ - if( status_xilinx == 5 ) { - snd_printk( KERN_ERR "miXart is resetting !\n"); + if (status_xilinx == 5) { + snd_printk(KERN_ERR "miXart is resetting !\n"); return -EAGAIN; /* try again later */ } @@ -354,13 +354,14 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_XLX_INDEX: /* xilinx already loaded ? */ - if( status_xilinx == 4 ) { - snd_printk( KERN_DEBUG "xilinx is already loaded !\n"); + if (status_xilinx == 4) { + snd_printk(KERN_DEBUG "xilinx is already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_xilinx != 0 ) { - snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx); + if (status_xilinx != 0) { + snd_printk(KERN_ERR "xilinx load error ! status = %d\n", + status_xilinx); return -EIO; /* modprob -r may help ? */ } @@ -389,21 +390,23 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_ELF_INDEX: - if( status_elf == 4 ) { - snd_printk( KERN_DEBUG "elf file already loaded !\n"); + if (status_elf == 4) { + snd_printk(KERN_DEBUG "elf file already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_elf != 0 ) { - snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf); + if (status_elf != 0) { + snd_printk(KERN_ERR "elf load error ! status = %d\n", + status_elf); return -EIO; /* modprob -r may help ? */ } /* wait for xilinx status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */ if (err < 0) { - snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n"); + snd_printk(KERN_ERR "xilinx was not loaded or " + "could not be started\n"); return err; } @@ -424,7 +427,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for elf status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "elf could not be started\n"); + snd_printk(KERN_ERR "elf could not be started\n"); return err; } @@ -437,15 +440,16 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw default: /* elf and xilinx should be loaded */ - if( (status_elf != 4) || (status_xilinx != 4) ) { - printk( KERN_ERR "xilinx or elf not successfully loaded\n"); + if (status_elf != 4 || status_xilinx != 4) { + printk(KERN_ERR "xilinx or elf not " + "successfully loaded\n"); return -EIO; /* modprob -r may help ? */ } /* wait for daughter detection != 0 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "error starting elf file\n"); + snd_printk(KERN_ERR "error starting elf file\n"); return err; } @@ -460,8 +464,9 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw return -EINVAL; /* daughter should be idle */ - if( status_daught != 0 ) { - printk( KERN_ERR "daughter load error ! status = %d\n", status_daught); + if (status_daught != 0) { + printk(KERN_ERR "daughter load error ! status = %d\n", + status_daught); return -EIO; /* modprob -r may help ? */ } @@ -480,7 +485,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for status == 2 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board load error\n"); + snd_printk(KERN_ERR "daughter board load error\n"); return err; } @@ -502,7 +507,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for daughter status == 3 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board could not be initialised\n"); + snd_printk(KERN_ERR + "daughter board could not be initialised\n"); return err; } @@ -512,7 +518,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* first communication with embedded */ err = mixart_first_init(mgr); if (err < 0) { - snd_printk( KERN_ERR "miXart could not be set up\n"); + snd_printk(KERN_ERR "miXart could not be set up\n"); return err; } @@ -581,16 +587,6 @@ MODULE_FIRMWARE("mixart/miXart8AES.xlx"); /* miXart hwdep interface id string */ #define SND_MIXART_HWDEP_ID "miXart Loader" -static int mixart_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int mixart_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int mixart_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -643,8 +639,6 @@ int snd_mixart_setup_firmware(struct mixart_mgr *mgr) hw->iface = SNDRV_HWDEP_IFACE_MIXART; hw->private_data = mgr; - hw->ops.open = mixart_hwdep_open; - hw->ops.release = mixart_hwdep_release; hw->ops.dsp_status = mixart_hwdep_dsp_status; hw->ops.dsp_load = mixart_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 98c6a8c65d81..6c870c12a177 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -676,7 +676,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip) if (chip->uart_input_count >= 2 && chip->uart_input[chip->uart_input_count - 2] == 'O' && chip->uart_input[chip->uart_input_count - 1] == 'K') { - printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:"); + printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n"); print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, chip->uart_input, chip->uart_input_count); chip->uart_input_count = 0; @@ -899,6 +899,7 @@ static const struct oxygen_model model_xonar_hdav = { .dac_channels = 8, .dac_volume_min = 0x0f, .dac_volume_max = 0xff, + .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 84131a916c92..69d87dee6995 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -97,12 +97,12 @@ struct pcxhr_mgr { int capture_chips; int fw_file_set; int firmware_num; - int is_hr_stereo:1; - int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ - int board_has_analog:1; /* if 0 the board is digital only */ - int board_has_mic:1; /* if 1 the board has microphone input */ - int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ - int mono_capture:1; /* if 1 the board does mono capture */ + unsigned int is_hr_stereo:1; + unsigned int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ + unsigned int board_has_analog:1; /* if 0 the board is digital only */ + unsigned int board_has_mic:1; /* if 1 the board has microphone input */ + unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ + unsigned int mono_capture:1; /* if 1 the board does mono capture */ struct snd_dma_buffer hostport; diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 592743a298b0..17cb1233a903 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -471,16 +471,6 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, return 0; } -static int pcxhr_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int pcxhr_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) { int err; @@ -495,8 +485,6 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) hw->iface = SNDRV_HWDEP_IFACE_PCXHR; hw->private_data = mgr; - hw->ops.open = pcxhr_hwdep_open; - hw->ops.release = pcxhr_hwdep_release; hw->ops.dsp_status = pcxhr_hwdep_dsp_status; hw->ops.dsp_load = pcxhr_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 05b3f795a168..314e73531bd1 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ @@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); 26*3 values are read in ss mode 14*3 in ds mode, with no gap between values */ -#define HDSP_9652_peakBase 7164 +#define HDSP_9652_peakBase 7164 #define HDSP_9652_rmsBase 4096 /* c.f. the hdsp_9632_meters_t struct */ @@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */ #define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */ #define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */ -#define HDSP_SyncRef2 (1<<13) -#define HDSP_SPDIFInputSelect0 (1<<14) -#define HDSP_SPDIFInputSelect1 (1<<15) -#define HDSP_SyncRef0 (1<<16) +#define HDSP_SyncRef2 (1<<13) +#define HDSP_SPDIFInputSelect0 (1<<14) +#define HDSP_SPDIFInputSelect1 (1<<15) +#define HDSP_SyncRef0 (1<<16) #define HDSP_SyncRef1 (1<<17) -#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ +#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ #define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */ #define HDSP_Midi0InterruptEnable (1<<22) #define HDSP_Midi1InterruptEnable (1<<23) @@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_TimecodeSync (1<<27) #define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ #define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */ -#define HDSP_midi0IRQPending (1<<30) +#define HDSP_midi0IRQPending (1<<30) #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) @@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the Multiface still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -460,7 +460,7 @@ struct hdsp { unsigned char qs_in_channels; /* quad speed mode for H9632 */ unsigned char ds_in_channels; unsigned char ss_in_channels; /* different for multiface/digiface */ - unsigned char qs_out_channels; + unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; @@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = { static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ /* Analog */ - 0, 1, 2, 3, 4, 5, 6, 7, + 0, 1, 2, 3, 4, 5, 6, 7, /* ADAT 2 */ - 16, 17, 18, 19, 20, 21, 22, 23, + 16, 17, 18, 19, 20, 21, 22, 23, /* SPDIF */ 24, 25, -1, -1, -1, -1, -1, -1, -1, -1 @@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ - -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; @@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ @@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d static struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, - .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, + .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, .subvendor = PCI_ANY_ID, .subdevice = PCI_ANY_ID, }, /* RME Hammerfall-DSP */ @@ -653,7 +653,6 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { - if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); @@ -661,7 +660,29 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp) return -EIO; } return 0; +} +static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, + unsigned int delay) +{ + unsigned int i; + + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) + return 0; + + for (i = 0; i != loops; ++i) { + if (hdsp_read(hdsp, HDSP_statusRegister) & HDSP_ConfigError) + msleep(delay); + else { + snd_printd("Hammerfall-DSP: iobox found after %ums!\n", + i * delay); + return 0; + } + } + + snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { @@ -670,19 +691,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { unsigned long flags; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + snd_printk ("Hammerfall-DSP: loading firmware\n"); hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); return -EIO; } - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - + for (i = 0; i < 24413; ++i) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { @@ -692,7 +713,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { } ssleep(3); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); return -EIO; @@ -705,15 +726,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { #endif hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); - + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); spin_lock_irqsave(&hdsp->lock, flags); snd_hdsp_set_defaults(hdsp); - spin_unlock_irqrestore(&hdsp->lock, flags); + spin_unlock_irqrestore(&hdsp->lock, flags); } - + hdsp->state |= HDSP_FirmwareLoaded; return 0; @@ -722,7 +743,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) @@ -738,7 +759,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); } else { hdsp->io_type = Digiface; - } + } } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) @@ -786,13 +807,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout) -{ +{ int i; /* the fifoStatus registers reports on how many words are available in the command FIFO. */ - + for (i = 0; i < timeout; i++) { if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count) @@ -824,11 +845,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short if (addr >= HDSP_MATRIX_MIXER_SIZE) return -1; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) { /* from martin bjornsen: - + "You can only write dwords to the mixer memory which contain two mixer values in the low and high @@ -847,7 +868,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short hdsp->mixer_matrix[addr] = data; - + /* `addr' addresses a 16-bit wide address, but the address space accessed via hdsp_write uses byte offsets. put another way, addr @@ -856,17 +877,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short to access 0 to 2703 ... */ ad = addr/2; - - hdsp_write (hdsp, 4096 + (ad*4), - (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + + + hdsp_write (hdsp, 4096 + (ad*4), + (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + hdsp->mixer_matrix[addr&0x7fe]); - + return 0; } else { ad = (addr << 16) + data; - + if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT)) return -1; @@ -902,7 +923,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) if (status & HDSP_SPDIFErrorFlag) return 0; - + switch (rate_bits) { case HDSP_spdifFrequency32KHz: return 32000; case HDSP_spdifFrequency44_1KHz: return 44100; @@ -910,13 +931,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) case HDSP_spdifFrequency64KHz: return 64000; case HDSP_spdifFrequency88_2KHz: return 88200; case HDSP_spdifFrequency96KHz: return 96000; - case HDSP_spdifFrequency128KHz: + case HDSP_spdifFrequency128KHz: if (hdsp->io_type == H9632) return 128000; break; - case HDSP_spdifFrequency176_4KHz: + case HDSP_spdifFrequency176_4KHz: if (hdsp->io_type == H9632) return 176400; break; - case HDSP_spdifFrequency192KHz: + case HDSP_spdifFrequency192KHz: if (hdsp->io_type == H9632) return 192000; break; default: @@ -1027,7 +1048,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -1053,35 +1074,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) there is no need for it (e.g. during module initialization). */ - - if (!(hdsp->control_register & HDSP_ClockModeMaster)) { + + if (!(hdsp->control_register & HDSP_ClockModeMaster)) { if (called_internally) { /* request from ctl or card initialization */ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n"); return -1; - } else { + } else { /* hw_param request while in AutoSync mode */ int external_freq = hdsp_external_sample_rate(hdsp); int spdif_freq = hdsp_spdif_sample_rate(hdsp); - + if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n"); else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) - snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); else if (rate != external_freq) { snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n"); return -1; - } - } + } + } } current_rate = hdsp->system_sample_rate; /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -1089,7 +1110,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) if (rate > 96000 && hdsp->io_type != H9632) return -EINVAL; - + switch (rate) { case 32000: if (current_rate > 48000) @@ -1179,7 +1200,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) break; } } - + hdsp->system_sample_rate = rate; return 0; @@ -1245,16 +1266,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output) { if (!snd_rawmidi_transmit_empty (hmidi->output)) { if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]); } } @@ -1325,14 +1346,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data) { struct hdsp_midi *hmidi = (struct hdsp_midi *) data; unsigned long flags; - + snd_hdsp_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1501,7 +1522,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif); return 0; } @@ -1511,7 +1532,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif; @@ -1530,7 +1551,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream); return 0; } @@ -1540,7 +1561,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif_stream; @@ -1602,7 +1623,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp); return 0; } @@ -1612,7 +1633,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3); @@ -1649,7 +1670,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp); return 0; } @@ -1659,7 +1680,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1693,7 +1714,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp); return 0; } @@ -1703,7 +1724,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1737,7 +1758,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp); return 0; } @@ -1747,7 +1768,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1781,7 +1802,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp); return 0; } @@ -1791,7 +1812,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1828,7 +1849,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_spdif_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1858,7 +1879,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct ucontrol->value.enumerated.item[0] = 9; break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1882,7 +1903,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate; return 0; } @@ -1899,7 +1920,7 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; @@ -1912,7 +1933,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_external_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1940,9 +1961,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru break; case 192000: ucontrol->value.enumerated.item[0] = 9; - break; + break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1968,7 +1989,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp) static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Master", "Slave" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 2; @@ -1981,7 +2002,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp); return 0; } @@ -2018,7 +2039,7 @@ static int hdsp_clock_source(struct hdsp *hdsp) case 192000: return 9; default: - return 3; + return 3; } } else { return 0; @@ -2032,7 +2053,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) case HDSP_CLOCK_SOURCE_AUTOSYNC: if (hdsp_external_sample_rate(hdsp) != 0) { if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) { - hdsp->control_register &= ~HDSP_ClockModeMaster; + hdsp->control_register &= ~HDSP_ClockModeMaster; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); return 0; } @@ -2043,7 +2064,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) break; case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ: rate = 44100; - break; + break; case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ: rate = 48000; break; @@ -2078,13 +2099,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ { static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; if (hdsp->io_type == H9632) uinfo->value.enumerated.items = 10; else - uinfo->value.enumerated.items = 7; + uinfo->value.enumerated.items = 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2094,7 +2115,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp); return 0; } @@ -2104,7 +2125,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2130,7 +2151,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; return 0; } @@ -2165,7 +2186,7 @@ static int hdsp_da_gain(struct hdsp *hdsp) case HDSP_DAGainMinus10dBV: return 2; default: - return 1; + return 1; } } @@ -2180,8 +2201,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_DAGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_DAGainMinus10dBV; - break; + hdsp->control_register |= HDSP_DAGainMinus10dBV; + break; default: return -1; @@ -2193,7 +2214,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2206,7 +2227,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp); return 0; } @@ -2216,7 +2237,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2250,7 +2271,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp) case HDSP_ADGainLowGain: return 2; default: - return 1; + return 1; } } @@ -2262,11 +2283,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_ADGainMinus10dBV; break; case 1: - hdsp->control_register |= HDSP_ADGainPlus4dBu; + hdsp->control_register |= HDSP_ADGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_ADGainLowGain; - break; + hdsp->control_register |= HDSP_ADGainLowGain; + break; default: return -1; @@ -2278,7 +2299,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2291,7 +2312,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp); return 0; } @@ -2301,7 +2322,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2335,7 +2356,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp) case HDSP_PhoneGainMinus12dB: return 2; default: - return 0; + return 0; } } @@ -2347,11 +2368,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_PhoneGain0dB; break; case 1: - hdsp->control_register |= HDSP_PhoneGainMinus6dB; + hdsp->control_register |= HDSP_PhoneGainMinus6dB; break; case 2: - hdsp->control_register |= HDSP_PhoneGainMinus12dB; - break; + hdsp->control_register |= HDSP_PhoneGainMinus12dB; + break; default: return -1; @@ -2363,7 +2384,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2376,7 +2397,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp); return 0; } @@ -2386,7 +2407,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2432,7 +2453,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode) static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp); return 0; } @@ -2442,7 +2463,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2488,7 +2509,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode) static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp); return 0; } @@ -2498,7 +2519,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2576,7 +2597,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd { static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2595,7 +2616,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd uinfo->value.enumerated.items = 0; break; } - + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2605,7 +2626,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp); return 0; } @@ -2615,7 +2636,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; @@ -2664,7 +2685,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) case HDSP_SelSyncRef_SPDIF: return HDSP_AUTOSYNC_FROM_SPDIF; case HDSP_SelSyncRefMask: - return HDSP_AUTOSYNC_FROM_NONE; + return HDSP_AUTOSYNC_FROM_NONE; case HDSP_SelSyncRef_ADAT1: return HDSP_AUTOSYNC_FROM_ADAT1; case HDSP_SelSyncRef_ADAT2: @@ -2680,7 +2701,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 7; @@ -2693,7 +2714,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp); return 0; } @@ -2727,7 +2748,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp_line_out(hdsp); spin_unlock_irq(&hdsp->lock); @@ -2739,7 +2760,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2773,7 +2794,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise) static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->precise_ptr; spin_unlock_irq(&hdsp->lock); @@ -2785,7 +2806,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2819,7 +2840,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet) static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; spin_unlock_irq(&hdsp->lock); @@ -2831,7 +2852,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2873,12 +2894,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem source = ucontrol->value.integer.value[0]; destination = ucontrol->value.integer.value[1]; - + if (source >= hdsp->max_channels) addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination); else addr = hdsp_input_to_output_key(hdsp,source, destination); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr); spin_unlock_irq(&hdsp->lock); @@ -2926,7 +2947,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; + static char *texts[] = {"No Lock", "Lock", "Sync" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2971,7 +2992,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp) int status = hdsp_read(hdsp, HDSP_statusRegister); if (status & HDSP_SPDIFErrorFlag) return 0; - else { + else { if (status & HDSP_SPDIFSync) return 2; else @@ -3007,7 +3028,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp) return 1; } else return 0; -} +} static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3025,17 +3046,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc } static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx) -{ +{ int status = hdsp_read(hdsp, HDSP_statusRegister); - + if (status & (HDSP_Lock0>>idx)) { if (status & (HDSP_Sync0>>idx)) return 2; else - return 1; + return 1; } else return 0; -} +} static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3053,7 +3074,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn break; case Multiface: case H9632: - if (offset >= 1) + if (offset >= 1) return -EINVAL; break; default: @@ -3115,7 +3136,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); return 0; } @@ -3125,7 +3146,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -3170,7 +3191,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = { .get = snd_hdsp_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -3188,7 +3209,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0), HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0), HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), -/* 'Sample Clock Source' complies with the alsa control naming scheme */ +/* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3240,7 +3261,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) return err; } } - + /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */ if (hdsp->io_type == H9632) { for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) { @@ -3259,7 +3280,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3298,7 +3319,7 @@ 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); @@ -3362,17 +3383,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ: clock_source = "Internal 192 kHz"; - break; + break; default: - clock_source = "Error"; + clock_source = "Error"; } snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source); - + if (hdsp_system_clock_mode(hdsp)) system_clock_mode = "Slave"; else system_clock_mode = "Master"; - + switch (hdsp_pref_sync_ref (hdsp)) { case HDSP_SYNC_FROM_WORD: pref_sync_ref = "Word Clock"; @@ -3397,7 +3418,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref); - + switch (hdsp_autosync_ref (hdsp)) { case HDSP_AUTOSYNC_FROM_WORD: autosync_ref = "Word Clock"; @@ -3410,7 +3431,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_AUTOSYNC_FROM_NONE: autosync_ref = "None"; - break; + break; case HDSP_AUTOSYNC_FROM_ADAT1: autosync_ref = "ADAT1"; break; @@ -3425,14 +3446,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref); - + snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp)); - + snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); - + snd_iprintf(buffer, "\n"); switch (hdsp_spdif_in(hdsp)) { @@ -3452,7 +3473,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "IEC958 input: ???\n"); break; } - + if (hdsp->control_register & HDSP_SPDIFOpticalOut) snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); else @@ -3510,13 +3531,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf (buffer, "SPDIF: No Lock\n"); else snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock"); - + x = status2 & HDSP_wc_sync; if (status2 & HDSP_wc_lock) snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock"); else snd_iprintf (buffer, "Word Clock: No Lock\n"); - + x = status & HDSP_TimecodeSync; if (status & HDSP_TimecodeLock) snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock"); @@ -3524,11 +3545,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "ADAT Sync: No Lock\n"); snd_iprintf(buffer, "\n"); - + /* Informations about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; - + switch (hdsp_ad_gain(hdsp)) { case 0: tmp = "-10 dBV"; @@ -3554,7 +3575,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf(buffer, "DA Gain : %s\n", tmp); - + switch (hdsp_phone_gain(hdsp)) { case 0: tmp = "0 dB"; @@ -3568,8 +3589,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } snd_iprintf(buffer, "Phones Gain : %s\n", tmp); - snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); - + snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); + if (hdsp->control_register & HDSP_AnalogExtensionBoard) snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n"); else @@ -3632,18 +3653,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax Master clock mode maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - Enable line out. + Enable line out. */ - hdsp->control_register = HDSP_ClockModeMaster | - HDSP_SPDIFInputCoaxial | - hdsp_encode_latency(7) | + hdsp->control_register = HDSP_ClockModeMaster | + HDSP_SPDIFInputCoaxial | + hdsp_encode_latency(7) | HDSP_LineOut; - + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3661,7 +3682,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) hdsp_compute_period_size(hdsp); /* silence everything */ - + for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i) hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN; @@ -3669,7 +3690,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN)) return -EIO; } - + /* H9632 specific defaults */ if (hdsp->io_type == H9632) { hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB); @@ -3687,12 +3708,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) static void hdsp_midi_tasklet(unsigned long arg) { struct hdsp *hdsp = (struct hdsp *)arg; - + if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); if (hdsp->midi[1].pending) snd_hdsp_midi_input_read (&hdsp->midi[1]); -} +} static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) { @@ -3704,7 +3725,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) unsigned int midi0status; unsigned int midi1status; int schedule = 0; - + status = hdsp_read(hdsp, HDSP_statusRegister); audio = status & HDSP_audioIRQPending; @@ -3718,15 +3739,18 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; - + + if (!(hdsp->state & HDSP_InitializationComplete)) + return IRQ_HANDLED; + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - + if (hdsp->playback_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); } - + if (midi0 && midi0status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ @@ -3769,10 +3793,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp, if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels)) return NULL; - + if ((mapped_channel = hdsp->channel_map[channel]) < 0) return NULL; - + if (stream == SNDRV_PCM_STREAM_CAPTURE) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else @@ -3965,7 +3989,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; int running; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; @@ -4059,10 +4083,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4088,10 +4112,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4170,7 +4194,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_in_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_in_channels, @@ -4201,7 +4225,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_out_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_out_channels, @@ -4318,8 +4342,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; - } - + } + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -4413,13 +4437,6 @@ static int snd_hdsp_capture_release(struct snd_pcm_substream *substream) return 0; } -static int snd_hdsp_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file) -{ - /* we have nothing to initialize but the call is required */ - return 0; -} - - /* helper functions for copying meter values */ static inline int copy_u32_le(void __user *dest, void __iomem *src) { @@ -4536,7 +4553,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_playbackRmsLevel + i * 8)) return -EFAULT; - if (copy_u64_le(&peak_rms->input_rms[i], + if (copy_u64_le(&peak_rms->input_rms[i], hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_inputRmsLevel + i * 8)) return -EFAULT; @@ -4546,7 +4563,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; int err; @@ -4580,7 +4597,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_config_info info; unsigned long flags; int i; - + err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; @@ -4614,7 +4631,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp); info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); - + } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4625,7 +4642,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne } case SNDRV_HDSP_IOCTL_GET_9632_AEB: { struct hdsp_9632_aeb h9632_aeb; - + if (hdsp->io_type != H9632) return -EINVAL; h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS; h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS; @@ -4636,7 +4653,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne case SNDRV_HDSP_IOCTL_GET_VERSION: { struct hdsp_version hdsp_version; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; if (hdsp->io_type == Undefined) { if ((err = hdsp_get_iobox_version(hdsp)) < 0) @@ -4652,7 +4669,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_firmware __user *firmware; u32 __user *firmware_data; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */ if (hdsp->io_type == Undefined) return -EINVAL; @@ -4665,25 +4682,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (get_user(firmware_data, &firmware->firmware_data)) return -EFAULT; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0) return -EFAULT; - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; - - snd_hdsp_initialize_channels(hdsp); + + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - + if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n"); return err; @@ -4730,18 +4747,16 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) { struct snd_hwdep *hw; int err; - + if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0) return err; - + hdsp->hwdep = hw; hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); - hw->ops.open = snd_hdsp_hwdep_dummy_op; hw->ops.ioctl = snd_hdsp_hwdep_ioctl; - hw->ops.release = snd_hdsp_hwdep_dummy_op; - + return 0; } @@ -4774,24 +4789,24 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp) static int snd_hdsp_enable_io (struct hdsp *hdsp) { int i; - + if (hdsp_fifo_wait (hdsp, 0, 100)) { snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n"); return -EIO; } - + for (i = 0; i < hdsp->max_channels; ++i) { hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1); hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1); } - + return 0; } static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { int status, aebi_channels, aebo_channels; - + switch (hdsp->io_type) { case Digiface: hdsp->card_name = "RME Hammerfall DSP + Digiface"; @@ -4804,7 +4819,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS; break; - + case H9632: status = hdsp_read(hdsp, HDSP_statusRegister); /* HDSP_AEBx bits are low when AEB are connected */ @@ -4824,7 +4839,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; - + default: /* should never get here */ break; @@ -4840,12 +4855,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp) static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp) { int err; - + if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n"); return err; } - + if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n"); @@ -4876,19 +4891,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n"); return err; } - + if (!(hdsp->state & HDSP_InitializationComplete)) { strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - + if ((err = snd_card_register(card)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n"); return err; } hdsp->state |= HDSP_InitializationComplete; } - + return 0; } @@ -4899,7 +4914,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) const char *fwfile; const struct firmware *fw; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp->io_type == Undefined) { @@ -4908,7 +4923,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; } - + /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { case Multiface: @@ -4942,12 +4957,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache)); release_firmware(fw); - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; @@ -4994,14 +5009,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card, hdsp->max_channels = 26; hdsp->card = card; - + spin_lock_init(&hdsp->lock); tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); - + pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; - + /* From Martin Bjoernsen : "It is important that the card's latency timer register in the PCI configuration space is set to a value much larger @@ -5010,7 +5025,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, to its maximum 255 to avoid problems with some computers." */ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF); - + strcpy(card->driver, "H-DSP"); strcpy(card->mixername, "Xilinx FPGA"); @@ -5024,7 +5039,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, } else { hdsp->card_name = "RME HDSP 9632"; hdsp->max_channels = 16; - is_9632 = 1; + is_9632 = 1; } if ((err = pci_enable_device(pci)) < 0) @@ -5053,12 +5068,12 @@ static int __devinit snd_hdsp_create(struct snd_card *card, if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) return err; - + if (!is_9652 && !is_9632) { - /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ - ssleep(2); + /* we wait a maximum of 10 seconds to let freshly + * inserted cardbus cards do their hardware init */ + err = hdsp_wait_for_iobox(hdsp, 1000, 10); - err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; @@ -5080,35 +5095,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; return 0; } else { - snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; - else + else hdsp->io_type = Digiface; } } - + if ((err = snd_hdsp_enable_io(hdsp)) != 0) return err; - + if (is_9652) hdsp->io_type = H9652; - + if (is_9632) hdsp->io_type = H9632; if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; - + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - hdsp->state |= HDSP_FirmwareLoaded; + hdsp->state |= HDSP_FirmwareLoaded; if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0) return err; - return 0; + return 0; } static int snd_hdsp_free(struct hdsp *hdsp) @@ -5124,13 +5139,13 @@ static int snd_hdsp_free(struct hdsp *hdsp) free_irq(hdsp->irq, (void *)hdsp); snd_hdsp_free_buffers(hdsp); - + if (hdsp->iobase) iounmap(hdsp->iobase); if (hdsp->port) pci_release_regions(hdsp->pci); - + pci_disable_device(hdsp->pci); return 0; } @@ -5175,7 +5190,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, } strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); if ((err = snd_card_register(card)) < 0) { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d4b4e0d0fee8..bac2dc0c5d85 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4100,13 +4100,6 @@ static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) return 0; } -static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file) -{ - /* we have nothing to initialize but the call is required */ - return 0; -} - - static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg) { @@ -4213,9 +4206,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->private_data = hdspm; strcpy(hw->name, "HDSPM hwdep interface"); - hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; - hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; } diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index c5601b0ad7cc..d989215f3556 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -273,7 +273,8 @@ static inline void snd_sonicvibes_setdmaa(struct sonicvibes * sonic, outl(count, sonic->dmaa_port + SV_DMA_COUNT0); outb(0x18, sonic->dmaa_port + SV_DMA_MODE); #if 0 - printk("program dmaa: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmaa_port + SV_DMA_ADDR0)); + printk(KERN_DEBUG "program dmaa: addr = 0x%x, paddr = 0x%x\n", + addr, inl(sonic->dmaa_port + SV_DMA_ADDR0)); #endif } @@ -288,7 +289,8 @@ static inline void snd_sonicvibes_setdmac(struct sonicvibes * sonic, outl(count, sonic->dmac_port + SV_DMA_COUNT0); outb(0x14, sonic->dmac_port + SV_DMA_MODE); #if 0 - printk("program dmac: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmac_port + SV_DMA_ADDR0)); + printk(KERN_DEBUG "program dmac: addr = 0x%x, paddr = 0x%x\n", + addr, inl(sonic->dmac_port + SV_DMA_ADDR0)); #endif } @@ -355,71 +357,104 @@ static unsigned char snd_sonicvibes_in(struct sonicvibes * sonic, unsigned char #if 0 static void snd_sonicvibes_debug(struct sonicvibes * sonic) { - printk("SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX))); + printk(KERN_DEBUG + "SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX))); printk(" STATUS = 0x%02x\n", inb(SV_REG(sonic, STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00)); + printk(KERN_DEBUG + " 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00)); printk(" 0x20: synth rate low = 0x%02x\n", snd_sonicvibes_in(sonic, 0x20)); - printk(" 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01)); + printk(KERN_DEBUG + " 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01)); printk(" 0x21: synth rate high = 0x%02x\n", snd_sonicvibes_in(sonic, 0x21)); - printk(" 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02)); + printk(KERN_DEBUG + " 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02)); printk(" 0x22: ADC clock = 0x%02x\n", snd_sonicvibes_in(sonic, 0x22)); - printk(" 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03)); + printk(KERN_DEBUG + " 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03)); printk(" 0x23: ADC alt rate = 0x%02x\n", snd_sonicvibes_in(sonic, 0x23)); - printk(" 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04)); + printk(KERN_DEBUG + " 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04)); printk(" 0x24: ADC pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x24)); - printk(" 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05)); + printk(KERN_DEBUG + " 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05)); printk(" 0x25: ADC pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x25)); - printk(" 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06)); + printk(KERN_DEBUG + " 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06)); printk(" 0x26: Synth pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x26)); - printk(" 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07)); + printk(KERN_DEBUG + " 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07)); printk(" 0x27: Synth pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x27)); - printk(" 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08)); + printk(KERN_DEBUG + " 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08)); printk(" 0x28: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x28)); - printk(" 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09)); + printk(KERN_DEBUG + " 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09)); printk(" 0x29: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x29)); - printk(" 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a)); + printk(KERN_DEBUG + " 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a)); printk(" 0x2a: MPU401 = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2a)); - printk(" 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b)); + printk(KERN_DEBUG + " 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b)); printk(" 0x2b: drive ctrl = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2b)); - printk(" 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c)); + printk(KERN_DEBUG + " 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c)); printk(" 0x2c: SRS space = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2c)); - printk(" 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d)); + printk(KERN_DEBUG + " 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d)); printk(" 0x2d: SRS center = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2d)); - printk(" 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e)); + printk(KERN_DEBUG + " 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e)); printk(" 0x2e: wave source = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2e)); - printk(" 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f)); + printk(KERN_DEBUG + " 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f)); printk(" 0x2f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2f)); - printk(" 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10)); + printk(KERN_DEBUG + " 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10)); printk(" 0x30: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x30)); - printk(" 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11)); + printk(KERN_DEBUG + " 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11)); printk(" 0x31: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x31)); - printk(" 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12)); + printk(KERN_DEBUG + " 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12)); printk(" 0x32: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x32)); - printk(" 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13)); + printk(KERN_DEBUG + " 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13)); printk(" 0x33: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x33)); - printk(" 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14)); + printk(KERN_DEBUG + " 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14)); printk(" 0x34: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x34)); - printk(" 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15)); + printk(KERN_DEBUG + " 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15)); printk(" 0x35: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x35)); - printk(" 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16)); + printk(KERN_DEBUG + " 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16)); printk(" 0x36: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x36)); - printk(" 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17)); + printk(KERN_DEBUG + " 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17)); printk(" 0x37: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x37)); - printk(" 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18)); + printk(KERN_DEBUG + " 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18)); printk(" 0x38: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x38)); - printk(" 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19)); + printk(KERN_DEBUG + " 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19)); printk(" 0x39: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x39)); - printk(" 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a)); + printk(KERN_DEBUG + " 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a)); printk(" 0x3a: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3a)); - printk(" 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b)); + printk(KERN_DEBUG + " 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b)); printk(" 0x3b: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3b)); - printk(" 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c)); + printk(KERN_DEBUG + " 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c)); printk(" 0x3c: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3c)); - printk(" 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d)); + printk(KERN_DEBUG + " 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d)); printk(" 0x3d: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3d)); - printk(" 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e)); + printk(KERN_DEBUG + " 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e)); printk(" 0x3e: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3e)); - printk(" 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f)); + printk(KERN_DEBUG + " 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f)); printk(" 0x3f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3f)); } @@ -476,8 +511,8 @@ static void snd_sonicvibes_pll(unsigned int rate, *res_m = m; *res_n = n; #if 0 - printk("metric = %i, xm = %i, xn = %i\n", metric, xm, xn); - printk("pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n); + printk(KERN_DEBUG "metric = %i, xm = %i, xn = %i\n", metric, xm, xn); + printk(KERN_DEBUG "pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n); #endif } diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index c612b435ca2b..a9da9c184660 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -68,40 +68,40 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) { unsigned int val, tmp; - printk("Trident voice %i:\n", voice); + printk(KERN_DEBUG "Trident voice %i:\n", voice); outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR)); val = inl(TRID_REG(trident, CH_LBA)); - printk("LBA: 0x%x\n", val); + printk(KERN_DEBUG "LBA: 0x%x\n", val); val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); - printk("GVSel: %i\n", val >> 31); - printk("Pan: 0x%x\n", (val >> 24) & 0x7f); - printk("Vol: 0x%x\n", (val >> 16) & 0xff); - printk("CTRL: 0x%x\n", (val >> 12) & 0x0f); - printk("EC: 0x%x\n", val & 0x0fff); + printk(KERN_DEBUG "GVSel: %i\n", val >> 31); + printk(KERN_DEBUG "Pan: 0x%x\n", (val >> 24) & 0x7f); + printk(KERN_DEBUG "Vol: 0x%x\n", (val >> 16) & 0xff); + printk(KERN_DEBUG "CTRL: 0x%x\n", (val >> 12) & 0x0f); + printk(KERN_DEBUG "EC: 0x%x\n", val & 0x0fff); if (trident->device != TRIDENT_DEVICE_ID_NX) { val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS)); - printk("CSO: 0x%x\n", val >> 16); + printk(KERN_DEBUG "CSO: 0x%x\n", val >> 16); printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff); - printk("FMS: 0x%x\n", val & 0x0f); + printk(KERN_DEBUG "FMS: 0x%x\n", val & 0x0f); val = inl(TRID_REG(trident, CH_DX_ESO_DELTA)); - printk("ESO: 0x%x\n", val >> 16); - printk("Delta: 0x%x\n", val & 0xffff); + printk(KERN_DEBUG "ESO: 0x%x\n", val >> 16); + printk(KERN_DEBUG "Delta: 0x%x\n", val & 0xffff); val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); } else { // TRIDENT_DEVICE_ID_NX val = inl(TRID_REG(trident, CH_NX_DELTA_CSO)); tmp = (val >> 24) & 0xff; - printk("CSO: 0x%x\n", val & 0x00ffffff); + printk(KERN_DEBUG "CSO: 0x%x\n", val & 0x00ffffff); val = inl(TRID_REG(trident, CH_NX_DELTA_ESO)); tmp |= (val >> 16) & 0xff00; - printk("Delta: 0x%x\n", tmp); - printk("ESO: 0x%x\n", val & 0x00ffffff); + printk(KERN_DEBUG "Delta: 0x%x\n", tmp); + printk(KERN_DEBUG "ESO: 0x%x\n", val & 0x00ffffff); val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL)); - printk("Alpha: 0x%x\n", val >> 20); - printk("FMS: 0x%x\n", (val >> 16) & 0x0f); + printk(KERN_DEBUG "Alpha: 0x%x\n", val >> 20); + printk(KERN_DEBUG "FMS: 0x%x\n", (val >> 16) & 0x0f); } - printk("FMC: 0x%x\n", (val >> 14) & 3); - printk("RVol: 0x%x\n", (val >> 7) & 0x7f); - printk("CVol: 0x%x\n", val & 0x7f); + printk(KERN_DEBUG "FMC: 0x%x\n", (val >> 14) & 3); + printk(KERN_DEBUG "RVol: 0x%x\n", (val >> 7) & 0x7f); + printk(KERN_DEBUG "CVol: 0x%x\n", val & 0x7f); } #endif @@ -496,12 +496,17 @@ void snd_trident_write_voice_regs(struct snd_trident * trident, outl(regs[4], TRID_REG(trident, CH_START + 16)); #if 0 - printk("written %i channel:\n", voice->number); - printk(" regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0))); - printk(" regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4))); - printk(" regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8))); - printk(" regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12))); - printk(" regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16))); + printk(KERN_DEBUG "written %i channel:\n", voice->number); + printk(KERN_DEBUG " regs[0] = 0x%x/0x%x\n", + regs[0], inl(TRID_REG(trident, CH_START + 0))); + printk(KERN_DEBUG " regs[1] = 0x%x/0x%x\n", + regs[1], inl(TRID_REG(trident, CH_START + 4))); + printk(KERN_DEBUG " regs[2] = 0x%x/0x%x\n", + regs[2], inl(TRID_REG(trident, CH_START + 8))); + printk(KERN_DEBUG " regs[3] = 0x%x/0x%x\n", + regs[3], inl(TRID_REG(trident, CH_START + 12))); + printk(KERN_DEBUG " regs[4] = 0x%x/0x%x\n", + regs[4], inl(TRID_REG(trident, CH_START + 16))); #endif } @@ -583,7 +588,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident, outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2)); break; case TRIDENT_DEVICE_ID_SI7018: - // printk("voice->Vol = 0x%x\n", voice->Vol); + /* printk(KERN_DEBUG "voice->Vol = 0x%x\n", voice->Vol); */ outw((voice->CTRL << 12) | voice->Vol, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); break; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index d8705547dae1..809b233dd4a3 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -466,7 +466,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* + printk(KERN_DEBUG "via: tbl %d: at %d size %d " + "(rest %d)\n", idx, ofs, r, rest); + */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; @@ -2360,14 +2363,14 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K), SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x1019, 0x0a85, "ECS L7VMM2", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1019, 0, "ESC K8", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1019, "ESC K8", VIA_DXS_SRC), SND_PCI_QUIRK(0x1019, 0xaa01, "ESC K8T890-A", VIA_DXS_SRC), SND_PCI_QUIRK(0x1025, 0x0033, "Acer Inspire 1353LM", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x1025, 0x0046, "Acer Aspire 1524 WLMi", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1043, 0, "ASUS A7/A8", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1071, 0, "Diverse Notebook", VIA_DXS_NO_VRA), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS A7/A8", VIA_DXS_NO_VRA), + SND_PCI_QUIRK_VENDOR(0x1071, "Diverse Notebook", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1106, "ASRock", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC), @@ -2375,7 +2378,7 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7142, "MSI K8MM-V", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1462, 0, "MSI Mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1462, "MSI Mobo", VIA_DXS_SRC), SND_PCI_QUIRK(0x147b, 0x1401, "ABIT KD7(-RAID)", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x147b, 0x1411, "ABIT VA-20", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x147b, 0x1413, "ABIT KV8 Pro", VIA_DXS_ENABLE), @@ -2389,11 +2392,11 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x161f, 0x2032, "m680x machines", VIA_DXS_48K), SND_PCI_QUIRK(0x1631, 0xe004, "PB EasyNote 3174", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1695, 0x3005, "EPoX EP-8K9A", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1695, 0, "EPoX mobo", VIA_DXS_SRC), - SND_PCI_QUIRK(0x16f3, 0, "Jetway K8", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1734, 0, "FSC Laptop", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1695, "EPoX mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x16f3, "Jetway K8", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1734, "FSC Laptop", VIA_DXS_SRC), SND_PCI_QUIRK(0x1849, 0x3059, "ASRock K7VM2", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1849, 0, "ASRock mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1849, "ASRock mobo", VIA_DXS_SRC), SND_PCI_QUIRK(0x1919, 0x200a, "Soltek SL-K8", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x4005, 0x4710, "MSI K7T266", VIA_DXS_SRC), { } /* terminator */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index c086b762c150..0d54e3503c1e 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -328,7 +328,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* + printk(KERN_DEBUG "via: tbl %d: at %d size %d " + "(rest %d)\n", idx, ofs, r, rest); + */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7e87f398ff0b..c0efe4491116 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -107,7 +107,9 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset) static void vx2_outb(struct vx_core *chip, int offset, unsigned char val) { outb(val, vx2_reg_addr(chip, offset)); - //printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* + printk(KERN_DEBUG "outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + */ } /** @@ -126,7 +128,9 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset) */ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) { - // printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* + printk(KERN_DEBUG "outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + */ outl(val, vx2_reg_addr(chip, offset)); } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 90d0d62bd0b4..2f0925236a1b 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -318,7 +318,12 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ ypcm->period_pos += delta; ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* + printk(KERN_DEBUG + "done - active_bank = 0x%x, start = 0x%x\n", + chip->active_bank, + voice->bank[chip->active_bank].start); + */ ypcm->period_pos %= ypcm->period_size; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(ypcm->substream); @@ -366,7 +371,12 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { ypcm->period_pos %= ypcm->period_size; - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* + printk(KERN_DEBUG + "done - active_bank = 0x%x, start = 0x%x\n", + chip->active_bank, + voice->bank[chip->active_bank].start); + */ spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(substream); spin_lock(&chip->reg_lock); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index dfa40b0ed86d..5d2afa0b0ce4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -82,14 +82,21 @@ static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned c #if 0 void pdacf_dump(struct snd_pdacf *chip) { - printk("PDAUDIOCF DUMP (0x%lx):\n", chip->port); - printk("WPD : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_WDP)); - printk("RDP : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_RDP)); - printk("TCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_TCR)); - printk("SCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_SCR)); - printk("ISR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_ISR)); - printk("IER : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_IER)); - printk("AK_IFR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_AK_IFR)); + printk(KERN_DEBUG "PDAUDIOCF DUMP (0x%lx):\n", chip->port); + printk(KERN_DEBUG "WPD : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_WDP)); + printk(KERN_DEBUG "RDP : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_RDP)); + printk(KERN_DEBUG "TCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_TCR)); + printk(KERN_DEBUG "SCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_SCR)); + printk(KERN_DEBUG "ISR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_ISR)); + printk(KERN_DEBUG "IER : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_IER)); + printk(KERN_DEBUG "AK_IFR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_AK_IFR)); } #endif diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c index ea903c8e90dd..dcd32201bc8c 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c @@ -269,7 +269,7 @@ void pdacf_tasklet(unsigned long private_data) rdp = inw(chip->port + PDAUDIOCF_REG_RDP); wdp = inw(chip->port + PDAUDIOCF_REG_WDP); - // printk("TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); + /* printk(KERN_DEBUG "TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); */ size = wdp - rdp; if (size < 0) size += 0x10000; @@ -321,5 +321,5 @@ void pdacf_tasklet(unsigned long private_data) spin_lock(&chip->reg_lock); } spin_unlock(&chip->reg_lock); - // printk("TASKLET: end\n"); + /* printk(KERN_DEBUG "TASKLET: end\n"); */ } diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index ef2c3f417175..f361c26506aa 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -477,7 +477,7 @@ static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream) card->dma_start_bus_addr[SND_PS3_CH_R] = runtime->dma_addr + (runtime->dma_bytes / 2); - pr_debug("%s: vaddr=%p bus=%#lx\n", __func__, + pr_debug("%s: vaddr=%p bus=%#llx\n", __func__, card->dma_start_vaddr[SND_PS3_CH_L], card->dma_start_bus_addr[SND_PS3_CH_L]); @@ -1028,7 +1028,7 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) pr_info("%s: nullbuffer alloc failed\n", __func__); goto clean_preallocate; } - pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__, + pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, the_card.null_buffer_start_vaddr, the_card.null_buffer_start_dma_addr); /* set default sample rate/word width */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index ef025c66cc66..3d2bb6fc6dcc 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -6,6 +6,7 @@ menuconfig SND_SOC tristate "ALSA for SoC audio support" select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS + select SND_JACK if INPUT=y || INPUT=SND ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 86a9b1f5b0f3..0237879fd412 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 1fac5efd285b..9ef6b96373f5 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -44,8 +44,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> -#include <mach/hardware.h> - #include "atmel-pcm.h" @@ -349,7 +347,7 @@ static int atmel_pcm_mmap(struct snd_pcm_substream *substream, vma->vm_end - vma->vm_start, vma->vm_page_prot); } -struct snd_pcm_ops atmel_pcm_ops = { +static struct snd_pcm_ops atmel_pcm_ops = { .open = atmel_pcm_open, .close = atmel_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index c5d67900d666..e588e63f18d2 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino@endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops atmel_ssc_dai_ops = { + .startup = atmel_ssc_startup, + .shutdown = atmel_ssc_shutdown, + .prepare = atmel_ssc_prepare, + .hw_params = atmel_ssc_hw_params, + .set_fmt = atmel_ssc_set_dai_fmt, + .set_clkdiv = atmel_ssc_set_dai_clkdiv, +}; + struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { { .name = "atmel-ssc0", .id = 0, @@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[0], }, #if NUM_SSC_DEVICES == 3 @@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[1], }, { .name = "atmel-ssc2", @@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .channels_max = 2, .rates = ATMEL_SSC_RATES, .formats = ATMEL_SSC_FORMATS,}, - .ops = { - .startup = atmel_ssc_startup, - .shutdown = atmel_ssc_shutdown, - .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params, - .set_fmt = atmel_ssc_set_dai_fmt, - .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, + .ops = &atmel_ssc_dai_ops, .private_data = &ssc_info[2], }, #endif diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index a828746e8a2f..391135f9c6c1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -10,7 +10,7 @@ * Based on at91-ssc.c by * Frank Mandarino <fmandarino@endrelia.com> * Based on pxa2xx Platform drivers by - * Liam Girdwood <liam.girdwood@wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> * * 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 diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 43dd8cee83c6..70657534e6b1 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -164,38 +164,38 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, */ switch (params_rate(params)) { case 48000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_1; + pll_out = 24576000; + mclk_div = WM8510_MCLKDIV_2; bclk = WM8510_BCLKDIV_8; break; case 44100: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_1; + pll_out = 22579200; + mclk_div = WM8510_MCLKDIV_2; bclk = WM8510_BCLKDIV_8; break; case 22050: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_2; + pll_out = 22579200; + mclk_div = WM8510_MCLKDIV_4; bclk = WM8510_BCLKDIV_8; break; case 16000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_3; + pll_out = 24576000; + mclk_div = WM8510_MCLKDIV_6; bclk = WM8510_BCLKDIV_8; break; case 11025: - pll_out = 11289600; - mclk_div = WM8510_MCLKDIV_4; + pll_out = 22579200; + mclk_div = WM8510_MCLKDIV_8; bclk = WM8510_BCLKDIV_8; break; case 8000: - pll_out = 12288000; - mclk_div = WM8510_MCLKDIV_6; + pll_out = 24576000; + mclk_div = WM8510_MCLKDIV_12; bclk = WM8510_BCLKDIV_8; break; diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 6ea04be911d0..173a239a541c 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -36,6 +36,7 @@ #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> +#include <linux/i2c.h> #include <linux/atmel-ssc.h> @@ -45,6 +46,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> @@ -52,6 +54,9 @@ #include "atmel-pcm.h" #include "atmel_ssc_dai.h" +#define MCLK_RATE 12000000 + +static struct clk *mclk; static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) { @@ -59,11 +64,12 @@ static int at91sam9g20ek_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; int ret; - /* codec system clock is supplied by PCK0, set to 12MHz */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, - 12000000, SND_SOC_CLOCK_IN); - if (ret < 0) + MCLK_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) { + clk_disable(mclk); return ret; + } return 0; } @@ -189,6 +195,31 @@ static struct snd_soc_ops at91sam9g20ek_ops = { .shutdown = at91sam9g20ek_shutdown, }; +static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, + enum snd_soc_bias_level level) +{ + static int mclk_on; + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + if (!mclk_on) + ret = clk_enable(mclk); + if (ret == 0) + mclk_on = 1; + break; + + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + if (mclk_on) + clk_disable(mclk); + mclk_on = 0; + break; + } + + return ret; +} static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), @@ -243,21 +274,48 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { }; static struct snd_soc_card snd_soc_at91sam9g20ek = { - .name = "WM8731", + .name = "AT91SAMG20-EK", .platform = &atmel_soc_platform, .dai_link = &at91sam9g20ek_dai, .num_links = 1, + .set_bias_level = at91sam9g20ek_set_bias_level, }; -static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8731 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8731_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} static struct snd_soc_device at91sam9g20ek_snd_devdata = { .card = &snd_soc_at91sam9g20ek, .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &at91sam9g20ek_wm8731_setup, }; static struct platform_device *at91sam9g20ek_snd_device; @@ -266,23 +324,56 @@ static int __init at91sam9g20ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; + struct clk *pllb; int ret; + if (!machine_is_at91sam9g20ek()) + return -ENODEV; + + /* + * Codec MCLK is supplied by PCK0 - set it up. + */ + mclk = clk_get(NULL, "pck0"); + if (IS_ERR(mclk)) { + printk(KERN_ERR "ASoC: Failed to get MCLK\n"); + ret = PTR_ERR(mclk); + goto err; + } + + pllb = clk_get(NULL, "pllb"); + if (IS_ERR(mclk)) { + printk(KERN_ERR "ASoC: Failed to get PLLB\n"); + ret = PTR_ERR(mclk); + goto err_mclk; + } + ret = clk_set_parent(mclk, pllb); + clk_put(pllb); + if (ret != 0) { + printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); + goto err_mclk; + } + + clk_set_rate(mclk, MCLK_RATE); + /* * Request SSC device */ ssc = ssc_request(0); if (IS_ERR(ssc)) { + printk(KERN_ERR "ASoC: Failed to request SSC 0\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; + ret = wm8731_i2c_register(); + if (ret != 0) + goto err_ssc; + at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g20ek_snd_device) { - printk(KERN_DEBUG - "platform device allocation failed\n"); + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } @@ -292,14 +383,19 @@ static int __init at91sam9g20ek_init(void) ret = platform_device_add(at91sam9g20ek_snd_device); if (ret) { - printk(KERN_DEBUG - "platform device allocation failed\n"); + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g20ek_snd_device); } return ret; err_ssc: + ssc_free(ssc); + ssc_p->ssc = NULL; +err_mclk: + clk_put(mclk); + mclk = NULL; +err: return ret; } @@ -317,6 +413,8 @@ static void __exit at91sam9g20ek_exit(void) platform_device_unregister(at91sam9g20ek_snd_device); at91sam9g20ek_snd_device = NULL; + clk_put(mclk); + mclk = NULL; } module_init(at91sam9g20ek_init); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index bc8d654576c0..30490a259148 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -305,7 +305,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream) return 0; } -struct snd_pcm_ops au1xpsc_pcm_ops = { +static struct snd_pcm_ops au1xpsc_pcm_ops = { .open = au1xpsc_pcm_open, .close = au1xpsc_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index f0e30aec7f23..479d7bdf1865 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", .ac97_control = 1, @@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, - }, + .ops = &au1xpsc_ac97_dai_ops, }; EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index f916de4400ed..bb589327ee32 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) return 0; } +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", .probe = au1xpsc_i2s_probe, @@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .channels_min = 2, .channels_max = 8, /* 2 without external help */ }, - .ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, - }, + .ops = &au1xpsc_i2s_dai_ops, }; EXPORT_SYMBOL(au1xpsc_i2s_dai); diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 8067cfafa3a7..8cfed1a5dcbe 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -297,7 +297,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, } #endif -struct snd_pcm_ops bf5xx_pcm_ac97_ops = { +static struct snd_pcm_ops bf5xx_pcm_ac97_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 3be2be60576d..8a935f2d1767 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -31,72 +31,46 @@ #include "bf5xx-sport.h" #include "bf5xx-ac97.h" -#if defined(CONFIG_BF54x) -#define PIN_REQ_SPORT_0 {P_SPORT0_TFS, P_SPORT0_DTPRI, P_SPORT0_TSCLK, \ - P_SPORT0_RFS, P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0} - -#define PIN_REQ_SPORT_1 {P_SPORT1_TFS, P_SPORT1_DTPRI, P_SPORT1_TSCLK, \ - P_SPORT1_RFS, P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0} - -#define PIN_REQ_SPORT_2 {P_SPORT2_TFS, P_SPORT2_DTPRI, P_SPORT2_TSCLK, \ - P_SPORT2_RFS, P_SPORT2_DRPRI, P_SPORT2_RSCLK, 0} - -#define PIN_REQ_SPORT_3 {P_SPORT3_TFS, P_SPORT3_DTPRI, P_SPORT3_TSCLK, \ - P_SPORT3_RFS, P_SPORT3_DRPRI, P_SPORT3_RSCLK, 0} -#else -#define PIN_REQ_SPORT_0 {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, \ - P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0} - -#define PIN_REQ_SPORT_1 {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, \ - P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0} -#endif - static int *cmd_count; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; +#define SPORT_REQ(x) \ + [x] = {P_SPORT##x##_TFS, P_SPORT##x##_DTPRI, P_SPORT##x##_TSCLK, \ + P_SPORT##x##_RFS, P_SPORT##x##_DRPRI, P_SPORT##x##_RSCLK, 0} static u16 sport_req[][7] = { - PIN_REQ_SPORT_0, -#ifdef PIN_REQ_SPORT_1 - PIN_REQ_SPORT_1, +#ifdef SPORT0_TCR1 + SPORT_REQ(0), +#endif +#ifdef SPORT1_TCR1 + SPORT_REQ(1), #endif -#ifdef PIN_REQ_SPORT_2 - PIN_REQ_SPORT_2, +#ifdef SPORT2_TCR1 + SPORT_REQ(2), #endif -#ifdef PIN_REQ_SPORT_3 - PIN_REQ_SPORT_3, +#ifdef SPORT3_TCR1 + SPORT_REQ(3), #endif - }; +}; +#define SPORT_PARAMS(x) \ + [x] = { \ + .dma_rx_chan = CH_SPORT##x##_RX, \ + .dma_tx_chan = CH_SPORT##x##_TX, \ + .err_irq = IRQ_SPORT##x##_ERROR, \ + .regs = (struct sport_register *)SPORT##x##_TCR1, \ + } static struct sport_param sport_params[4] = { - { - .dma_rx_chan = CH_SPORT0_RX, - .dma_tx_chan = CH_SPORT0_TX, - .err_irq = IRQ_SPORT0_ERROR, - .regs = (struct sport_register *)SPORT0_TCR1, - }, -#ifdef PIN_REQ_SPORT_1 - { - .dma_rx_chan = CH_SPORT1_RX, - .dma_tx_chan = CH_SPORT1_TX, - .err_irq = IRQ_SPORT1_ERROR, - .regs = (struct sport_register *)SPORT1_TCR1, - }, +#ifdef SPORT0_TCR1 + SPORT_PARAMS(0), #endif -#ifdef PIN_REQ_SPORT_2 - { - .dma_rx_chan = CH_SPORT2_RX, - .dma_tx_chan = CH_SPORT2_TX, - .err_irq = IRQ_SPORT2_ERROR, - .regs = (struct sport_register *)SPORT2_TCR1, - }, +#ifdef SPORT1_TCR1 + SPORT_PARAMS(1), #endif -#ifdef PIN_REQ_SPORT_3 - { - .dma_rx_chan = CH_SPORT3_RX, - .dma_tx_chan = CH_SPORT3_TX, - .err_irq = IRQ_SPORT3_ERROR, - .regs = (struct sport_register *)SPORT3_TCR1, - } +#ifdef SPORT2_TCR1 + SPORT_PARAMS(2), +#endif +#ifdef SPORT3_TCR1 + SPORT_PARAMS(3), #endif }; @@ -332,11 +306,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev, if (cmd_count == NULL) return -ENOMEM; - if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) { + if (peripheral_request_list(sport_req[sport_num], "soc-audio")) { pr_err("Requesting Peripherals failed\n"); ret = -EFAULT; goto peripheral_err; - } + } #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET /* Request PB3 as reset pin */ @@ -383,9 +357,9 @@ sport_config_err: sport_err: #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); -#endif gpio_err: - peripheral_free_list(&sport_req[sport_num][0]); +#endif + peripheral_free_list(sport_req[sport_num]); peripheral_err: free_page((unsigned long)cmd_count); cmd_count = NULL; @@ -398,7 +372,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, { free_page((unsigned long)cmd_count); cmd_count = NULL; - peripheral_free_list(&sport_req[sport_num][0]); + peripheral_free_list(sport_req[sport_num]); #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM); #endif diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 7f2a5e199075..edfbdc024e66 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -114,7 +114,7 @@ static int snd_ad73311_configure(void) SSYNC(); /* When TUVF is set, the data is already send out */ - while (!(status & TUVF) && count++ < 10000) { + while (!(status & TUVF) && ++count < 10000) { udelay(1); status = bfin_read_SPORT_STAT(); SSYNC(); @@ -123,7 +123,7 @@ static int snd_ad73311_configure(void) SSYNC(); local_irq_enable(); - if (count == 10000) { + if (count >= 10000) { printk(KERN_ERR "ad73311: failed to configure codec\n"); return -1; } diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 53d290b3ea47..1318c4f627b7 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -184,7 +184,7 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, return 0 ; } -struct snd_pcm_ops bf5xx_pcm_i2s_ops = { +static struct snd_pcm_ops bf5xx_pcm_i2s_ops = { .open = bf5xx_pcm_open, .ioctl = snd_pcm_lib_ioctl, .hw_params = bf5xx_pcm_hw_params, diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index d1d95d2393fe..964824419678 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { + .startup = bf5xx_i2s_startup, + .shutdown = bf5xx_i2s_shutdown, + .hw_params = bf5xx_i2s_hw_params, + .set_fmt = bf5xx_i2s_set_dai_fmt, +}; + struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, @@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = { .channels_max = 2, .rates = BF5XX_I2S_RATES, .formats = BF5XX_I2S_FORMATS,}, - .ops = { - .startup = bf5xx_i2s_startup, - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, - }, + .ops = &bf5xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(bf5xx_i2s_dai); diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c index 3b99e484d555..b7953c8cf838 100644 --- a/sound/soc/blackfin/bf5xx-sport.c +++ b/sound/soc/blackfin/bf5xx-sport.c @@ -133,7 +133,7 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount, int i; for (i = 0; i < fragcount; ++i) { - desc[i].next_desc_addr = (unsigned long)&(desc[i + 1]); + desc[i].next_desc_addr = &(desc[i + 1]); desc[i].start_addr = (unsigned long)buf + i*fragsize; desc[i].cfg = cfg; desc[i].x_count = x_count; @@ -143,12 +143,12 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount, } /* make circular */ - desc[fragcount-1].next_desc_addr = (unsigned long)desc; + desc[fragcount-1].next_desc_addr = desc; - pr_debug("setup desc: desc0=%p, next0=%lx, desc1=%p," - "next1=%lx\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n", - &(desc[0]), desc[0].next_desc_addr, - &(desc[1]), desc[1].next_desc_addr, + pr_debug("setup desc: desc0=%p, next0=%p, desc1=%p," + "next1=%p\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n", + desc, desc[0].next_desc_addr, + desc+1, desc[1].next_desc_addr, desc[0].x_count, desc[0].y_count, desc[0].start_addr, desc[0].cfg); } @@ -184,22 +184,20 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport) BUG_ON(sport->curr_rx_desc == sport->dummy_rx_desc); /* Maybe the dummy buffer descriptor ring is damaged */ - sport->dummy_rx_desc->next_desc_addr = \ - (unsigned long)(sport->dummy_rx_desc+1); + sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1; local_irq_save(flags); - desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_rx_chan); + desc = get_dma_next_desc_ptr(sport->dma_rx_chan); /* Copy the descriptor which will be damaged to backup */ temp_desc = *desc; desc->x_count = 0xa; desc->y_count = 0; - desc->next_desc_addr = (unsigned long)(sport->dummy_rx_desc); + desc->next_desc_addr = sport->dummy_rx_desc; local_irq_restore(flags); /* Waiting for dummy buffer descriptor is already hooked*/ while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != - (unsigned long)sport->dummy_rx_desc) - ; + sizeof(struct dmasg)) != sport->dummy_rx_desc) + continue; sport->curr_rx_desc = sport->dummy_rx_desc; /* Restore the damaged descriptor */ *desc = temp_desc; @@ -210,14 +208,12 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport) static inline int sport_rx_dma_start(struct sport_device *sport, int dummy) { if (dummy) { - sport->dummy_rx_desc->next_desc_addr = \ - (unsigned long) sport->dummy_rx_desc; + sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc; sport->curr_rx_desc = sport->dummy_rx_desc; } else sport->curr_rx_desc = sport->dma_rx_desc; - set_dma_next_desc_addr(sport->dma_rx_chan, \ - (unsigned long)(sport->curr_rx_desc)); + set_dma_next_desc_addr(sport->dma_rx_chan, sport->curr_rx_desc); set_dma_x_count(sport->dma_rx_chan, 0); set_dma_x_modify(sport->dma_rx_chan, 0); set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \ @@ -231,14 +227,12 @@ static inline int sport_rx_dma_start(struct sport_device *sport, int dummy) static inline int sport_tx_dma_start(struct sport_device *sport, int dummy) { if (dummy) { - sport->dummy_tx_desc->next_desc_addr = \ - (unsigned long) sport->dummy_tx_desc; + sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc; sport->curr_tx_desc = sport->dummy_tx_desc; } else sport->curr_tx_desc = sport->dma_tx_desc; - set_dma_next_desc_addr(sport->dma_tx_chan, \ - (unsigned long)(sport->curr_tx_desc)); + set_dma_next_desc_addr(sport->dma_tx_chan, sport->curr_tx_desc); set_dma_x_count(sport->dma_tx_chan, 0); set_dma_x_modify(sport->dma_tx_chan, 0); set_dma_config(sport->dma_tx_chan, @@ -261,11 +255,9 @@ int sport_rx_start(struct sport_device *sport) BUG_ON(sport->curr_rx_desc != sport->dummy_rx_desc); local_irq_save(flags); while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != - (unsigned long)sport->dummy_rx_desc) - ; - sport->dummy_rx_desc->next_desc_addr = - (unsigned long)(sport->dma_rx_desc); + sizeof(struct dmasg)) != sport->dummy_rx_desc) + continue; + sport->dummy_rx_desc->next_desc_addr = sport->dma_rx_desc; local_irq_restore(flags); sport->curr_rx_desc = sport->dma_rx_desc; } else { @@ -310,23 +302,21 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport) BUG_ON(sport->dummy_tx_desc == NULL); BUG_ON(sport->curr_tx_desc == sport->dummy_tx_desc); - sport->dummy_tx_desc->next_desc_addr = \ - (unsigned long)(sport->dummy_tx_desc+1); + sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1; /* Shorten the time on last normal descriptor */ local_irq_save(flags); - desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_tx_chan); + desc = get_dma_next_desc_ptr(sport->dma_tx_chan); /* Store the descriptor which will be damaged */ temp_desc = *desc; desc->x_count = 0xa; desc->y_count = 0; - desc->next_desc_addr = (unsigned long)(sport->dummy_tx_desc); + desc->next_desc_addr = sport->dummy_tx_desc; local_irq_restore(flags); /* Waiting for dummy buffer descriptor is already hooked*/ while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \ - sizeof(struct dmasg)) != \ - (unsigned long)sport->dummy_tx_desc) - ; + sizeof(struct dmasg)) != sport->dummy_tx_desc) + continue; sport->curr_tx_desc = sport->dummy_tx_desc; /* Restore the damaged descriptor */ *desc = temp_desc; @@ -347,11 +337,9 @@ int sport_tx_start(struct sport_device *sport) /* Hook the normal buffer descriptor */ local_irq_save(flags); while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - - sizeof(struct dmasg)) != - (unsigned long)sport->dummy_tx_desc) - ; - sport->dummy_tx_desc->next_desc_addr = - (unsigned long)(sport->dma_tx_desc); + sizeof(struct dmasg)) != sport->dummy_tx_desc) + continue; + sport->dummy_tx_desc->next_desc_addr = sport->dma_tx_desc; local_irq_restore(flags); sport->curr_tx_desc = sport->dma_tx_desc; } else { @@ -536,19 +524,17 @@ static int sport_config_rx_dummy(struct sport_device *sport) unsigned config; pr_debug("%s entered\n", __func__); -#if L1_DATA_A_LENGTH != 0 - desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc)); -#else - { + if (L1_DATA_A_LENGTH) + desc = l1_data_sram_zalloc(2 * sizeof(*desc)); + else { dma_addr_t addr; desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); + memset(desc, 0, 2 * sizeof(*desc)); } -#endif if (desc == NULL) { pr_err("Failed to allocate memory for dummy rx desc\n"); return -ENOMEM; } - memset(desc, 0, 2 * sizeof(*desc)); sport->dummy_rx_desc = desc; desc->start_addr = (unsigned long)sport->dummy_buf; config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize) @@ -559,8 +545,8 @@ static int sport_config_rx_dummy(struct sport_device *sport) desc->y_count = 0; desc->y_modify = 0; memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = (unsigned long)(desc+1); - desc[1].next_desc_addr = (unsigned long)desc; + desc->next_desc_addr = desc + 1; + desc[1].next_desc_addr = desc; return 0; } @@ -571,19 +557,17 @@ static int sport_config_tx_dummy(struct sport_device *sport) pr_debug("%s entered\n", __func__); -#if L1_DATA_A_LENGTH != 0 - desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc)); -#else - { + if (L1_DATA_A_LENGTH) + desc = l1_data_sram_zalloc(2 * sizeof(*desc)); + else { dma_addr_t addr; desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); + memset(desc, 0, 2 * sizeof(*desc)); } -#endif if (!desc) { pr_err("Failed to allocate memory for dummy tx desc\n"); return -ENOMEM; } - memset(desc, 0, 2 * sizeof(*desc)); sport->dummy_tx_desc = desc; desc->start_addr = (unsigned long)sport->dummy_buf + \ sport->dummy_count; @@ -595,8 +579,8 @@ static int sport_config_tx_dummy(struct sport_device *sport) desc->y_count = 0; desc->y_modify = 0; memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = (unsigned long)(desc+1); - desc[1].next_desc_addr = (unsigned long)desc; + desc->next_desc_addr = desc + 1; + desc[1].next_desc_addr = desc; return 0; } @@ -872,17 +856,15 @@ struct sport_device *sport_init(struct sport_param *param, unsigned wdsize, sport->wdsize = wdsize; sport->dummy_count = dummy_count; -#if L1_DATA_A_LENGTH != 0 - sport->dummy_buf = l1_data_sram_alloc(dummy_count * 2); -#else - sport->dummy_buf = kmalloc(dummy_count * 2, GFP_KERNEL); -#endif + if (L1_DATA_A_LENGTH) + sport->dummy_buf = l1_data_sram_zalloc(dummy_count * 2); + else + sport->dummy_buf = kzalloc(dummy_count * 2, GFP_KERNEL); if (sport->dummy_buf == NULL) { pr_err("Failed to allocate dummy buffer\n"); goto __error; } - memset(sport->dummy_buf, 0, dummy_count * 2); ret = sport_config_rx_dummy(sport); if (ret) { pr_err("Failed to config rx dummy ring\n"); @@ -939,6 +921,7 @@ void sport_done(struct sport_device *sport) sport = NULL; } EXPORT_SYMBOL(sport_done); + /* * It is only used to send several bytes when dma is not enabled * sport controller is configured but not enabled. @@ -1029,4 +1012,3 @@ EXPORT_SYMBOL(sport_send_and_recv); MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("SPORT driver for ADI Blackfin"); MODULE_LICENSE("GPL"); - diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d0e0d691ae51..b6c7f7a01cb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,9 +10,11 @@ config SND_SOC_I2C_AND_SPI 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_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_CS4270 if I2C select SND_SOC_PCM3008 @@ -24,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C 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_WM8580 if I2C select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI @@ -34,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C select SND_SOC_WM8990 if I2C + select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS select SND_SOC_WM9713 if SND_SOC_AC97_BUS help @@ -58,6 +62,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_AK4104 + tristate + config SND_SOC_AK4535 tristate @@ -65,12 +72,6 @@ config SND_SOC_AK4535 config SND_SOC_CS4270 tristate -# Cirrus Logic CS4270 Codec Hardware Mute Support -# Select if you have external muting circuitry attached to your CS4270. -config SND_SOC_CS4270_HWMUTE - bool - depends on SND_SOC_CS4270 - # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function # if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will @@ -90,7 +91,6 @@ config SND_SOC_SSM2602 config SND_SOC_TLV320AIC23 tristate - depends on I2C config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE @@ -98,15 +98,12 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate - depends on I2C config SND_SOC_TWL4030 tristate - depends on TWL4030_CORE config SND_SOC_UDA134X tristate - select SND_SOC_L3 config SND_SOC_UDA1380 tristate @@ -114,6 +111,9 @@ config SND_SOC_UDA1380 config SND_SOC_WM8350 tristate +config SND_SOC_WM8400 + tristate + config SND_SOC_WM8510 tristate @@ -144,6 +144,9 @@ config SND_SOC_WM8971 config SND_SOC_WM8990 tristate +config SND_SOC_WM9705 + tristate + config SND_SOC_WM9712 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c4ddc9aa2bbd..030d2454725f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,6 +1,7 @@ snd-soc-ac97-objs := ac97.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-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o @@ -13,6 +14,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o 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-wm8580-objs := wm8580.o snd-soc-wm8728-objs := wm8728.o @@ -23,12 +25,14 @@ snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o snd-soc-wm8990-objs := wm8990.o +snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(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_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o @@ -41,6 +45,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(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_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o @@ -51,5 +56,7 @@ obj-$(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-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o +obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.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 diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index fb53e6511af2..b0d4af145b87 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; @@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ac97_dai_ops = { + .prepare = ac97_prepare, +}; + struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", .ac97_control = 1, @@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = { .channels_max = 2, .rates = STD_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &ac97_dai_ops, }; EXPORT_SYMBOL_GPL(ac97_dai); @@ -84,10 +87,10 @@ static int ac97_soc_probe(struct platform_device *pdev) printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (!socdev->codec) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->card->codec) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->name = "AC97"; @@ -123,23 +126,21 @@ bus_err: snd_soc_free_pcms(socdev); err: - kfree(socdev->codec->reg_cache); - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int ac97_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (!codec) return 0; snd_soc_free_pcms(socdev); - kfree(socdev->codec->reg_cache); - kfree(socdev->codec); + kfree(socdev->card->codec); return 0; } @@ -149,7 +150,7 @@ static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - snd_ac97_suspend(socdev->codec->ac97); + snd_ac97_suspend(socdev->card->codec->ac97); return 0; } @@ -158,7 +159,7 @@ static int ac97_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - snd_ac97_resume(socdev->codec->ac97); + snd_ac97_resume(socdev->card->codec->ac97); return 0; } diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 73fdbb4d4a3d..ddb3b08ac23c 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; -/* add non dapm controls */ -static int ad1980_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, snd_soc_cnew( - &ad1980_snd_ac97_controls[i], codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -123,7 +109,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, default: reg = reg >> 1; - if (reg >= (ARRAY_SIZE(ad1980_reg))) + if (reg >= ARRAY_SIZE(ad1980_reg)) return -EINVAL; return cache[reg]; @@ -137,7 +123,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg < (ARRAY_SIZE(ad1980_reg))) + if (reg < ARRAY_SIZE(ad1980_reg)) cache[reg] = val; return 0; @@ -200,10 +186,10 @@ static int ad1980_soc_probe(struct platform_device *pdev) printk(KERN_INFO "AD1980 SoC Audio Codec\n"); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = @@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - ad1980_add_controls(codec); + snd_soc_add_controls(codec, ad1980_snd_ac97_controls, + ARRAY_SIZE(ad1980_snd_ac97_controls)); ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register card\n"); @@ -288,15 +275,15 @@ codec_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int ad1980_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b09289a1e55a..e61dac5e7b8f 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -53,7 +53,7 @@ static int ad73311_soc_probe(struct platform_device *pdev) codec->owner = THIS_MODULE; codec->dai = &ad73311_dai; codec->num_dai = 1; - socdev->codec = codec; + socdev->card->codec = codec; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -75,15 +75,15 @@ static int ad73311_soc_probe(struct platform_device *pdev) register_err: snd_soc_free_pcms(socdev); pcm_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int ad73311_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h index 507ce0c30edf..569573d2d4d7 100644 --- a/sound/soc/codecs/ad73311.h +++ b/sound/soc/codecs/ad73311.h @@ -70,7 +70,7 @@ #define REGD_IGS(x) (x & 0x7) #define REGD_RMOD (1 << 3) #define REGD_OGS(x) ((x & 0x7) << 4) -#define REGD_MUTE (x << 7) +#define REGD_MUTE (1 << 7) /* Control register E */ #define CTRL_REG_E (4 << 8) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c new file mode 100644 index 000000000000..4d47bc4f7428 --- /dev/null +++ b/sound/soc/codecs/ak4104.c @@ -0,0 +1,365 @@ +/* + * AK4104 ALSA SoC (ASoC) driver + * + * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de> + * + * 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 <sound/core.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <linux/spi/spi.h> +#include <sound/asoundef.h> + +#include "ak4104.h" + +/* AK4104 registers addresses */ +#define AK4104_REG_CONTROL1 0x00 +#define AK4104_REG_RESERVED 0x01 +#define AK4104_REG_CONTROL2 0x02 +#define AK4104_REG_TX 0x03 +#define AK4104_REG_CHN_STATUS(x) ((x) + 0x04) +#define AK4104_NUM_REGS 10 + +#define AK4104_REG_MASK 0x1f +#define AK4104_READ 0xc0 +#define AK4104_WRITE 0xe0 +#define AK4104_RESERVED_VAL 0x5b + +/* Bit masks for AK4104 registers */ +#define AK4104_CONTROL1_RSTN (1 << 0) +#define AK4104_CONTROL1_PW (1 << 1) +#define AK4104_CONTROL1_DIF0 (1 << 2) +#define AK4104_CONTROL1_DIF1 (1 << 3) + +#define AK4104_CONTROL2_SEL0 (1 << 0) +#define AK4104_CONTROL2_SEL1 (1 << 1) +#define AK4104_CONTROL2_MODE (1 << 2) + +#define AK4104_TX_TXE (1 << 0) +#define AK4104_TX_V (1 << 1) + +#define DRV_NAME "ak4104" + +struct ak4104_private { + struct snd_soc_codec codec; + u8 reg_cache[AK4104_NUM_REGS]; +}; + +static int ak4104_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 = spi_w8r8(spi, i | AK4104_READ); + if (ret < 0) { + dev_err(&spi->dev, "SPI write failure\n"); + return ret; + } + + reg_cache[i] = ret; + } + + return 0; +} + +static unsigned int ak4104_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]; +} + +static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + struct spi_device *spi = codec->control_data; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + reg &= AK4104_REG_MASK; + reg |= AK4104_WRITE; + + /* only write to the hardware if value has changed */ + if (cache[reg] != value) { + u8 tmp[2] = { reg, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { + dev_err(&spi->dev, "SPI write failed\n"); + return -EIO; + } + + cache[reg] = value; + } + + return 0; +} + +static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int val = 0; + + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1); + + /* set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= AK4104_CONTROL1_DIF0; + break; + case SND_SOC_DAIFMT_I2S: + val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1; + break; + default: + dev_err(codec->dev, "invalid dai format\n"); + return -EINVAL; + } + + /* This device can only be slave */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); +} + +static int ak4104_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; + int val = 0; + + /* set the IEC958 bits: consumer mode, no copyright bit */ + val |= IEC958_AES0_CON_NOT_COPYRIGHT; + ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val); + + val = 0; + + switch (params_rate(params)) { + case 44100: + val |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + val |= IEC958_AES3_CON_FS_48000; + break; + case 32000: + val |= IEC958_AES3_CON_FS_32000; + break; + default: + dev_err(codec->dev, "unsupported sampling rate\n"); + return -EINVAL; + } + + return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val); +} + +static struct snd_soc_dai_ops ak4101_dai_ops = { + .hw_params = ak4104_hw_params, + .set_fmt = ak4104_set_dai_fmt, +}; + +struct snd_soc_dai ak4104_dai = { + .name = DRV_NAME, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_32000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S24_LE + }, + .ops = &ak4101_dai_ops, +}; + +static struct snd_soc_codec *ak4104_codec; + +static int ak4104_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ak4104_private *ak4104; + int ret, val; + + spi->bits_per_word = 8; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL); + if (!ak4104) { + dev_err(&spi->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + codec = &ak4104->codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &spi->dev; + codec->name = DRV_NAME; + codec->owner = THIS_MODULE; + codec->dai = &ak4104_dai; + codec->num_dai = 1; + codec->private_data = ak4104; + codec->control_data = spi; + codec->reg_cache = ak4104->reg_cache; + codec->reg_cache_size = AK4104_NUM_REGS; + + /* read all regs and fill the cache */ + ret = ak4104_fill_cache(codec); + if (ret < 0) { + dev_err(&spi->dev, "failed to fill register cache\n"); + return ret; + } + + /* read the 'reserved' register - according to the datasheet, it + * should contain 0x5b. Not a good way to verify the presence of + * the device, but there is no hardware ID register. */ + if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) != + AK4104_RESERVED_VAL) { + ret = -ENODEV; + goto error_free_codec; + } + + /* set power-up and non-reset bits */ + val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); + val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN; + ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + goto error_free_codec; + + /* enable transmitter */ + val = ak4104_read_reg_cache(codec, AK4104_REG_TX); + val |= AK4104_TX_TXE; + ret = ak4104_spi_write(codec, AK4104_REG_TX, val); + if (ret < 0) + goto error_free_codec; + + ak4104_codec = codec; + ret = snd_soc_register_dai(&ak4104_dai); + if (ret < 0) { + dev_err(&spi->dev, "failed to register DAI\n"); + goto error_free_codec; + } + + spi_set_drvdata(spi, ak4104); + dev_info(&spi->dev, "SPI device initialized\n"); + return 0; + +error_free_codec: + kfree(ak4104); + ak4104_dai.dev = NULL; + return ret; +} + +static int __devexit ak4104_spi_remove(struct spi_device *spi) +{ + int ret, val; + struct ak4104_private *ak4104 = spi_get_drvdata(spi); + + val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1); + if (val < 0) + return val; + + /* clear power-up and non-reset bits */ + val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val); + if (ret < 0) + return ret; + + ak4104_codec = NULL; + kfree(ak4104); + return 0; +} + +static int ak4104_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = ak4104_codec; + int ret; + + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = 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\n"); + return ret; + } + + /* Register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return 0; +} + +static int ak4104_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + return 0; +}; + +struct snd_soc_codec_device soc_codec_device_ak4104 = { + .probe = ak4104_probe, + .remove = ak4104_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_ak4104); + +static struct spi_driver ak4104_spi_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = ak4104_spi_probe, + .remove = __devexit_p(ak4104_spi_remove), +}; + +static int __init ak4104_init(void) +{ + pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); + return spi_register_driver(&ak4104_spi_driver); +} +module_init(ak4104_init); + +static void __exit ak4104_exit(void) +{ + spi_unregister_driver(&ak4104_spi_driver); +} +module_exit(ak4104_exit); + +MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); +MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h new file mode 100644 index 000000000000..eb88fe7e4def --- /dev/null +++ b/sound/soc/codecs/ak4104.h @@ -0,0 +1,7 @@ +#ifndef _AK4104_H +#define _AK4104_H + +extern struct snd_soc_dai ak4104_dai; +extern struct snd_soc_codec_device soc_codec_device_ak4104; + +#endif diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 81300d8d42ca..1f63d387a2f4 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = { SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), }; -/* add non dapm controls */ -static int ak4535_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Mono 1 Mixer */ static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), @@ -344,7 +329,7 @@ static int ak4535_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ak4535_priv *ak4535 = codec->private_data; u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5); int rate = params_rate(params), fs = 256; @@ -436,6 +421,13 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops ak4535_dai_ops = { + .hw_params = ak4535_hw_params, + .set_fmt = ak4535_set_dai_fmt, + .digital_mute = ak4535_mute, + .set_sysclk = ak4535_set_dai_sysclk, +}; + struct snd_soc_dai ak4535_dai = { .name = "AK4535", .playback = { @@ -450,19 +442,14 @@ struct snd_soc_dai ak4535_dai = { .channels_max = 2, .rates = AK4535_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = ak4535_hw_params, - .set_fmt = ak4535_set_dai_fmt, - .digital_mute = ak4535_mute, - .set_sysclk = ak4535_set_dai_sysclk, - }, + .ops = &ak4535_dai_ops, }; EXPORT_SYMBOL_GPL(ak4535_dai); static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -471,7 +458,7 @@ static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) static int ak4535_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; ak4535_sync(codec); ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); ak4535_set_bias_level(codec, codec->suspend_bias_level); @@ -484,7 +471,7 @@ static int ak4535_resume(struct platform_device *pdev) */ static int ak4535_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "AK4535"; @@ -510,7 +497,8 @@ static int ak4535_init(struct snd_soc_device *socdev) /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ak4535_add_controls(codec); + snd_soc_add_controls(codec, ak4535_snd_controls, + ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -537,7 +525,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = ak4535_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -636,7 +624,7 @@ static int ak4535_probe(struct platform_device *pdev) } codec->private_data = ak4535; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -663,7 +651,7 @@ static int ak4535_probe(struct platform_device *pdev) static int ak4535_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index f1aa0c34421c..7fa09a387622 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -3,27 +3,22 @@ * * Author: Timur Tabi <timur@freescale.com> * - * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under - * the terms of the GNU General Public License version 2. This program - * is licensed "as is" without any warranty of any kind, whether express - * or implied. + * Copyright 2007-2009 Freescale Semiconductor, Inc. This file is licensed + * under the terms of the GNU General Public License version 2. This + * program is licensed "as is" without any warranty of any kind, whether + * express or implied. * * This is an ASoC device driver for the Cirrus Logic CS4270 codec. * * Current features/limitations: * - * 1) Software mode is supported. Stand-alone mode is automatically - * selected if I2C is disabled or if a CS4270 is not found on the I2C - * bus. However, stand-alone mode is only partially implemented because - * there is no mechanism yet for this driver and the machine driver to - * communicate the values of the M0, M1, MCLK1, and MCLK2 pins. - * 2) Only I2C is supported, not SPI - * 3) Only Master mode is supported, not Slave. - * 4) The machine driver's 'startup' function must call - * cs4270_set_dai_sysclk() with the value of MCLK. - * 5) Only I2S and left-justified modes are supported - * 6) Power management is not supported - * 7) The only supported control is volume and hardware mute (if enabled) + * - Software mode is supported. Stand-alone mode is not supported. + * - Only I2C is supported, not SPI + * - Support for master and slave mode + * - The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * - Only I2S and left-justified modes are supported + * - Power management is not supported */ #include <linux/module.h> @@ -35,18 +30,6 @@ #include "cs4270.h" -/* If I2C is defined, then we support software mode. However, if we're - not compiled as module but I2C is, then we can't use I2C calls. */ -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -#define USE_I2C -#endif - -/* Private data for the CS4270 */ -struct cs4270_private { - unsigned int mclk; /* Input frequency of the MCLK pin */ - unsigned int mode; /* The mode (I2S or left-justified) */ -}; - /* * The codec isn't really big-endian or little-endian, since the I2S * interface requires data to be sent serially with the MSbit first. @@ -60,8 +43,6 @@ struct cs4270_private { SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -#ifdef USE_I2C - /* CS4270 registers addresses */ #define CS4270_CHIPID 0x01 /* Chip ID */ #define CS4270_PWRCTL 0x02 /* Power Control */ @@ -121,8 +102,22 @@ struct cs4270_private { #define CS4270_MUTE_DAC_A 0x01 #define CS4270_MUTE_DAC_B 0x02 -/* - * Clock Ratio Selection for Master Mode with I2C enabled +/* Private data for the CS4270 */ +struct cs4270_private { + struct snd_soc_codec codec; + u8 reg_cache[CS4270_NUMREGS]; + unsigned int mclk; /* Input frequency of the MCLK pin */ + unsigned int mode; /* The mode (I2S or left-justified) */ + unsigned int slave_mode; +}; + +/** + * struct cs4270_mode_ratios - clock ratio tables + * @ratio: the ratio of MCLK to the sample rate + * @speed_mode: the Speed Mode bits to set in the Mode Control register for + * this ratio + * @mclk: the Ratio Select bits to set in the Mode Control register for this + * ratio * * The data for this chart is taken from Table 5 of the CS4270 reference * manual. @@ -131,31 +126,30 @@ struct cs4270_private { * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling * rates the CS4270 currently supports. * - * Each element in this array corresponds to the ratios in mclk_ratios[]. - * These two arrays need to be in sync. - * - * 'speed_mode' is the corresponding bit pattern to be written to the + * @speed_mode is the corresponding bit pattern to be written to the * MODE bits of the Mode Control Register * - * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of * the Mode Control Register. * * In situations where a single ratio is represented by multiple speed * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick * double-speed instead of quad-speed. However, the CS4270 errata states - * that Divide-By-1.5 can cause failures, so we avoid that mode where + * that divide-By-1.5 can cause failures, so we avoid that mode where * possible. * - * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not - * work if VD = 3.3V. If this effects you, select the + * Errata: There is an errata for the CS4270 where divide-by-1.5 does not + * work if Vd is 3.3V. If this effects you, select the * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will * never select any sample rates that require divide-by-1.5. */ -static struct { +struct cs4270_mode_ratios { unsigned int ratio; u8 speed_mode; u8 mclk; -} cs4270_mode_ratios[] = { +}; + +static struct cs4270_mode_ratios cs4270_mode_ratios[] = { {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, @@ -172,34 +166,27 @@ static struct { /* The number of MCLK/LRCK ratios supported by the CS4270 */ #define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) -/* - * Determine the CS4270 samples rates. +/** + * cs4270_set_dai_sysclk - determine the CS4270 samples rates. + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) * - * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * This function is used to tell the codec driver what the input MCLK + * frequency is. * * The value of MCLK is used to determine which sample rates are supported * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine - * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * supported values - 64, 96, 128, 192, 256, 384, 512, 768, and 1024. * * This function calculates the nine ratios and determines which ones match * a standard sample rate. If there's a match, then it is added to the list - * of support sample rates. + * of supported sample rates. * * This function must be called by the machine driver's 'startup' function, * otherwise the list of supported sample rates will not be available in * time for ALSA. - * - * Note that in stand-alone mode, the sample rate is determined by input - * pins M0, M1, MDIV1, and MDIV2. Also in stand-alone mode, divide-by-3 - * is not a programmable option. However, divide-by-3 is not an available - * option in stand-alone mode. This cases two problems: a ratio of 768 is - * not available (it requires divide-by-3) and B) ratios 192 and 384 can - * only be selected with divide-by-1.5, but there is an errate that make - * this selection difficult. - * - * In addition, there is no mechanism for communicating with the machine - * driver what the input settings can be. This would need to be implemented - * for stand-alone mode to work. */ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) @@ -225,7 +212,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, rates &= ~SNDRV_PCM_RATE_KNOT; if (!rates) { - printk(KERN_ERR "cs4270: could not find a valid sample rate\n"); + dev_err(codec->dev, "could not find a valid sample rate\n"); return -EINVAL; } @@ -240,8 +227,10 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -/* - * Configure the codec for the selected audio format +/** + * cs4270_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @format: a SND_SOC_DAIFMT_x value indicating the data format * * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the * codec accordingly. @@ -258,32 +247,43 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai, struct cs4270_private *cs4270 = codec->private_data; int ret = 0; + /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: case SND_SOC_DAIFMT_LEFT_J: cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; break; default: - printk(KERN_ERR "cs4270: invalid DAI format\n"); + dev_err(codec->dev, "invalid dai format\n"); + ret = -EINVAL; + } + + /* set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs4270->slave_mode = 1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs4270->slave_mode = 0; + break; + default: + /* all other modes are unsupported by the hardware */ ret = -EINVAL; } return ret; } -/* - * A list of addresses on which this CS4270 could use. I2C addresses are - * 7 bits. For the CS4270, the upper four bits are always 1001, and the - * lower three bits are determined via the AD2, AD1, and AD0 pins - * (respectively). - */ -static const unsigned short normal_i2c[] = { - 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END -}; -I2C_CLIENT_INSMOD; - -/* - * Pre-fill the CS4270 register cache. +/** + * cs4270_fill_cache - pre-fill the CS4270 register cache. + * @codec: the codec for this CS4270 + * + * This function fills in the CS4270 register cache by reading the register + * values from the hardware. + * + * This CS4270 registers are cached to avoid excessive I2C I/O operations. + * After the initial read to pre-fill the cache, the CS4270 never updates + * the register values, so we won't have a cache coherency problem. * * We use the auto-increment feature of the CS4270 to read all registers in * one shot. @@ -298,7 +298,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { - printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n", + dev_err(codec->dev, "i2c read failure, addr=0x%x\n", i2c_client->addr); return -EIO; } @@ -306,12 +306,17 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) return 0; } -/* - * Read from the CS4270 register cache. +/** + * cs4270_read_reg_cache - read from the CS4270 register cache. + * @codec: the codec for this CS4270 + * @reg: the register to read + * + * This function returns the value for a given register. It reads only from + * the register cache, not the hardware itself. * * This CS4270 registers are cached to avoid excessive I2C I/O operations. * After the initial read to pre-fill the cache, the CS4270 never updates - * the register values, so we won't have a cache coherncy problem. + * the register values, so we won't have a cache coherency problem. */ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) @@ -324,8 +329,11 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, return cache[reg - CS4270_FIRSTREG]; } -/* - * Write to a CS4270 register via the I2C bus. +/** + * cs4270_i2c_write - write to a CS4270 register via the I2C bus. + * @codec: the codec for this CS4270 + * @reg: the register to write + * @value: the value to write to the register * * This function writes the given value to the given CS4270 register, and * also updates the register cache. @@ -346,7 +354,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, if (cache[reg - CS4270_FIRSTREG] != value) { struct i2c_client *client = codec->control_data; if (i2c_smbus_write_byte_data(client, reg, value)) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -357,11 +365,17 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -/* - * Program the CS4270 with the given hardware parameters. +/** + * cs4270_hw_params - program the CS4270 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. * - * The .ops functions are used to provide board-specific data, like - * input frequencies, to this driver. This function takes that information, + * The .ops functions are used to provide board-specific data, like input + * frequencies, to this driver. This function takes that information, * combines it with the hardware parameters provided, and programs the * hardware accordingly. */ @@ -371,7 +385,7 @@ static int cs4270_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct cs4270_private *cs4270 = codec->private_data; int ret; unsigned int i; @@ -391,33 +405,28 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, if (i == NUM_MCLK_RATIOS) { /* We did not find a matching ratio */ - printk(KERN_ERR "cs4270: could not find matching ratio\n"); + dev_err(codec->dev, "could not find matching ratio\n"); return -EINVAL; } - /* Freeze and power-down the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | - CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | - CS4270_PWRCTL_PDN); - if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); - return ret; - } - - /* Program the mode control register */ + /* Set the sample rate */ reg = snd_soc_read(codec, CS4270_MODE); reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); - reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + reg |= cs4270_mode_ratios[i].mclk; + + if (cs4270->slave_mode) + reg |= CS4270_MODE_SLAVE; + else + reg |= cs4270_mode_ratios[i].speed_mode; ret = snd_soc_write(codec, CS4270_MODE, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } - /* Program the format register */ + /* Set the DAI format */ reg = snd_soc_read(codec, CS4270_FORMAT); reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); @@ -430,55 +439,23 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; break; default: - printk(KERN_ERR "cs4270: unknown format\n"); + dev_err(codec->dev, "unknown dai format\n"); return -EINVAL; } ret = snd_soc_write(codec, CS4270_FORMAT, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); - return ret; - } - - /* Disable auto-mute. This feature appears to be buggy, because in - some situations, auto-mute will not deactivate when it should. */ - - reg = snd_soc_read(codec, CS4270_MUTE); - reg &= ~CS4270_MUTE_AUTO; - ret = snd_soc_write(codec, CS4270_MUTE, reg); - if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); - return ret; - } - - /* Disable automatic volume control. It's enabled by default, and - * it causes volume change commands to be delayed, sometimes until - * after playback has started. - */ - - reg = cs4270_read_reg_cache(codec, CS4270_TRANS); - reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); - ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); - if (ret < 0) { - printk(KERN_ERR "I2C write failed\n"); - return ret; - } - - /* Thaw and power-up the codec */ - - ret = snd_soc_write(codec, CS4270_PWRCTL, 0); - if (ret < 0) { - printk(KERN_ERR "cs4270: I2C write failed\n"); + dev_err(codec->dev, "i2c write failed\n"); return ret; } return ret; } -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - -/* - * Set the CS4270 external mute +/** + * cs4270_mute - enable/disable the CS4270 external mute + * @dai: the SOC DAI + * @mute: 0 = disable mute, 1 = enable mute * * This function toggles the mute bits in the MUTE register. The CS4270's * mute capability is intended for external muting circuitry, so if the @@ -493,276 +470,306 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute) reg6 = snd_soc_read(codec, CS4270_MUTE); if (mute) - reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; else - reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | - CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); return snd_soc_write(codec, CS4270_MUTE, reg6); } -#endif - -static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *); - /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) -}; - -static const struct i2c_device_id cs4270_id[] = { - {"cs4270", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs4270_id); - -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "CS4270 I2C", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1), + SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), + SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), + SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), + SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), + SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0) }; /* - * Global variable to store socdev for i2c probe function. + * cs4270_codec - global variable to store codec for the ASoC probe function * * If struct i2c_driver had a private_data field, we wouldn't need to use - * cs4270_socdec. This is the only way to pass the socdev structure to - * cs4270_i2c_probe(). - * - * The real solution to cs4270_socdev is to create a mechanism - * that maps I2C addresses to snd_soc_device structures. Perhaps the - * creation of the snd_soc_device object should be moved out of - * cs4270_probe() and into cs4270_i2c_probe(), but that would make this - * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby - * the chip is *not* connected to the I2C bus, but is instead configured via - * input pins. + * cs4270_codec. This is the only way to pass the codec structure from + * cs4270_i2c_probe() to cs4270_probe(). Unfortunately, there is no good + * way to synchronize these two functions. cs4270_i2c_probe() can be called + * multiple times before cs4270_probe() is called even once. So for now, we + * also only allow cs4270_i2c_probe() to be run once. That means that we do + * not support more than one cs4270 device in the system, at least for now. */ -static struct snd_soc_device *cs4270_socdev; +static struct snd_soc_codec *cs4270_codec; -/* - * Initialize the I2C interface of the CS4270 - * - * This function is called for whenever the I2C subsystem finds a device - * at a particular address. +static struct snd_soc_dai_ops cs4270_dai_ops = { + .hw_params = cs4270_hw_params, + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + .digital_mute = cs4270_mute, +}; + +struct snd_soc_dai cs4270_dai = { + .name = "cs4270", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .ops = &cs4270_dai_ops, +}; +EXPORT_SYMBOL_GPL(cs4270_dai); + +/** + * cs4270_probe - ASoC probe function + * @pdev: platform device * - * Note: snd_soc_new_pcms() must be called before this function can be called, - * because of snd_ctl_add(). + * This function is called when ASoC has all the pieces it needs to + * instantiate a sound driver. */ -static int cs4270_i2c_probe(struct i2c_client *i2c_client, - const struct i2c_device_id *id) +static int cs4270_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = cs4270_socdev; - struct snd_soc_codec *codec = socdev->codec; - int i; - int ret = 0; - - /* Probing all possible addresses has one drawback: if there are - multiple CS4270s on the bus, then you cannot specify which - socdev is matched with which CS4270. For now, we just reject - this I2C device if the socdev already has one attached. */ - if (codec->control_data) - return -ENODEV; - - /* Note: codec_dai->codec is NULL here */ - - codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); - if (!codec->reg_cache) { - printk(KERN_ERR "cs4270: could not allocate register cache\n"); - ret = -ENOMEM; - goto error; - } + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = cs4270_codec; + int ret; - /* Verify that we have a CS4270 */ + /* Connect the codec to the socdev. snd_soc_new_pcms() needs this. */ + socdev->card->codec = codec; - ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); + /* Register PCMs */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to read I2C\n"); - goto error; - } - /* The top four bits of the chip ID should be 1100. */ - if ((ret & 0xF0) != 0xC0) { - /* The device at this address is not a CS4270 codec */ - ret = -ENODEV; - goto error; + dev_err(codec->dev, "failed to create pcms\n"); + return ret; } - printk(KERN_INFO "cs4270: found device at I2C address %X\n", - i2c_client->addr); - printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); - - codec->control_data = i2c_client; - codec->read = cs4270_read_reg_cache; - codec->write = cs4270_i2c_write; - codec->reg_cache_size = CS4270_NUMREGS; - - /* The I2C interface is set up, so pre-fill our register cache */ - - ret = cs4270_fill_cache(codec); + /* Add the non-DAPM controls */ + ret = snd_soc_add_controls(codec, cs4270_snd_controls, + ARRAY_SIZE(cs4270_snd_controls)); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to fill register cache\n"); - goto error; + dev_err(codec->dev, "failed to add controls\n"); + goto error_free_pcms; } - /* Add the non-DAPM controls */ - - for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { - struct snd_kcontrol *kctrl = - snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); - - ret = snd_ctl_add(codec->card, kctrl); - if (ret < 0) - goto error; + /* And finally, register the socdev */ + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card\n"); + goto error_free_pcms; } - i2c_set_clientdata(i2c_client, codec); - return 0; -error: - codec->control_data = NULL; - - kfree(codec->reg_cache); - codec->reg_cache = NULL; - codec->reg_cache_size = 0; +error_free_pcms: + snd_soc_free_pcms(socdev); return ret; } -#endif /* USE_I2C*/ +/** + * cs4270_remove - ASoC remove function + * @pdev: platform device + * + * This function is the counterpart to cs4270_probe(). + */ +static int cs4270_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); -struct snd_soc_dai cs4270_dai = { - .name = "CS4270", - .playback = { - .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, - .rates = 0, - .formats = CS4270_FORMATS, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 1, - .channels_max = 2, - .rates = 0, - .formats = CS4270_FORMATS, - }, + snd_soc_free_pcms(socdev); + + return 0; }; -EXPORT_SYMBOL_GPL(cs4270_dai); -/* - * ASoC probe function +/** + * cs4270_i2c_probe - initialize the I2C interface of the CS4270 + * @i2c_client: the I2C client object + * @id: the I2C device ID (ignored) * - * This function is called when the machine driver calls - * platform_device_add(). + * This function is called whenever the I2C subsystem finds a device that + * matches the device ID given via a prior call to i2c_add_driver(). */ -static int cs4270_probe(struct platform_device *pdev) +static int cs4270_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; - int ret = 0; + struct cs4270_private *cs4270; + unsigned int reg; + int ret; - printk(KERN_INFO "CS4270 ALSA SoC Codec\n"); + /* For now, we only support one cs4270 device in the system. See the + * comment for cs4270_codec. + */ + if (cs4270_codec) { + dev_err(&i2c_client->dev, "ignoring CS4270 at addr %X\n", + i2c_client->addr); + dev_err(&i2c_client->dev, "only one per board allowed\n"); + /* Should we return something other than ENODEV here? */ + return -ENODEV; + } + + /* Verify that we have a CS4270 */ + + ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n", + i2c_client->addr); + return ret; + } + /* The top four bits of the chip ID should be 1100. */ + if ((ret & 0xF0) != 0xC0) { + dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n", + i2c_client->addr); + return -ENODEV; + } + + dev_info(&i2c_client->dev, "found device at i2c address %X\n", + i2c_client->addr); + dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF); /* Allocate enough space for the snd_soc_codec structure and our private data together. */ - codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) + - sizeof(struct cs4270_private), GFP_KERNEL); - if (!codec) { - printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL); + if (!cs4270) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); return -ENOMEM; } + codec = &cs4270->codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &i2c_client->dev; codec->name = "CS4270"; codec->owner = THIS_MODULE; codec->dai = &cs4270_dai; codec->num_dai = 1; - codec->private_data = (void *) codec + - ALIGN(sizeof(struct snd_soc_codec), 4); - - socdev->codec = codec; + codec->private_data = cs4270; + codec->control_data = i2c_client; + codec->read = cs4270_read_reg_cache; + codec->write = cs4270_i2c_write; + codec->reg_cache = cs4270->reg_cache; + codec->reg_cache_size = CS4270_NUMREGS; - /* Register PCMs */ + /* The I2C interface is set up, so pre-fill our register cache */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + ret = cs4270_fill_cache(codec); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to create PCMs\n"); + dev_err(&i2c_client->dev, "failed to fill register cache\n"); goto error_free_codec; } -#ifdef USE_I2C - cs4270_socdev = socdev; + /* Disable auto-mute. This feature appears to be buggy. In some + * situations, auto-mute will not deactivate when it should, so we want + * this feature disabled by default. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ - ret = i2c_add_driver(&cs4270_i2c_driver); - if (ret) { - printk(KERN_ERR "cs4270: failed to attach driver"); - goto error_free_pcms; + reg = cs4270_read_reg_cache(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = cs4270_i2c_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; } - /* Did we find a CS4270 on the I2C bus? */ - if (codec->control_data) { - /* Initialize codec ops */ - cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; -#ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.ops.digital_mute = cs4270_mute; -#endif - } else - printk(KERN_INFO "cs4270: no I2C device found, " - "using stand-alone mode\n"); -#else - printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); -#endif + /* Disable automatic volume control. The hardware enables, and it + * causes volume change commands to be delayed, sometimes until after + * playback has started. An application (e.g. alsactl) can + * re-enabled it by using the controls. + */ - ret = snd_soc_init_card(socdev); + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); if (ret < 0) { - printk(KERN_ERR "cs4270: failed to register card\n"); - goto error_del_driver; + dev_err(&i2c_client->dev, "i2c write failed\n"); + return ret; } - return 0; + /* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI + * structure for each CS4270 device, but the machine driver needs to + * have a pointer to the DAI structure, so for now it must be a global + * variable. + */ + cs4270_dai.dev = &i2c_client->dev; -error_del_driver: -#ifdef USE_I2C - i2c_del_driver(&cs4270_i2c_driver); + /* Register the DAI. If all the other ASoC driver have already + * registered, then this will call our probe function, so + * cs4270_codec needs to be ready. + */ + cs4270_codec = codec; + ret = snd_soc_register_dai(&cs4270_dai); + if (ret < 0) { + dev_err(&i2c_client->dev, "failed to register DAIe\n"); + goto error_free_codec; + } -error_free_pcms: -#endif - snd_soc_free_pcms(socdev); + i2c_set_clientdata(i2c_client, cs4270); + + return 0; error_free_codec: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(cs4270); + cs4270_codec = NULL; + cs4270_dai.dev = NULL; return ret; } -static int cs4270_remove(struct platform_device *pdev) +/** + * cs4270_i2c_remove - remove an I2C device + * @i2c_client: the I2C client object + * + * This function is the counterpart to cs4270_i2c_probe(). + */ +static int cs4270_i2c_remove(struct i2c_client *i2c_client) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - - snd_soc_free_pcms(socdev); - -#ifdef USE_I2C - i2c_del_driver(&cs4270_i2c_driver); -#endif + struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client); - kfree(socdev->codec); - socdev->codec = NULL; + kfree(cs4270); + cs4270_codec = NULL; + cs4270_dai.dev = NULL; return 0; } /* + * cs4270_id - I2C device IDs supported by this driver + */ +static struct i2c_device_id cs4270_id[] = { + {"cs4270", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs4270_id); + +/* + * cs4270_i2c_driver - I2C device identification + * + * This structure tells the I2C subsystem how to identify and support a + * given I2C device type. + */ +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, +}; + +/* * ASoC codec device structure * * Assign this variable to the codec_dev field of the machine driver's @@ -776,13 +783,15 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); static int __init cs4270_init(void) { - return snd_soc_register_dai(&cs4270_dai); + pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); + + return i2c_add_driver(&cs4270_i2c_driver); } module_init(cs4270_init); static void __exit cs4270_exit(void) { - snd_soc_unregister_dai(&cs4270_dai); + i2c_del_driver(&cs4270_i2c_driver); } module_exit(cs4270_exit); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 9a3e67e5319c..5cda9e6b5a74 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -67,11 +67,11 @@ static int pcm3008_soc_probe(struct platform_device *pdev) printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (!socdev->codec) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->card->codec) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->name = "PCM3008"; @@ -139,7 +139,7 @@ gpio_err: card_err: snd_soc_free_pcms(socdev); pcm_err: - kfree(socdev->codec); + kfree(socdev->card->codec); return ret; } @@ -147,7 +147,7 @@ pcm_err: static int pcm3008_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct pcm3008_setup_data *setup = socdev->codec_data; if (!codec) @@ -155,7 +155,7 @@ static int pcm3008_soc_remove(struct platform_device *pdev) pcm3008_gpio_free(setup); snd_soc_free_pcms(socdev); - kfree(socdev->codec); + kfree(socdev->card->codec); return 0; } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cac373616768..87f606c76822 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]), SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]), }; -/* add non dapm controls */ -static int ssm2602_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0), @@ -291,7 +276,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; @@ -336,7 +321,7 @@ static int ssm2602_startup(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; @@ -373,7 +358,7 @@ static int ssm2602_pcm_prepare(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* set active */ ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC); @@ -385,7 +370,7 @@ static void ssm2602_shutdown(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) @@ -521,6 +506,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, #define SSM2602_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 ssm2602_dai_ops = { + .startup = ssm2602_startup, + .prepare = ssm2602_pcm_prepare, + .hw_params = ssm2602_hw_params, + .shutdown = ssm2602_shutdown, + .digital_mute = ssm2602_mute, + .set_sysclk = ssm2602_set_dai_sysclk, + .set_fmt = ssm2602_set_dai_fmt, +}; + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -535,22 +530,14 @@ struct snd_soc_dai ssm2602_dai = { .channels_max = 2, .rates = SSM2602_RATES, .formats = SSM2602_FORMATS,}, - .ops = { - .startup = ssm2602_startup, - .prepare = ssm2602_pcm_prepare, - .hw_params = ssm2602_hw_params, - .shutdown = ssm2602_shutdown, - .digital_mute = ssm2602_mute, - .set_sysclk = ssm2602_set_dai_sysclk, - .set_fmt = ssm2602_set_dai_fmt, - } + .ops = &ssm2602_dai_ops, }; EXPORT_SYMBOL_GPL(ssm2602_dai); static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -559,7 +546,7 @@ static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state) static int ssm2602_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -581,7 +568,7 @@ static int ssm2602_resume(struct platform_device *pdev) */ static int ssm2602_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "SSM2602"; @@ -622,7 +609,8 @@ static int ssm2602_init(struct snd_soc_device *socdev) APANA_ENABLE_MIC_BOOST); ssm2602_write(codec, SSM2602_PWR, 0); - ssm2602_add_controls(codec); + snd_soc_add_controls(codec, ssm2602_snd_controls, + ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -653,7 +641,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = ssm2602_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -747,7 +735,7 @@ static int ssm2602_probe(struct platform_device *pdev) } codec->private_data = ssm2602; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -768,7 +756,7 @@ static int ssm2602_probe(struct platform_device *pdev) static int ssm2602_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index cfdea007c4cb..c3f4afb5d017 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = { SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph), }; -/* add non dapm controls */ -static int tlv320aic23_add_controls(struct snd_soc_codec *codec) -{ - - int err, i; - - for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&tlv320aic23_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; - -} - /* PGA Mixer controls for Line and Mic switch */ static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0), @@ -423,7 +405,7 @@ static int tlv320aic23_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 iface_reg; int ret; struct aic23 *aic23 = container_of(codec, struct aic23, codec); @@ -471,7 +453,7 @@ static int tlv320aic23_pcm_prepare(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* set active */ tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001); @@ -484,7 +466,7 @@ static void tlv320aic23_shutdown(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic23 *aic23 = container_of(codec, struct aic23, codec); /* deactivate */ @@ -598,6 +580,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, #define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops tlv320aic23_dai_ops = { + .prepare = tlv320aic23_pcm_prepare, + .hw_params = tlv320aic23_hw_params, + .shutdown = tlv320aic23_shutdown, + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, +}; + struct snd_soc_dai tlv320aic23_dai = { .name = "tlv320aic23", .playback = { @@ -612,14 +603,7 @@ struct snd_soc_dai tlv320aic23_dai = { .channels_max = 2, .rates = AIC23_RATES, .formats = AIC23_FORMATS,}, - .ops = { - .prepare = tlv320aic23_pcm_prepare, - .hw_params = tlv320aic23_hw_params, - .shutdown = tlv320aic23_shutdown, - .digital_mute = tlv320aic23_mute, - .set_fmt = tlv320aic23_set_dai_fmt, - .set_sysclk = tlv320aic23_set_dai_sysclk, - } + .ops = &tlv320aic23_dai_ops, }; EXPORT_SYMBOL_GPL(tlv320aic23_dai); @@ -627,7 +611,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -638,7 +622,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev, static int tlv320aic23_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u16 reg; @@ -660,7 +644,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) */ static int tlv320aic23_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; u16 reg; @@ -718,7 +702,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1); - tlv320aic23_add_controls(codec); + snd_soc_add_controls(codec, tlv320aic23_snd_controls, + ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -746,7 +731,7 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { struct snd_soc_device *socdev = tlv320aic23_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) @@ -804,7 +789,7 @@ static int tlv320aic23_probe(struct platform_device *pdev) if (aic23 == NULL) return -ENOMEM; codec = &aic23->codec; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -823,7 +808,7 @@ static int tlv320aic23_probe(struct platform_device *pdev) static int tlv320aic23_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic23 *aic23 = container_of(codec, struct aic23, codec); if (codec->control_data) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 29f2f1a017fd..3387d9e736ea 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -130,7 +130,7 @@ static int aic26_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic26 *aic26 = codec->private_data; int fsref, divisor, wlen, pval, jval, dval, qval; u16 reg; @@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) #define AIC26_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) +static struct snd_soc_dai_ops aic26_dai_ops = { + .hw_params = aic26_hw_params, + .digital_mute = aic26_mute, + .set_sysclk = aic26_set_sysclk, + .set_fmt = aic26_set_fmt, +}; + struct snd_soc_dai aic26_dai = { .name = "tlv320aic26", .playback = { @@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = { .rates = AIC26_RATES, .formats = AIC26_FORMATS, }, - .ops = { - .hw_params = aic26_hw_params, - .digital_mute = aic26_mute, - .set_sysclk = aic26_set_sysclk, - .set_fmt = aic26_set_fmt, - }, + .ops = &aic26_dai_ops, }; EXPORT_SYMBOL_GPL(aic26_dai); @@ -322,9 +324,8 @@ static int aic26_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; - struct snd_kcontrol *kcontrol; struct aic26 *aic26; - int i, ret, err; + int ret, err; dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n"); dev_dbg(&pdev->dev, "socdev=%p\n", socdev); @@ -338,7 +339,7 @@ static int aic26_probe(struct platform_device *pdev) return -ENODEV; } codec = &aic26->codec; - socdev->codec = codec; + socdev->card->codec = codec; dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n", &pdev->dev, socdev->dev); @@ -351,11 +352,9 @@ static int aic26_probe(struct platform_device *pdev) /* register controls */ dev_dbg(&pdev->dev, "Registering controls\n"); - for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) { - kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL); - err = snd_ctl_add(codec->card, kcontrol); - WARN_ON(err < 0); - } + err = snd_soc_add_controls(codec, aic26_snd_controls, + ARRAY_SIZE(aic26_snd_controls)); + WARN_ON(err < 0); /* CODEC is setup, we can register the card now */ dev_dbg(&pdev->dev, "Registering card\n"); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b47a749c5ea2..ab099f482487 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -45,6 +45,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "tlv320aic3x.h" @@ -165,10 +166,13 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0x0f; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0x01; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; unsigned short val, val_mask; int ret; struct snd_soc_dapm_path *path; @@ -247,56 +251,86 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +/* + * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0); +/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); +/* + * Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB. + * Step size is approximately 0.5 dB over most of the scale but increasing + * near the very low levels. + * Define dB scale so that it is mostly correct for range about -55 to 0 dB + * but having increasing dB difference below that (and where it doesn't count + * so much). This setting shows -50 dB (actual is -50.3 dB) for register + * value 100 and -58.5 dB (actual is -78.3 dB) for register value 117. + */ +static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ - SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R_TLV("PCM Playback Volume", + LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv), - SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_RLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R_TLV("Line DAC Playback Volume", + DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, - DACR1_2_LLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - 0, 0x7f, 1), - SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, - 0, 0x7f, 1), - SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, - LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, - LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), - - SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, - DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R_TLV("LineL DAC Playback Volume", + DACL1_2_LLOPM_VOL, DACR1_2_LLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("LineL Left PGA Bypass Playback Volume", + PGAL_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("LineR Right PGA Bypass Playback Volume", + PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("LineL Line2 Bypass Playback Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_LLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("LineR Line2 Bypass Playback Volume", + LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("Mono DAC Playback Volume", + DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, - PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, - LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1), - - SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, - DACR1_2_HPROUT_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R_TLV("Mono PGA Bypass Playback Volume", + PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("Mono Line2 Bypass Playback Volume", + LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP DAC Playback Volume", + DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - 0, 0x7f, 1), - SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, - 0, 0x7f, 1), - SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, - LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), - - SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, - DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R_TLV("HP Right PGA Bypass Playback Volume", + PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("HPL PGA Bypass Playback Volume", + PGAL_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("HPR PGA Bypass Playback Volume", + PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HP Line2 Bypass Playback Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume", + DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), - SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - 0, 0x7f, 1), - SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, - 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, - LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_SINGLE_TLV("HPLCOM PGA Bypass Playback Volume", + PGAL_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_SINGLE_TLV("HPRCOM PGA Bypass Playback Volume", + PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Playback Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), /* * Note: enable Automatic input Gain Controller with care. It can @@ -305,28 +339,13 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), /* Input */ - SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0), + SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, + 0, 119, 0, adc_tlv), SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; -/* add non dapm controls */ -static int aic3x_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&aic3x_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); @@ -743,7 +762,7 @@ static int aic3x_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_priv *aic3x = codec->private_data; int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0; u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1; @@ -1069,6 +1088,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed); #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops aic3x_dai_ops = { + .hw_params = aic3x_hw_params, + .digital_mute = aic3x_mute, + .set_sysclk = aic3x_set_dai_sysclk, + .set_fmt = aic3x_set_dai_fmt, +}; + struct snd_soc_dai aic3x_dai = { .name = "tlv320aic3x", .playback = { @@ -1083,19 +1109,14 @@ struct snd_soc_dai aic3x_dai = { .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, - .ops = { - .hw_params = aic3x_hw_params, - .digital_mute = aic3x_mute, - .set_sysclk = aic3x_set_dai_sysclk, - .set_fmt = aic3x_set_dai_fmt, - } + .ops = &aic3x_dai_ops, }; EXPORT_SYMBOL_GPL(aic3x_dai); static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1105,7 +1126,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state) static int aic3x_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u8 *cache = codec->reg_cache; @@ -1128,7 +1149,7 @@ static int aic3x_resume(struct platform_device *pdev) */ static int aic3x_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct aic3x_setup_data *setup = socdev->codec_data; int reg, ret = 0; @@ -1224,7 +1245,8 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4); aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4); - aic3x_add_controls(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) { @@ -1258,7 +1280,7 @@ 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->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -1363,7 +1385,7 @@ static int aic3x_probe(struct platform_device *pdev) } codec->private_data = aic3x; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1389,7 +1411,7 @@ static int aic3x_probe(struct platform_device *pdev) static int aic3x_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* power down chip */ if (codec->control_data) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index fd0f338374a7..97738e2ece04 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -42,7 +42,7 @@ */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x93, /* REG_CODEC_MODE (0x1) */ + 0x91, /* REG_CODEC_MODE (0x1) */ 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ @@ -117,6 +117,13 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_MISC_SET_2 (0x49) */ }; +/* codec private data */ +struct twl4030_priv { + unsigned int bypass_state; + unsigned int codec_powered; + unsigned int codec_muted; +}; + /* * read twl4030 register cache */ @@ -125,6 +132,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; + if (reg >= TWL4030_CACHEREGNUM) + return -EIO; + return cache[reg]; } @@ -151,26 +161,22 @@ static int twl4030_write(struct snd_soc_codec *codec, return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); } -static void twl4030_clear_codecpdz(struct snd_soc_codec *codec) +static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) { + struct twl4030_priv *twl4030 = codec->private_data; u8 mode; - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, - mode & ~TWL4030_CODECPDZ); - - /* REVISIT: this delay is present in TI sample drivers */ - /* but there seems to be no TRM requirement for it */ - udelay(10); -} - -static void twl4030_set_codecpdz(struct snd_soc_codec *codec) -{ - u8 mode; + if (enable == twl4030->codec_powered) + return; mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, - mode | TWL4030_CODECPDZ); + if (enable) + mode |= TWL4030_CODECPDZ; + else + mode &= ~TWL4030_CODECPDZ; + + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030->codec_powered = enable; /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -182,7 +188,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) int i; /* clear CODECPDZ prior to setting register defaults */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) @@ -190,6 +196,122 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) +{ + struct twl4030_priv *twl4030 = codec->private_data; + u8 reg_val; + + if (mute == twl4030->codec_muted) + 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_PRECKL_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; + twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); + } + + twl4030->codec_muted = mute; +} + +static void twl4030_power_up(struct snd_soc_codec *codec) +{ + struct twl4030_priv *twl4030 = codec->private_data; + u8 anamicl, regmisc1, byte; + int i = 0; + + if (twl4030->codec_powered) + return; + + /* set CODECPDZ to turn on codec */ + twl4030_codec_enable(codec, 1); + + /* initiate offset cancellation */ + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_write(codec, TWL4030_REG_ANAMICL, + anamicl | TWL4030_CNCL_OFFSET_START); + + /* wait for offset cancellation to complete */ + do { + /* this takes a little while, so don't slam i2c */ + udelay(2000); + twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_ANAMICL); + } while ((i++ < 100) && + ((byte & TWL4030_CNCL_OFFSET_START) == + TWL4030_CNCL_OFFSET_START)); + + /* Make sure that the reg_cache has the same value as the HW */ + twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte); + + /* anti-pop when changing analog gain */ + regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); + + /* toggle CODECPDZ as per TRM */ + twl4030_codec_enable(codec, 0); + twl4030_codec_enable(codec, 1); +} + +/* + * Unconditional power down + */ +static void twl4030_power_down(struct snd_soc_codec *codec) +{ + /* power down */ + twl4030_codec_enable(codec, 0); +} + /* Earpiece */ static const char *twl4030_earpiece_texts[] = {"Off", "DACL1", "DACL2", "DACR1"}; @@ -197,7 +319,7 @@ static const char *twl4030_earpiece_texts[] = static const unsigned int twl4030_earpiece_values[] = {0x0, 0x1, 0x2, 0x4}; -static const struct soc_value_enum twl4030_earpiece_enum = +static const struct soc_enum twl4030_earpiece_enum = SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, ARRAY_SIZE(twl4030_earpiece_texts), twl4030_earpiece_texts, @@ -213,7 +335,7 @@ static const char *twl4030_predrivel_texts[] = static const unsigned int twl4030_predrivel_values[] = {0x0, 0x1, 0x2, 0x4}; -static const struct soc_value_enum twl4030_predrivel_enum = +static const struct soc_enum twl4030_predrivel_enum = SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, ARRAY_SIZE(twl4030_predrivel_texts), twl4030_predrivel_texts, @@ -229,7 +351,7 @@ static const char *twl4030_predriver_texts[] = static const unsigned int twl4030_predriver_values[] = {0x0, 0x1, 0x2, 0x4}; -static const struct soc_value_enum twl4030_predriver_enum = +static const struct soc_enum twl4030_predriver_enum = SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, ARRAY_SIZE(twl4030_predriver_texts), twl4030_predriver_texts, @@ -317,7 +439,7 @@ static const char *twl4030_analoglmic_texts[] = static const unsigned int twl4030_analoglmic_values[] = {0x0, 0x1, 0x2, 0x4, 0x8}; -static const struct soc_value_enum twl4030_analoglmic_enum = +static const struct soc_enum twl4030_analoglmic_enum = SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf, ARRAY_SIZE(twl4030_analoglmic_texts), twl4030_analoglmic_texts, @@ -333,7 +455,7 @@ static const char *twl4030_analogrmic_texts[] = static const unsigned int twl4030_analogrmic_values[] = {0x0, 0x1, 0x4}; -static const struct soc_value_enum twl4030_analogrmic_enum = +static const struct soc_enum twl4030_analogrmic_enum = SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5, ARRAY_SIZE(twl4030_analogrmic_texts), twl4030_analogrmic_texts, @@ -366,6 +488,41 @@ static const struct soc_enum twl4030_micpathtx2_enum = static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control = SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum); +/* Analog bypass for AudioR1 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassr1_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR1_APGA_CTL, 2, 1, 0); + +/* Analog bypass for AudioL1 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassl1_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL1_APGA_CTL, 2, 1, 0); + +/* Analog bypass for AudioR2 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR2_APGA_CTL, 2, 1, 0); + +/* Analog bypass for AudioL2 */ +static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control = + SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0); + +/* Digital bypass gain, 0 mutes the bypass */ +static const unsigned int twl4030_dapm_dbypass_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1), + 4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0), +}; + +/* Digital bypass left (TX1L -> RX2L) */ +static const struct snd_kcontrol_new twl4030_dapm_dbypassl_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_ATX2ARXPGA, 3, 7, 0, + twl4030_dapm_dbypass_tlv); + +/* Digital bypass right (TX1R -> RX2R) */ +static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control = + SOC_DAPM_SINGLE_TLV("Volume", + TWL4030_REG_ATX2ARXPGA, 0, 7, 0, + twl4030_dapm_dbypass_tlv); + static int micpath_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -420,6 +577,79 @@ static int handsfree_event(struct snd_soc_dapm_widget *w, return 0; } +static int headsetl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + unsigned char hs_gain, hs_pop; + + /* Save the current volume */ + hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET); + hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Do the anti-pop/bias ramp enable according to the TRM */ + hs_pop |= TWL4030_VMID_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Is this needed? Can we just use whatever gain here? */ + twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, + (hs_gain & (~0x0f)) | 0x0a); + hs_pop |= TWL4030_RAMP_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + + /* Restore the original volume */ + twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + break; + case SND_SOC_DAPM_POST_PMD: + /* Do the anti-pop/bias ramp disable according to the TRM */ + hs_pop &= ~TWL4030_RAMP_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + /* Bypass the reg_cache to mute the headset */ + twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain & (~0x0f), + TWL4030_REG_HS_GAIN_SET); + hs_pop &= ~TWL4030_VMID_EN; + twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop); + break; + } + return 0; +} + +static int bypass_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_mixer_control *m = + (struct soc_mixer_control *)w->kcontrols->private_value; + struct twl4030_priv *twl4030 = w->codec->private_data; + unsigned char reg; + + reg = twl4030_read_reg_cache(w->codec, m->reg); + + if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { + /* Analog bypass */ + if (reg & (1 << m->shift)) + twl4030->bypass_state |= + (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + else + twl4030->bypass_state &= + ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); + } else { + /* Digital bypass */ + if (reg & (0x7 << m->shift)) + twl4030->bypass_state |= (1 << (m->shift ? 5 : 4)); + else + twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); + } + + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { + if (twl4030->bypass_state) + twl4030_codec_mute(w->codec, 0); + else + twl4030_codec_mute(w->codec, 1); + } + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -614,6 +844,17 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); */ static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); +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", + "3495/2581/1748 ms" +}; + +static const struct soc_enum twl4030_rampdelay_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2, + ARRAY_SIZE(twl4030_rampdelay_texts), + twl4030_rampdelay_texts); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -668,23 +909,9 @@ 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), -}; -/* add non dapm controls */ -static int twl4030_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&twl4030_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} + SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum), +}; static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* Left channel inputs */ @@ -714,13 +941,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", - TWL4030_REG_AVDAC_CTL, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", - TWL4030_REG_AVDAC_CTL, 1, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", - TWL4030_REG_AVDAC_CTL, 2, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", - TWL4030_REG_AVDAC_CTL, 3, 0), + SND_SOC_NOPM, 0, 0), /* Analog PGAs */ SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, @@ -732,6 +959,37 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + /* Analog bypasses */ + SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr1_control, bypass_event, + SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl1_control, + bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr2_control, + bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl2_control, + bypass_event, SND_SOC_DAPM_POST_REG), + + /* Digital bypasses */ + SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassl_control, bypass_event, + SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassr_control, bypass_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, + 3, 0, NULL, 0), + /* Output MUX controls */ /* Earpiece */ SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, @@ -742,8 +1000,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_predriver_control), /* HeadsetL/R */ - SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control), + SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_control, headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsor_control), /* CarkitL/R */ @@ -782,16 +1041,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), - /* Analog input muxes with power switch for the physical ADCL/R */ + /* Analog input muxes with switch for the capture amplifiers */ SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", - TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control), + TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control), SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", - TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control), + TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control), - SND_SOC_DAPM_PGA("Analog Left Amplifier", - TWL4030_REG_ANAMICL, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Analog Right Amplifier", - TWL4030_REG_ANAMICR, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC Physical Left", + TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC Physical Right", + TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Digimic0 Enable", TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), @@ -801,13 +1060,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0), + }; static const struct snd_soc_dapm_route intercon[] = { - {"ARXL1_APGA", NULL, "DAC Left1"}, - {"ARXR1_APGA", NULL, "DAC Right1"}, - {"ARXL2_APGA", NULL, "DAC Left2"}, - {"ARXR2_APGA", NULL, "DAC Right2"}, + {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, + {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, + {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, + {"Analog R2 Playback Mixer", NULL, "DAC Right2"}, + + {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"}, + {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"}, + {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"}, + {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"}, /* Internal playback routings */ /* Earpiece */ @@ -865,23 +1130,23 @@ static const struct snd_soc_dapm_route intercon[] = { {"Analog Right Capture Route", "Sub mic", "SUBMIC"}, {"Analog Right Capture Route", "AUXR", "AUXR"}, - {"Analog Left Amplifier", NULL, "Analog Left Capture Route"}, - {"Analog Right Amplifier", NULL, "Analog Right Capture Route"}, + {"ADC Physical Left", NULL, "Analog Left Capture Route"}, + {"ADC Physical Right", NULL, "Analog Right Capture Route"}, {"Digimic0 Enable", NULL, "DIGIMIC0"}, {"Digimic1 Enable", NULL, "DIGIMIC1"}, /* TX1 Left capture path */ - {"TX1 Capture Route", "Analog", "Analog Left Amplifier"}, + {"TX1 Capture Route", "Analog", "ADC Physical Left"}, {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, /* TX1 Right capture path */ - {"TX1 Capture Route", "Analog", "Analog Right Amplifier"}, + {"TX1 Capture Route", "Analog", "ADC Physical Right"}, {"TX1 Capture Route", "Digimic0", "Digimic0 Enable"}, /* TX2 Left capture path */ - {"TX2 Capture Route", "Analog", "Analog Left Amplifier"}, + {"TX2 Capture Route", "Analog", "ADC Physical Left"}, {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"}, /* TX2 Right capture path */ - {"TX2 Capture Route", "Analog", "Analog Right Amplifier"}, + {"TX2 Capture Route", "Analog", "ADC Physical Right"}, {"TX2 Capture Route", "Digimic1", "Digimic1 Enable"}, {"ADC Virtual Left1", NULL, "TX1 Capture Route"}, @@ -889,6 +1154,24 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"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"}, + + {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, + {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, + {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, + {"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"}, + + /* Digital bypass routes */ + {"Right Digital Loopback", "Volume", "TX1 Capture Route"}, + {"Left Digital Loopback", "Volume", "TX1 Capture Route"}, + + {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"}, + {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"}, + }; static int twl4030_add_widgets(struct snd_soc_codec *codec) @@ -902,82 +1185,28 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) return 0; } -static void twl4030_power_up(struct snd_soc_codec *codec) -{ - u8 anamicl, regmisc1, byte, popn; - int i = 0; - - /* set CODECPDZ to turn on codec */ - twl4030_set_codecpdz(codec); - - /* initiate offset cancellation */ - anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); - twl4030_write(codec, TWL4030_REG_ANAMICL, - anamicl | TWL4030_CNCL_OFFSET_START); - - - /* wait for offset cancellation to complete */ - do { - /* this takes a little while, so don't slam i2c */ - udelay(2000); - twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, - TWL4030_REG_ANAMICL); - } while ((i++ < 100) && - ((byte & TWL4030_CNCL_OFFSET_START) == - TWL4030_CNCL_OFFSET_START)); - - /* anti-pop when changing analog gain */ - regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); - twl4030_write(codec, TWL4030_REG_MISC_SET_1, - regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); - - /* toggle CODECPDZ as per TRM */ - twl4030_clear_codecpdz(codec); - twl4030_set_codecpdz(codec); - - /* program anti-pop with bias ramp delay */ - popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - popn &= TWL4030_RAMP_DELAY; - popn |= TWL4030_RAMP_DELAY_645MS; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - popn |= TWL4030_VMID_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - - /* enable anti-pop ramp */ - popn |= TWL4030_RAMP_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); -} - -static void twl4030_power_down(struct snd_soc_codec *codec) -{ - u8 popn; - - /* disable anti-pop ramp */ - popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); - popn &= ~TWL4030_RAMP_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - - /* disable bias out */ - popn &= ~TWL4030_VMID_EN; - twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - - /* power down */ - twl4030_clear_codecpdz(codec); -} - static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct twl4030_priv *twl4030 = codec->private_data; + switch (level) { case SND_SOC_BIAS_ON: - twl4030_power_up(codec); + twl4030_codec_mute(codec, 0); break; case SND_SOC_BIAS_PREPARE: - /* TODO: develop a twl4030_prepare function */ + twl4030_power_up(codec); + if (twl4030->bypass_state) + twl4030_codec_mute(codec, 0); + else + twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_STANDBY: - /* TODO: develop a twl4030_standby function */ - twl4030_power_down(codec); + twl4030_power_up(codec); + if (twl4030->bypass_state) + twl4030_codec_mute(codec, 0); + else + twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); @@ -994,10 +1223,9 @@ static int twl4030_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u8 mode, old_mode, format, old_format; - /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; @@ -1039,8 +1267,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, if (mode != old_mode) { /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); } /* sample size */ @@ -1063,13 +1292,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, if (format != old_format) { /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); /* change format */ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); /* set CODECPDZ afterwards */ - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); } return 0; } @@ -1139,13 +1368,13 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, if (format != old_format) { /* clear CODECPDZ before changing format (codec requirement) */ - twl4030_clear_codecpdz(codec); + twl4030_codec_enable(codec, 0); /* change format */ twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); /* set CODECPDZ afterwards */ - twl4030_set_codecpdz(codec); + twl4030_codec_enable(codec, 1); } return 0; @@ -1154,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) +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, +}; + struct snd_soc_dai twl4030_dai = { .name = "twl4030", .playback = { @@ -1168,18 +1403,14 @@ struct snd_soc_dai twl4030_dai = { .channels_max = 2, .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, - .ops = { - .hw_params = twl4030_hw_params, - .set_sysclk = twl4030_set_dai_sysclk, - .set_fmt = twl4030_set_dai_fmt, - } + .ops = &twl4030_dai_ops, }; EXPORT_SYMBOL_GPL(twl4030_dai); static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1189,7 +1420,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) static int twl4030_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); twl4030_set_bias_level(codec, codec->suspend_bias_level); @@ -1203,7 +1434,7 @@ static int twl4030_resume(struct platform_device *pdev) static int twl4030_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; printk(KERN_INFO "TWL4030 Audio Codec init \n"); @@ -1233,7 +1464,8 @@ static int twl4030_init(struct snd_soc_device *socdev) /* power on device */ twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - twl4030_add_controls(codec); + snd_soc_add_controls(codec, twl4030_snd_controls, + ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); ret = snd_soc_init_card(socdev); @@ -1258,12 +1490,20 @@ static int twl4030_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct twl4030_priv *twl4030; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); + if (twl4030 == NULL) { + kfree(codec); + return -ENOMEM; + } + + codec->private_data = twl4030; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1277,9 +1517,13 @@ static int twl4030_probe(struct platform_device *pdev) static int twl4030_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec->private_data); kfree(codec); return 0; diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 442e5a828617..33dbb144dad1 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -170,6 +170,9 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* EAR_CTL (0x21) */ +#define TWL4030_EAR_GAIN 0x30 + /* HS_GAIN_SET (0x23) Fields */ #define TWL4030_HSR_GAIN 0x0C @@ -198,6 +201,18 @@ #define TWL4030_RAMP_DELAY_2581MS 0x1C #define TWL4030_RAMP_EN 0x02 +/* PREDL_CTL (0x25) */ +#define TWL4030_PREDL_GAIN 0x30 + +/* PREDR_CTL (0x26) */ +#define TWL4030_PREDR_GAIN 0x30 + +/* PRECKL_CTL (0x27) */ +#define TWL4030_PRECKL_GAIN 0x30 + +/* PRECKR_CTL (0x28) */ +#define TWL4030_PRECKR_GAIN 0x30 + /* HFL_CTL (0x29, 0x2A) Fields */ #define TWL4030_HF_CTL_HB_EN 0x04 #define TWL4030_HF_CTL_LOOP_EN 0x08 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index a2c5064a774b..ddefb8f80145 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -173,7 +173,7 @@ static int uda134x_startup(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct uda134x_priv *uda134x = codec->private_data; struct snd_pcm_runtime *master_runtime; @@ -206,7 +206,7 @@ static void uda134x_shutdown(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct uda134x_priv *uda134x = codec->private_data; if (uda134x->master_substream == substream) @@ -221,7 +221,7 @@ static int uda134x_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct uda134x_priv *uda134x = codec->private_data; u8 hw_params; @@ -431,38 +431,14 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; -static int uda134x_add_controls(struct snd_soc_codec *codec) -{ - int err, i, n; - const struct snd_kcontrol_new *ctrls; - struct uda134x_platform_data *pd = codec->control_data; - - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - n = ARRAY_SIZE(uda1340_snd_controls); - ctrls = uda1340_snd_controls; - break; - case UDA134X_UDA1341: - n = ARRAY_SIZE(uda1341_snd_controls); - ctrls = uda1341_snd_controls; - break; - default: - printk(KERN_ERR "%s unkown codec type: %d", - __func__, pd->model); - return -EINVAL; - } - - for (i = 0; i < n; i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&ctrls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} +static struct snd_soc_dai_ops uda134x_dai_ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, +}; struct snd_soc_dai uda134x_dai = { .name = "UDA134X", @@ -483,14 +459,7 @@ struct snd_soc_dai uda134x_dai = { .formats = UDA134X_FORMATS, }, /* pcm operations */ - .ops = { - .startup = uda134x_startup, - .shutdown = uda134x_shutdown, - .hw_params = uda134x_hw_params, - .digital_mute = uda134x_mute, - .set_sysclk = uda134x_set_dai_sysclk, - .set_fmt = uda134x_set_dai_fmt, - } + .ops = &uda134x_dai_ops, }; EXPORT_SYMBOL(uda134x_dai); @@ -525,11 +494,11 @@ static int uda134x_soc_probe(struct platform_device *pdev) return -EINVAL; } - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->card->codec == NULL) return ret; - codec = socdev->codec; + codec = socdev->card->codec; uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); if (uda134x == NULL) @@ -572,7 +541,22 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = uda134x_add_controls(codec); + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + ret = snd_soc_add_controls(codec, uda1340_snd_controls, + ARRAY_SIZE(uda1340_snd_controls)); + break; + case UDA134X_UDA1341: + ret = snd_soc_add_controls(codec, uda1341_snd_controls, + ARRAY_SIZE(uda1341_snd_controls)); + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + if (ret < 0) { printk(KERN_ERR "UDA134X: failed to register controls\n"); goto pcm_err; @@ -602,7 +586,7 @@ priv_err: static int uda134x_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -622,7 +606,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -632,7 +616,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev, static int uda134x_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index e6bf0844fbf3..5b21594e0e58 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -25,6 +25,7 @@ #include <linux/ioctl.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> @@ -35,7 +36,8 @@ #include "uda1380.h" -#define UDA1380_VERSION "0.6" +static struct work_struct uda1380_work; +static struct snd_soc_codec *uda1380_codec; /* * uda1380 register cache @@ -52,6 +54,8 @@ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { 0x0000, 0x8000, 0x0002, 0x0000, }; +static unsigned long uda1380_cache_dirty; + /* * read uda1380 register cache */ @@ -73,8 +77,11 @@ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { u16 *cache = codec->reg_cache; + if (reg >= UDA1380_CACHEREGNUM) return; + if ((reg >= 0x10) && (cache[reg] != value)) + set_bit(reg - 0x10, &uda1380_cache_dirty); cache[reg] = value; } @@ -113,6 +120,8 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, (data[0]<<8) | data[1]); return -EIO; } + if (reg >= 0x10) + clear_bit(reg - 0x10, &uda1380_cache_dirty); return 0; } else return -EIO; @@ -120,6 +129,20 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, #define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) +static void uda1380_flush_work(struct work_struct *work) +{ + int bit, reg; + + for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) { + reg = 0x10 + bit; + pr_debug("uda1380: flush reg %x val %x:\n", reg, + uda1380_read_reg_cache(uda1380_codec, reg)); + uda1380_write(uda1380_codec, reg, + uda1380_read_reg_cache(uda1380_codec, reg)); + clear_bit(bit, &uda1380_cache_dirty); + } +} + /* declarations of ALSA reg_elem_REAL controls */ static const char *uda1380_deemp[] = { "None", @@ -254,7 +277,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0), /* DA_POL_INV */ SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum), /* SEL_NS */ SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum), /* MIX_POS, MIX */ - SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0), /* SILENCE, force DAC output to silence */ SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0), /* SDET_ON */ SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum), /* SD_VALUE */ SOC_ENUM("Oversampling Input", uda1380_os_enum), /* OS */ @@ -271,21 +293,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = { SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), }; -/* add non dapm controls */ -static int uda1380_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Input mux */ static const struct snd_kcontrol_new uda1380_input_mux_control = SOC_DAPM_ENUM("Route", uda1380_input_sel_enum); @@ -371,7 +378,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) return 0; } -static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, +static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; @@ -381,61 +388,107 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK); - /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: iface |= R01_SFORI_I2S | R01_SFORO_I2S; break; case SND_SOC_DAIFMT_LSB: - iface |= R01_SFORI_LSB16 | R01_SFORO_I2S; + iface |= R01_SFORI_LSB16 | R01_SFORO_LSB16; break; case SND_SOC_DAIFMT_MSB: - iface |= R01_SFORI_MSB | R01_SFORO_I2S; + iface |= R01_SFORI_MSB | R01_SFORO_MSB; } - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) - iface |= R01_SIM; + /* DATAI is slave only, so in single-link mode, this has to be slave */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; uda1380_write(codec, UDA1380_IFACE, iface); return 0; } -/* - * Flush reg cache - * We can only write the interpolator and decimator registers - * when the DAI is being clocked by the CPU DAI. It's up to the - * machine and cpu DAI driver to do this before we are called. - */ -static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai, + unsigned int fmt) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; - int reg, reg_start, reg_end, clk; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - reg_start = UDA1380_MVOL; - reg_end = UDA1380_MIXER; - } else { - reg_start = UDA1380_DEC; - reg_end = UDA1380_AGC; + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~R01_SFORI_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORI_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORI_LSB16; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORI_MSB; } - /* FIXME disable DAC_CLK */ - clk = uda1380_read_reg_cache(codec, UDA1380_CLK); - uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK); + /* DATAI is slave only, so this has to be slave */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int iface; + + /* set up DAI based upon fmt */ + iface = uda1380_read_reg_cache(codec, UDA1380_IFACE); + iface &= ~(R01_SIM | R01_SFORO_MASK); - for (reg = reg_start; reg <= reg_end; reg++) { - pr_debug("uda1380: flush reg %x val %x:", reg, - uda1380_read_reg_cache(codec, reg)); - uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg)); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= R01_SFORO_I2S; + break; + case SND_SOC_DAIFMT_LSB: + iface |= R01_SFORO_LSB16; + break; + case SND_SOC_DAIFMT_MSB: + iface |= R01_SFORO_MSB; } - /* FIXME enable DAC_CLK */ - uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK); + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + iface |= R01_SIM; + uda1380_write(codec, UDA1380_IFACE, iface); + + return 0; +} + +static int uda1380_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + uda1380_write_reg_cache(codec, UDA1380_MIXER, + mixer & ~R14_SILENCE); + 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); + break; + } return 0; } @@ -445,7 +498,7 @@ static int uda1380_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* set WSPLL power and divider if running from this clock */ @@ -484,7 +537,7 @@ static void uda1380_pcm_shutdown(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ @@ -501,24 +554,6 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, uda1380_write(codec, UDA1380_CLK, clk); } -static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute) -{ - struct snd_soc_codec *codec = codec_dai->codec; - u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM; - - /* FIXME: mute(codec,0) is called when the magician clock is already - * set to WSPLL, but for some unknown reason writing to interpolator - * registers works only when clocked by SYSCLK */ - u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); - uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk); - if (mute) - uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM); - else - uda1380_write(codec, UDA1380_DEEMP, mute_reg); - uda1380_write(codec, UDA1380_CLK, clk); - return 0; -} - static int uda1380_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -544,6 +579,27 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops uda1380_dai_ops = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .trigger = uda1380_trigger, + .set_fmt = uda1380_set_dai_fmt_both, +}; + +static struct snd_soc_dai_ops uda1380_dai_ops_playback = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .trigger = uda1380_trigger, + .set_fmt = uda1380_set_dai_fmt_playback, +}; + +static struct snd_soc_dai_ops uda1380_dai_ops_capture = { + .hw_params = uda1380_pcm_hw_params, + .shutdown = uda1380_pcm_shutdown, + .trigger = uda1380_trigger, + .set_fmt = uda1380_set_dai_fmt_capture, +}; + struct snd_soc_dai uda1380_dai[] = { { .name = "UDA1380", @@ -559,13 +615,7 @@ struct snd_soc_dai uda1380_dai[] = { .channels_max = 2, .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops, }, { /* playback only - dual interface */ .name = "UDA1380", @@ -576,13 +626,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .digital_mute = uda1380_mute, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops_playback, }, { /* capture only - dual interface*/ .name = "UDA1380", @@ -593,12 +637,7 @@ struct snd_soc_dai uda1380_dai[] = { .rates = UDA1380_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .hw_params = uda1380_pcm_hw_params, - .shutdown = uda1380_pcm_shutdown, - .prepare = uda1380_pcm_prepare, - .set_fmt = uda1380_set_dai_fmt, - }, + .ops = &uda1380_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(uda1380_dai); @@ -606,7 +645,7 @@ EXPORT_SYMBOL_GPL(uda1380_dai); static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -615,7 +654,7 @@ static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) static int uda1380_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -637,7 +676,7 @@ static int uda1380_resume(struct platform_device *pdev) */ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "UDA1380"; @@ -655,6 +694,9 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) codec->reg_cache_step = 1; uda1380_reset(codec); + uda1380_codec = codec; + INIT_WORK(&uda1380_work, uda1380_flush_work); + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -675,7 +717,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) } /* uda1380 init */ - uda1380_add_controls(codec); + 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) { @@ -702,7 +745,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c, { struct snd_soc_device *socdev = uda1380_socdev; struct uda1380_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -786,14 +829,12 @@ static int uda1380_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret; - pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION); - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -817,7 +858,7 @@ static int uda1380_probe(struct platform_device *pdev) static int uda1380_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e3989d406f54..3b1d0993bed9 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -3,7 +3,7 @@ * * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. * - * Author: Liam Girdwood <lg@opensource.wolfsonmicro.com> + * Author: Liam Girdwood <lrg@slimlogic.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 @@ -51,10 +51,17 @@ struct wm8350_output { u16 mute; }; +struct wm8350_jack_data { + struct snd_soc_jack *jack; + int report; +}; + struct wm8350_data { struct snd_soc_codec codec; struct wm8350_output out1; struct wm8350_output out2; + struct wm8350_jack_data hpl; + struct wm8350_jack_data hpr; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; @@ -775,21 +782,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Beep", NULL, "IN3R PGA"}, }; -static int wm8350_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8350_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static int wm8350_add_widgets(struct snd_soc_codec *codec) { int ret; @@ -1309,7 +1301,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, static int wm8350_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -1318,7 +1310,7 @@ static int wm8350_suspend(struct platform_device *pdev, pm_message_t state) static int wm8350_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1328,6 +1320,95 @@ static int wm8350_resume(struct platform_device *pdev) return 0; } +static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) +{ + struct wm8350_data *priv = data; + u16 reg; + int report; + int mask; + struct wm8350_jack_data *jack = NULL; + + switch (irq) { + case WM8350_IRQ_CODEC_JCK_DET_L: + jack = &priv->hpl; + mask = WM8350_JACK_L_LVL; + break; + + case WM8350_IRQ_CODEC_JCK_DET_R: + jack = &priv->hpr; + mask = WM8350_JACK_R_LVL; + break; + + default: + BUG(); + } + + if (!jack->jack) { + dev_warn(wm8350->dev, "Jack interrupt called with no jack\n"); + return; + } + + /* Debounce */ + msleep(200); + + reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS); + if (reg & mask) + report = jack->report; + else + report = 0; + + snd_soc_jack_report(jack->jack, report, jack->report); +} + +/** + * wm8350_hp_jack_detect - Enable headphone jack detection. + * + * @codec: WM8350 codec + * @which: left or right jack detect signal + * @jack: jack to report detection events on + * @report: value to report + * + * Enables the headphone jack detection of the WM8350. + */ +int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, + struct snd_soc_jack *jack, int report) +{ + struct wm8350_data *priv = codec->private_data; + struct wm8350 *wm8350 = codec->control_data; + int irq; + int ena; + + switch (which) { + case WM8350_JDL: + priv->hpl.jack = jack; + priv->hpl.report = report; + irq = WM8350_IRQ_CODEC_JCK_DET_L; + ena = WM8350_JDL_ENA; + break; + + case WM8350_JDR: + priv->hpr.jack = jack; + priv->hpr.report = report; + irq = WM8350_IRQ_CODEC_JCK_DET_R; + ena = WM8350_JDR_ENA; + break; + + default: + return -EINVAL; + } + + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); + + /* Sync status */ + wm8350_hp_jack_handler(wm8350, irq, priv); + + wm8350_unmask_irq(wm8350, irq); + + return 0; +} +EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); + static struct snd_soc_codec *wm8350_codec; static int wm8350_probe(struct platform_device *pdev) @@ -1342,8 +1423,8 @@ static int wm8350_probe(struct platform_device *pdev) BUG_ON(!wm8350_codec); - socdev->codec = wm8350_codec; - codec = socdev->codec; + socdev->card->codec = wm8350_codec; + codec = socdev->card->codec; wm8350 = codec->control_data; priv = codec->private_data; @@ -1381,13 +1462,21 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, WM8350_OUT2_VU | WM8350_OUT2R_MUTE); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + wm8350_hp_jack_handler, priv); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + wm8350_hp_jack_handler, priv); + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { dev_err(&pdev->dev, "failed to create pcms\n"); return ret; } - wm8350_add_controls(codec); + snd_soc_add_controls(codec, wm8350_snd_controls, + ARRAY_SIZE(wm8350_snd_controls)); wm8350_add_widgets(codec); wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1409,10 +1498,23 @@ card_err: static int wm8350_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *priv = codec->private_data; int ret; + wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, + WM8350_JDL_ENA | WM8350_JDR_ENA); + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); + + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + + priv->hpl.jack = NULL; + priv->hpr.jack = NULL; + /* cancel any work waiting to be queued. */ ret = cancel_delayed_work(&codec->delayed_work); @@ -1436,6 +1538,16 @@ static int wm8350_remove(struct platform_device *pdev) SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8350_dai_ops = { + .hw_params = wm8350_pcm_hw_params, + .digital_mute = wm8350_mute, + .trigger = wm8350_pcm_trigger, + .set_fmt = wm8350_set_dai_fmt, + .set_sysclk = wm8350_set_dai_sysclk, + .set_pll = wm8350_set_fll, + .set_clkdiv = wm8350_set_clkdiv, +}; + struct snd_soc_dai wm8350_dai = { .name = "WM8350", .playback = { @@ -1452,15 +1564,7 @@ struct snd_soc_dai wm8350_dai = { .rates = WM8350_RATES, .formats = WM8350_FORMATS, }, - .ops = { - .hw_params = wm8350_pcm_hw_params, - .digital_mute = wm8350_mute, - .trigger = wm8350_pcm_trigger, - .set_fmt = wm8350_set_dai_fmt, - .set_sysclk = wm8350_set_dai_sysclk, - .set_pll = wm8350_set_fll, - .set_clkdiv = wm8350_set_clkdiv, - }, + .ops = &wm8350_dai_ops, }; EXPORT_SYMBOL_GPL(wm8350_dai); @@ -1472,7 +1576,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8350 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350); -static int wm8350_codec_probe(struct platform_device *pdev) +static __devinit int wm8350_codec_probe(struct platform_device *pdev) { struct wm8350 *wm8350 = platform_get_drvdata(pdev); struct wm8350_data *priv; diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h index cc2887aa6c38..d11bd9288cf9 100644 --- a/sound/soc/codecs/wm8350.h +++ b/sound/soc/codecs/wm8350.h @@ -17,4 +17,12 @@ extern struct snd_soc_dai wm8350_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8350; +enum wm8350_jack { + WM8350_JDL = 1, + WM8350_JDR = 2, +}; + +int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, + struct snd_soc_jack *jack, int report); + #endif diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c new file mode 100644 index 000000000000..510efa604008 --- /dev/null +++ b/sound/soc/codecs/wm8400.c @@ -0,0 +1,1582 @@ +/* + * wm8400.c -- WM8400 ALSA Soc Audio driver + * + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/mfd/wm8400-audio.h> +#include <linux/mfd/wm8400-private.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 "wm8400.h" + +/* Fake register for internal state */ +#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) +#define WM8400_INMIXL_PWR 0 +#define WM8400_AINLMUX_PWR 1 +#define WM8400_INMIXR_PWR 2 +#define WM8400_AINRMUX_PWR 3 + +static struct regulator_bulk_data power[] = { + { + .supply = "I2S1VDD", + }, + { + .supply = "I2S2VDD", + }, + { + .supply = "DCVDD", + }, + { + .supply = "AVDD", + }, + { + .supply = "FLLVDD", + }, + { + .supply = "HPVDD", + }, + { + .supply = "SPKVDD", + }, +}; + +/* codec private data */ +struct wm8400_priv { + struct snd_soc_codec codec; + struct wm8400 *wm8400; + u16 fake_register; + unsigned int sysclk; + unsigned int pcmclk; + struct work_struct work; + int fll_in, fll_out; +}; + +static inline unsigned int wm8400_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct wm8400_priv *wm8400 = codec->private_data; + + if (reg == WM8400_INTDRIVBITS) + return wm8400->fake_register; + else + return wm8400_reg_read(wm8400->wm8400, reg); +} + +/* + * write to the wm8400 register space + */ +static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8400_priv *wm8400 = codec->private_data; + + if (reg == WM8400_INTDRIVBITS) { + wm8400->fake_register = value; + return 0; + } else + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); +} + +static void wm8400_codec_reset(struct snd_soc_codec *codec) +{ + struct wm8400_priv *wm8400 = codec->private_data; + + wm8400_reset_codec_reg_cache(wm8400->wm8400); +} + +static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600); + +static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000); + +static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0); + +static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600); + +static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0); + +static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0); + +static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763); + +static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0); + +static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int ret; + u16 val; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8400_read(codec, reg); + return wm8400_write(codec, reg, val | 0x0100); +} + +#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, 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 = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \ + .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } + + +static const char *wm8400_digital_sidetone[] = + {"None", "Left ADC", "Right ADC", "Reserved"}; + +static const struct soc_enum wm8400_left_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone); + +static const struct soc_enum wm8400_right_digital_sidetone_enum = +SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone); + +static const char *wm8400_adcmode[] = + {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; + +static const struct soc_enum wm8400_right_adcmode_enum = +SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode); + +static const struct snd_kcontrol_new wm8400_snd_controls[] = { +/* INMIXL */ +SOC_SINGLE("LIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L12MNBST_SHIFT, + 1, 0), +SOC_SINGLE("LIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L34MNBST_SHIFT, + 1, 0), +/* INMIXR */ +SOC_SINGLE("RIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R12MNBST_SHIFT, + 1, 0), +SOC_SINGLE("RIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R34MNBST_SHIFT, + 1, 0), + +/* LOMIX */ +SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER3, + WM8400_LLI3LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3, + WM8400_LR12LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3, + WM8400_LL12LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER5, + WM8400_LRI3LOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER5, + WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER5, + WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv), + +/* ROMIX */ +SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER4, + WM8400_RRI3ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4, + WM8400_RL12ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4, + WM8400_RR12ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER6, + WM8400_RLI3ROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER6, + WM8400_RLBROVOL_SHIFT, 7, 0, out_mix_tlv), +SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER6, + WM8400_RRBROVOL_SHIFT, 7, 0, out_mix_tlv), + +/* LOUT */ +WM8400_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8400_LEFT_OUTPUT_VOLUME, + WM8400_LOUTVOL_SHIFT, WM8400_LOUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOUT ZC", WM8400_LEFT_OUTPUT_VOLUME, WM8400_LOZC_SHIFT, 1, 0), + +/* ROUT */ +WM8400_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8400_RIGHT_OUTPUT_VOLUME, + WM8400_ROUTVOL_SHIFT, WM8400_ROUTVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROUT ZC", WM8400_RIGHT_OUTPUT_VOLUME, WM8400_ROZC_SHIFT, 1, 0), + +/* LOPGA */ +WM8400_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8400_LEFT_OPGA_VOLUME, + WM8400_LOPGAVOL_SHIFT, WM8400_LOPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("LOPGA ZC Switch", WM8400_LEFT_OPGA_VOLUME, + WM8400_LOPGAZC_SHIFT, 1, 0), + +/* ROPGA */ +WM8400_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8400_RIGHT_OPGA_VOLUME, + WM8400_ROPGAVOL_SHIFT, WM8400_ROPGAVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("ROPGA ZC Switch", WM8400_RIGHT_OPGA_VOLUME, + WM8400_ROPGAZC_SHIFT, 1, 0), + +SOC_SINGLE("LON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_LONMUTE_SHIFT, 1, 0), +SOC_SINGLE("LOP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_LOPMUTE_SHIFT, 1, 0), +SOC_SINGLE("LOP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_LOATTN_SHIFT, 1, 0), +SOC_SINGLE("RON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_RONMUTE_SHIFT, 1, 0), +SOC_SINGLE("ROP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_ROPMUTE_SHIFT, 1, 0), +SOC_SINGLE("ROP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME, + WM8400_ROATTN_SHIFT, 1, 0), + +SOC_SINGLE("OUT3 Mute Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT3MUTE_SHIFT, 1, 0), +SOC_SINGLE("OUT3 Attenuation Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT3ATTN_SHIFT, 1, 0), + +SOC_SINGLE("OUT4 Mute Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT4MUTE_SHIFT, 1, 0), +SOC_SINGLE("OUT4 Attenuation Switch", WM8400_OUT3_4_VOLUME, + WM8400_OUT4ATTN_SHIFT, 1, 0), + +SOC_SINGLE("Speaker Mode Switch", WM8400_CLASSD1, + WM8400_CDMODE_SHIFT, 1, 0), + +SOC_SINGLE("Speaker Output Attenuation Volume", WM8400_SPEAKER_VOLUME, + WM8400_SPKATTN_SHIFT, WM8400_SPKATTN_MASK, 0), +SOC_SINGLE("Speaker DC Boost Volume", WM8400_CLASSD3, + WM8400_DCGAIN_SHIFT, 6, 0), +SOC_SINGLE("Speaker AC Boost Volume", WM8400_CLASSD3, + WM8400_ACGAIN_SHIFT, 6, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", + WM8400_LEFT_DAC_DIGITAL_VOLUME, WM8400_DACL_VOL_SHIFT, + 127, 0, out_dac_tlv), + +WM8400_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume", + WM8400_RIGHT_DAC_DIGITAL_VOLUME, WM8400_DACR_VOL_SHIFT, + 127, 0, out_dac_tlv), + +SOC_ENUM("Left Digital Sidetone", wm8400_left_digital_sidetone_enum), +SOC_ENUM("Right Digital Sidetone", wm8400_right_digital_sidetone_enum), + +SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE, + WM8400_ADCL_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv), +SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE, + WM8400_ADCR_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv), + +SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8400_ADC_CTRL, + WM8400_ADC_HPF_ENA_SHIFT, 1, 0), + +SOC_ENUM("ADC HPF Mode", wm8400_right_adcmode_enum), + +WM8400_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume", + WM8400_LEFT_ADC_DIGITAL_VOLUME, + WM8400_ADCL_VOL_SHIFT, + WM8400_ADCL_VOL_MASK, + 0, + in_adc_tlv), + +WM8400_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume", + WM8400_RIGHT_ADC_DIGITAL_VOLUME, + WM8400_ADCR_VOL_SHIFT, + WM8400_ADCR_VOL_MASK, + 0, + in_adc_tlv), + +WM8400_OUTPGA_SINGLE_R_TLV("LIN12 Volume", + WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + WM8400_LIN12VOL_SHIFT, + WM8400_LIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN12 ZC Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + WM8400_LI12ZC_SHIFT, 1, 0), + +SOC_SINGLE("LIN12 Mute Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + WM8400_LI12MUTE_SHIFT, 1, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("LIN34 Volume", + WM8400_LEFT_LINE_INPUT_3_4_VOLUME, + WM8400_LIN34VOL_SHIFT, + WM8400_LIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("LIN34 ZC Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME, + WM8400_LI34ZC_SHIFT, 1, 0), + +SOC_SINGLE("LIN34 Mute Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME, + WM8400_LI34MUTE_SHIFT, 1, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("RIN12 Volume", + WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8400_RIN12VOL_SHIFT, + WM8400_RIN12VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN12 ZC Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8400_RI12ZC_SHIFT, 1, 0), + +SOC_SINGLE("RIN12 Mute Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + WM8400_RI12MUTE_SHIFT, 1, 0), + +WM8400_OUTPGA_SINGLE_R_TLV("RIN34 Volume", + WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8400_RIN34VOL_SHIFT, + WM8400_RIN34VOL_MASK, + 0, + in_pga_tlv), + +SOC_SINGLE("RIN34 ZC Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8400_RI34ZC_SHIFT, 1, 0), + +SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, + WM8400_RI34MUTE_SHIFT, 1, 0), + +}; + +/* add non dapm controls */ +static int wm8400_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, wm8400_snd_controls, + ARRAY_SIZE(wm8400_snd_controls)); +} + +/* + * _DAPM_ Controls + */ + +static int inmixer_event (struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 reg, fakepower; + + reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2); + fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS); + + if (fakepower & ((1 << WM8400_INMIXL_PWR) | + (1 << WM8400_AINLMUX_PWR))) { + reg |= WM8400_AINL_ENA; + } else { + reg &= ~WM8400_AINL_ENA; + } + + if (fakepower & ((1 << WM8400_INMIXR_PWR) | + (1 << WM8400_AINRMUX_PWR))) { + reg |= WM8400_AINR_ENA; + } else { + reg &= ~WM8400_AINL_ENA; + } + wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); + + return 0; +} + +static int outmixer_event (struct snd_soc_dapm_widget *w, + struct snd_kcontrol * kcontrol, int event) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + u32 reg_shift = mc->shift; + int ret = 0; + u16 reg; + + switch (reg_shift) { + case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) : + reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1); + if (reg & WM8400_LDLO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 1 LDLO Set\n"); + ret = -1; + } + break; + case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8): + reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2); + if (reg & WM8400_RDRO) { + printk(KERN_WARNING + "Cannot set as Output Mixer 2 RDRO Set\n"); + ret = -1; + } + break; + case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8): + reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + if (reg & WM8400_LDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer LDSPK Set\n"); + ret = -1; + } + break; + case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8): + reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER); + if (reg & WM8400_RDSPK) { + printk(KERN_WARNING + "Cannot set as Speaker Mixer RDSPK Set\n"); + ret = -1; + } + break; + } + + return ret; +} + +/* INMIX dB values */ +static const unsigned int in_mix_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 0,7, TLV_DB_LINEAR_ITEM(-1200, 600), +}; + +/* Left In PGA Connections */ +static const struct snd_kcontrol_new wm8400_dapm_lin12_pga_controls[] = { +SOC_DAPM_SINGLE("LIN1 Switch", WM8400_INPUT_MIXER2, WM8400_LMN1_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LIN2 Switch", WM8400_INPUT_MIXER2, WM8400_LMP2_SHIFT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8400_dapm_lin34_pga_controls[] = { +SOC_DAPM_SINGLE("LIN3 Switch", WM8400_INPUT_MIXER2, WM8400_LMN3_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LIN4 Switch", WM8400_INPUT_MIXER2, WM8400_LMP4_SHIFT, 1, 0), +}; + +/* Right In PGA Connections */ +static const struct snd_kcontrol_new wm8400_dapm_rin12_pga_controls[] = { +SOC_DAPM_SINGLE("RIN1 Switch", WM8400_INPUT_MIXER2, WM8400_RMN1_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RIN2 Switch", WM8400_INPUT_MIXER2, WM8400_RMP2_SHIFT, 1, 0), +}; + +static const struct snd_kcontrol_new wm8400_dapm_rin34_pga_controls[] = { +SOC_DAPM_SINGLE("RIN3 Switch", WM8400_INPUT_MIXER2, WM8400_RMN3_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RIN4 Switch", WM8400_INPUT_MIXER2, WM8400_RMP4_SHIFT, 1, 0), +}; + +/* INMIXL */ +static const struct snd_kcontrol_new wm8400_dapm_inmixl_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8400_INPUT_MIXER3, + WM8400_LDBVOL_SHIFT, WM8400_LDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8400_INPUT_MIXER5, WM8400_LI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("LINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT, + 1, 0), +SOC_DAPM_SINGLE("LINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, + 1, 0), +}; + +/* INMIXR */ +static const struct snd_kcontrol_new wm8400_dapm_inmixr_controls[] = { +SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8400_INPUT_MIXER4, + WM8400_RDBVOL_SHIFT, WM8400_RDBVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8400_INPUT_MIXER6, WM8400_RI2BVOL_SHIFT, + 7, 0, in_mix_tlv), +SOC_DAPM_SINGLE("RINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT, + 1, 0), +SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, + 1, 0), +}; + +/* AINLMUX */ +static const char *wm8400_ainlmux[] = + {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; + +static const struct soc_enum wm8400_ainlmux_enum = +SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT, + ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux); + +static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls = +SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); + +/* DIFFINL */ + +/* AINRMUX */ +static const char *wm8400_ainrmux[] = + {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; + +static const struct soc_enum wm8400_ainrmux_enum = +SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT, + ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux); + +static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = +SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); + +/* RXVOICE */ +static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = { +SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT, + WM8400_LR4BVOL_MASK, 0, in_mix_tlv), +SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT, + WM8400_RL4BVOL_MASK, 0, in_mix_tlv), +}; + +/* LOMIX */ +static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = { +SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LRBLO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LLBLO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LRI3LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LLI3LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LR12LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1, + WM8400_LL12LO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8400_OUTPUT_MIXER1, + WM8400_LDLO_SHIFT, 1, 0), +}; + +/* ROMIX */ +static const struct snd_kcontrol_new wm8400_dapm_romix_controls[] = { +SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RLBRO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RRBRO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RLI3RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RRI3RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RL12RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2, + WM8400_RR12RO_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8400_OUTPUT_MIXER2, + WM8400_RDRO_SHIFT, 1, 0), +}; + +/* LONMIX */ +static const struct snd_kcontrol_new wm8400_dapm_lonmix_controls[] = { +SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1, + WM8400_LLOPGALON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER1, + WM8400_LROPGALON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8400_LINE_MIXER1, + WM8400_LOPLON_SHIFT, 1, 0), +}; + +/* LOPMIX */ +static const struct snd_kcontrol_new wm8400_dapm_lopmix_controls[] = { +SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER1, + WM8400_LR12LOP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER1, + WM8400_LL12LOP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1, + WM8400_LLOPGALOP_SHIFT, 1, 0), +}; + +/* RONMIX */ +static const struct snd_kcontrol_new wm8400_dapm_ronmix_controls[] = { +SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2, + WM8400_RROPGARON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER2, + WM8400_RLOPGARON_SHIFT, 1, 0), +SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8400_LINE_MIXER2, + WM8400_ROPRON_SHIFT, 1, 0), +}; + +/* ROPMIX */ +static const struct snd_kcontrol_new wm8400_dapm_ropmix_controls[] = { +SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER2, + WM8400_RL12ROP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER2, + WM8400_RR12ROP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2, + WM8400_RROPGAROP_SHIFT, 1, 0), +}; + +/* OUT3MIX */ +static const struct snd_kcontrol_new wm8400_dapm_out3mix_controls[] = { +SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER, + WM8400_LI4O3_SHIFT, 1, 0), +SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8400_OUT3_4_MIXER, + WM8400_LPGAO3_SHIFT, 1, 0), +}; + +/* OUT4MIX */ +static const struct snd_kcontrol_new wm8400_dapm_out4mix_controls[] = { +SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8400_OUT3_4_MIXER, + WM8400_RPGAO4_SHIFT, 1, 0), +SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER, + WM8400_RI4O4_SHIFT, 1, 0), +}; + +/* SPKMIX */ +static const struct snd_kcontrol_new wm8400_dapm_spkmix_controls[] = { +SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_LI2SPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_LB2SPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8400_SPEAKER_MIXER, + WM8400_LOPGASPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8400_SPEAKER_MIXER, + WM8400_LDSPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8400_SPEAKER_MIXER, + WM8400_RDSPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8400_SPEAKER_MIXER, + WM8400_ROPGASPK_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_RL12ROP_SHIFT, 1, 0), +SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8400_SPEAKER_MIXER, + WM8400_RI2SPK_SHIFT, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8400_dapm_widgets[] = { +/* Input Side */ +/* Input Lines */ +SND_SOC_DAPM_INPUT("LIN1"), +SND_SOC_DAPM_INPUT("LIN2"), +SND_SOC_DAPM_INPUT("LIN3"), +SND_SOC_DAPM_INPUT("LIN4/RXN"), +SND_SOC_DAPM_INPUT("RIN3"), +SND_SOC_DAPM_INPUT("RIN4/RXP"), +SND_SOC_DAPM_INPUT("RIN1"), +SND_SOC_DAPM_INPUT("RIN2"), +SND_SOC_DAPM_INPUT("Internal ADC Source"), + +/* DACs */ +SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8400_POWER_MANAGEMENT_2, + WM8400_ADCL_ENA_SHIFT, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8400_POWER_MANAGEMENT_2, + WM8400_ADCR_ENA_SHIFT, 0), + +/* Input PGAs */ +SND_SOC_DAPM_MIXER("LIN12 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_LIN12_ENA_SHIFT, + 0, &wm8400_dapm_lin12_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_lin12_pga_controls)), +SND_SOC_DAPM_MIXER("LIN34 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_LIN34_ENA_SHIFT, + 0, &wm8400_dapm_lin34_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_lin34_pga_controls)), +SND_SOC_DAPM_MIXER("RIN12 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_RIN12_ENA_SHIFT, + 0, &wm8400_dapm_rin12_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_rin12_pga_controls)), +SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, + WM8400_RIN34_ENA_SHIFT, + 0, &wm8400_dapm_rin34_pga_controls[0], + ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), + +/* INMIXL */ +SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, + &wm8400_dapm_inmixl_controls[0], + ARRAY_SIZE(wm8400_dapm_inmixl_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINLMUX */ +SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, + &wm8400_dapm_ainlmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* INMIXR */ +SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, + &wm8400_dapm_inmixr_controls[0], + ARRAY_SIZE(wm8400_dapm_inmixr_controls), + inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* AINRMUX */ +SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, + &wm8400_dapm_ainrmux_controls, inmixer_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + +/* Output Side */ +/* DACs */ +SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8400_POWER_MANAGEMENT_3, + WM8400_DACL_ENA_SHIFT, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8400_POWER_MANAGEMENT_3, + WM8400_DACR_ENA_SHIFT, 0), + +/* LOMIX */ +SND_SOC_DAPM_MIXER_E("LOMIX", WM8400_POWER_MANAGEMENT_3, + WM8400_LOMIX_ENA_SHIFT, + 0, &wm8400_dapm_lomix_controls[0], + ARRAY_SIZE(wm8400_dapm_lomix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LONMIX */ +SND_SOC_DAPM_MIXER("LONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LON_ENA_SHIFT, + 0, &wm8400_dapm_lonmix_controls[0], + ARRAY_SIZE(wm8400_dapm_lonmix_controls)), + +/* LOPMIX */ +SND_SOC_DAPM_MIXER("LOPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LOP_ENA_SHIFT, + 0, &wm8400_dapm_lopmix_controls[0], + ARRAY_SIZE(wm8400_dapm_lopmix_controls)), + +/* OUT3MIX */ +SND_SOC_DAPM_MIXER("OUT3MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT3_ENA_SHIFT, + 0, &wm8400_dapm_out3mix_controls[0], + ARRAY_SIZE(wm8400_dapm_out3mix_controls)), + +/* SPKMIX */ +SND_SOC_DAPM_MIXER_E("SPKMIX", WM8400_POWER_MANAGEMENT_1, WM8400_SPK_ENA_SHIFT, + 0, &wm8400_dapm_spkmix_controls[0], + ARRAY_SIZE(wm8400_dapm_spkmix_controls), outmixer_event, + SND_SOC_DAPM_PRE_REG), + +/* OUT4MIX */ +SND_SOC_DAPM_MIXER("OUT4MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT4_ENA_SHIFT, + 0, &wm8400_dapm_out4mix_controls[0], + ARRAY_SIZE(wm8400_dapm_out4mix_controls)), + +/* ROPMIX */ +SND_SOC_DAPM_MIXER("ROPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_ROP_ENA_SHIFT, + 0, &wm8400_dapm_ropmix_controls[0], + ARRAY_SIZE(wm8400_dapm_ropmix_controls)), + +/* RONMIX */ +SND_SOC_DAPM_MIXER("RONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_RON_ENA_SHIFT, + 0, &wm8400_dapm_ronmix_controls[0], + ARRAY_SIZE(wm8400_dapm_ronmix_controls)), + +/* ROMIX */ +SND_SOC_DAPM_MIXER_E("ROMIX", WM8400_POWER_MANAGEMENT_3, + WM8400_ROMIX_ENA_SHIFT, + 0, &wm8400_dapm_romix_controls[0], + ARRAY_SIZE(wm8400_dapm_romix_controls), + outmixer_event, SND_SOC_DAPM_PRE_REG), + +/* LOUT PGA */ +SND_SOC_DAPM_PGA("LOUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_LOUT_ENA_SHIFT, + 0, NULL, 0), + +/* ROUT PGA */ +SND_SOC_DAPM_PGA("ROUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_ROUT_ENA_SHIFT, + 0, NULL, 0), + +/* LOPGA */ +SND_SOC_DAPM_PGA("LOPGA", WM8400_POWER_MANAGEMENT_3, WM8400_LOPGA_ENA_SHIFT, 0, + NULL, 0), + +/* ROPGA */ +SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0, + NULL, 0), + +/* MICBIAS */ +SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1, + WM8400_MIC1BIAS_ENA_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("LON"), +SND_SOC_DAPM_OUTPUT("LOP"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("SPKN"), +SND_SOC_DAPM_OUTPUT("SPKP"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_OUTPUT("ROP"), +SND_SOC_DAPM_OUTPUT("RON"), + +SND_SOC_DAPM_OUTPUT("Internal DAC Sink"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Make DACs turn on when playing even if not mixed into any outputs */ + {"Internal DAC Sink", NULL, "Left DAC"}, + {"Internal DAC Sink", NULL, "Right DAC"}, + + /* Make ADCs turn on when recording + * even if not mixed from any inputs */ + {"Left ADC", NULL, "Internal ADC Source"}, + {"Right ADC", NULL, "Internal ADC Source"}, + + /* Input Side */ + /* LIN12 PGA */ + {"LIN12 PGA", "LIN1 Switch", "LIN1"}, + {"LIN12 PGA", "LIN2 Switch", "LIN2"}, + /* LIN34 PGA */ + {"LIN34 PGA", "LIN3 Switch", "LIN3"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, + /* INMIXL */ + {"INMIXL", "Record Left Volume", "LOMIX"}, + {"INMIXL", "LIN2 Volume", "LIN2"}, + {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, + {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, + /* AILNMUX */ + {"AILNMUX", "INMIXL Mix", "INMIXL"}, + {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Left ADC", NULL, "AILNMUX"}, + + /* RIN12 PGA */ + {"RIN12 PGA", "RIN1 Switch", "RIN1"}, + {"RIN12 PGA", "RIN2 Switch", "RIN2"}, + /* RIN34 PGA */ + {"RIN34 PGA", "RIN3 Switch", "RIN3"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, + /* INMIXL */ + {"INMIXR", "Record Right Volume", "ROMIX"}, + {"INMIXR", "RIN2 Volume", "RIN2"}, + {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, + {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, + /* AIRNMUX */ + {"AIRNMUX", "INMIXR Mix", "INMIXR"}, + {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AIRNMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* ADC */ + {"Right ADC", NULL, "AIRNMUX"}, + + /* LOMIX */ + {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, + {"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"}, + {"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"LOMIX", "LOMIX Right ADC Bypass Switch", "AIRNMUX"}, + {"LOMIX", "LOMIX Left ADC Bypass Switch", "AILNMUX"}, + {"LOMIX", "LOMIX Left DAC Switch", "Left DAC"}, + + /* ROMIX */ + {"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"}, + {"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"}, + {"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"}, + {"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"}, + {"ROMIX", "ROMIX Right ADC Bypass Switch", "AIRNMUX"}, + {"ROMIX", "ROMIX Left ADC Bypass Switch", "AILNMUX"}, + {"ROMIX", "ROMIX Right DAC Switch", "Right DAC"}, + + /* SPKMIX */ + {"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"}, + {"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"}, + {"SPKMIX", "SPKMIX LADC Bypass Switch", "AILNMUX"}, + {"SPKMIX", "SPKMIX RADC Bypass Switch", "AIRNMUX"}, + {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, + {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, + {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + + /* LONMIX */ + {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, + {"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"}, + {"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"}, + + /* LOPMIX */ + {"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, + + /* OUT3MIX */ + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"}, + {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, + + /* OUT4MIX */ + {"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"}, + {"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"}, + + /* RONMIX */ + {"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"}, + {"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"}, + {"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"}, + + /* ROPMIX */ + {"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"}, + {"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"}, + + /* Out Mixer PGAs */ + {"LOPGA", NULL, "LOMIX"}, + {"ROPGA", NULL, "ROMIX"}, + + {"LOUT PGA", NULL, "LOMIX"}, + {"ROUT PGA", NULL, "ROMIX"}, + + /* Output Pins */ + {"LON", NULL, "LONMIX"}, + {"LOP", NULL, "LOPMIX"}, + {"OUT3", NULL, "OUT3MIX"}, + {"LOUT", NULL, "LOUT PGA"}, + {"SPKN", NULL, "SPKMIX"}, + {"ROUT", NULL, "ROUT PGA"}, + {"OUT4", NULL, "OUT4MIX"}, + {"ROP", NULL, "ROPMIX"}, + {"RON", NULL, "RONMIX"}, +}; + +static int wm8400_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets, + ARRAY_SIZE(wm8400_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +/* + * Clock after FLL and dividers + */ +static int wm8400_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 wm8400_priv *wm8400 = codec->private_data; + + wm8400->sysclk = freq; + return 0; +} + +struct fll_factors { + u16 n; + u16 k; + u16 outdiv; + u16 fratio; + u16 freq_ref; +}; + +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, + unsigned int Fref, unsigned int Fout) +{ + u64 Kpart; + unsigned int K, Nmod, target; + + factors->outdiv = 2; + while (Fout * factors->outdiv < 90000000 || + Fout * factors->outdiv > 100000000) { + factors->outdiv *= 2; + if (factors->outdiv > 32) { + dev_err(wm8400->wm8400->dev, + "Unsupported FLL output frequency %dHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * factors->outdiv; + factors->outdiv = factors->outdiv >> 2; + + if (Fref < 48000) + factors->freq_ref = 1; + else + factors->freq_ref = 0; + + if (Fref < 1000000) + factors->fratio = 9; + else + factors->fratio = 0; + + /* Ensure we have a fractional part */ + do { + if (Fref < 1000000) + factors->fratio--; + else + factors->fratio++; + + if (factors->fratio < 1 || factors->fratio > 8) { + dev_err(wm8400->wm8400->dev, + "Unable to calculate FRATIO\n"); + return -EINVAL; + } + + factors->n = target / (Fref * factors->fratio); + Nmod = target % (Fref * factors->fratio); + } while (Nmod == 0); + + /* Calculate fractional part - scale up so we can round. */ + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + + do_div(Kpart, (Fref * factors->fratio)); + + K = Kpart & 0xFFFFFFFF; + + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + factors->k = K / 10; + + dev_dbg(wm8400->wm8400->dev, + "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n", + Fref, Fout, + factors->n, factors->k, factors->fratio, factors->outdiv); + + return 0; +} + +static int wm8400_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; + struct wm8400_priv *wm8400 = codec->private_data; + struct fll_factors factors; + int ret; + u16 reg; + + if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out) + return 0; + + if (freq_out != 0) { + ret = fll_factors(wm8400, &factors, freq_in, freq_out); + if (ret != 0) + return ret; + } + + wm8400->fll_out = freq_out; + wm8400->fll_in = freq_in; + + /* We *must* disable the FLL before any changes */ + reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2); + reg &= ~WM8400_FLL_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg); + + reg = wm8400_read(codec, WM8400_FLL_CONTROL_1); + reg &= ~WM8400_FLL_OSC_ENA; + wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + + if (freq_out == 0) + return 0; + + reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK); + reg |= WM8400_FLL_FRAC | factors.fratio; + reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT; + wm8400_write(codec, WM8400_FLL_CONTROL_1, reg); + + wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k); + wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); + + reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); + reg &= WM8400_FLL_OUTDIV_MASK; + reg |= factors.outdiv; + wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); + + return 0; +} + +/* + * Sets ADC and Voice DAC format. + */ +static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 audio1, audio3; + + audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + audio3 &= ~WM8400_AIF_MSTR1; + break; + case SND_SOC_DAIFMT_CBM_CFM: + audio3 |= WM8400_AIF_MSTR1; + break; + default: + return -EINVAL; + } + + audio1 &= ~WM8400_AIF_FMT_MASK; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + audio1 |= WM8400_AIF_FMT_I2S; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_RIGHT_J: + audio1 |= WM8400_AIF_FMT_RIGHTJ; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_LEFT_J: + audio1 |= WM8400_AIF_FMT_LEFTJ; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_A: + audio1 |= WM8400_AIF_FMT_DSP; + audio1 &= ~WM8400_AIF_LRCLK_INV; + break; + case SND_SOC_DAIFMT_DSP_B: + audio1 |= WM8400_AIF_FMT_DSP | WM8400_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3); + return 0; +} + +static int wm8400_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 WM8400_MCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_2) & + ~WM8400_MCLK_DIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + break; + case WM8400_DACCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_2) & + ~WM8400_DAC_CLKDIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + break; + case WM8400_ADCCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_2) & + ~WM8400_ADC_CLKDIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_2, reg | div); + break; + case WM8400_BCLK_DIV: + reg = wm8400_read(codec, WM8400_CLOCKING_1) & + ~WM8400_BCLK_DIV_MASK; + wm8400_write(codec, WM8400_CLOCKING_1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + */ +static int wm8400_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; + u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1); + + audio1 &= ~WM8400_AIF_WL_MASK; + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + audio1 |= WM8400_AIF_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + audio1 |= WM8400_AIF_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + audio1 |= WM8400_AIF_WL_32BITS; + break; + } + + wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1); + return 0; +} + +static int wm8400_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE; + + if (mute) + wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + else + wm8400_write(codec, WM8400_DAC_CTRL, val); + + return 0; +} + +/* TODO: set bias for best performance at standby */ +static int wm8400_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8400_priv *wm8400 = codec->private_data; + u16 val; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + ~WM8400_VMID_MODE_MASK; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(power), + &power[0]); + if (ret != 0) { + dev_err(wm8400->wm8400->dev, + "Failed to enable regulators: %d\n", + ret); + return ret; + } + + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, + WM8400_CODEC_ENA | WM8400_SYSCLK_ENA); + + /* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_BUFDCOPEN | WM8400_POBCTRL); + + msleep(50); + + /* Enable VREF & VMID at 2x50k */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val |= 0x2 | WM8400_VREF_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* Enable BUFIOEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_BUFDCOPEN | WM8400_POBCTRL | + WM8400_BUFIOEN); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN); + } + + /* VMID=2*300k */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) & + ~WM8400_VMID_MODE_MASK; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4); + break; + + case SND_SOC_BIAS_OFF: + /* Enable POBCTRL and SOFT_ST */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_POBCTRL | WM8400_BUFIOEN); + + /* Enable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST | + WM8400_BUFDCOPEN | WM8400_POBCTRL | + WM8400_BUFIOEN); + + /* mute DAC */ + val = wm8400_read(codec, WM8400_DAC_CTRL); + wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE); + + /* Enable any disabled outputs */ + val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + val |= WM8400_SPK_ENA | WM8400_OUT3_ENA | + WM8400_OUT4_ENA | WM8400_LOUT_ENA | + WM8400_ROUT_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* Disable VMID */ + val &= ~WM8400_VMID_MODE_MASK; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + msleep(300); + + /* Enable all output discharge bits */ + wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE | + WM8400_DIS_RLINE | WM8400_DIS_OUT3 | + WM8400_DIS_OUT4 | WM8400_DIS_LOUT | + WM8400_DIS_ROUT); + + /* Disable VREF */ + val &= ~WM8400_VREF_ENA; + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val); + + /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ + wm8400_write(codec, WM8400_ANTIPOP2, 0x0); + + ret = regulator_bulk_disable(ARRAY_SIZE(power), + &power[0]); + if (ret != 0) + return ret; + + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8400_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8400_dai_ops = { + .hw_params = wm8400_hw_params, + .digital_mute = wm8400_mute, + .set_fmt = wm8400_set_dai_fmt, + .set_clkdiv = wm8400_set_dai_clkdiv, + .set_sysclk = wm8400_set_dai_sysclk, + .set_pll = wm8400_set_dai_pll, +}; + +/* + * The WM8400 supports 2 different and mutually exclusive DAI + * configurations. + * + * 1. ADC/DAC on Primary Interface + * 2. ADC on Primary Interface/DAC on secondary + */ +struct snd_soc_dai wm8400_dai = { +/* ADC/DAC on primary */ + .name = "WM8400 ADC/DAC Primary", + .id = 1, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8400_RATES, + .formats = WM8400_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8400_RATES, + .formats = WM8400_FORMATS, + }, + .ops = &wm8400_dai_ops, +}; +EXPORT_SYMBOL_GPL(wm8400_dai); + +static int wm8400_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; + + wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8400_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec *wm8400_codec; + +static int wm8400_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (!wm8400_codec) { + dev_err(&pdev->dev, "wm8400 not yet discovered\n"); + return -ENODEV; + } + codec = wm8400_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; + } + + wm8400_add_controls(codec); + wm8400_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 wm8400_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_wm8400 = { + .probe = wm8400_probe, + .remove = wm8400_remove, + .suspend = wm8400_suspend, + .resume = wm8400_resume, +}; + +static void wm8400_probe_deferred(struct work_struct *work) +{ + struct wm8400_priv *priv = container_of(work, struct wm8400_priv, + work); + struct snd_soc_codec *codec = &priv->codec; + int ret; + + /* charge output caps */ + wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* We're done, tell the subsystem. */ + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(priv->wm8400->dev, + "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8400_dai); + if (ret != 0) { + dev_err(priv->wm8400->dev, + "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return; + +err_codec: + snd_soc_unregister_codec(codec); +err: + wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int wm8400_codec_probe(struct platform_device *dev) +{ + struct wm8400_priv *priv; + int ret; + u16 reg; + struct snd_soc_codec *codec; + + priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + codec = &priv->codec; + codec->private_data = priv; + codec->control_data = dev->dev.driver_data; + priv->wm8400 = dev->dev.driver_data; + + ret = regulator_bulk_get(priv->wm8400->dev, + ARRAY_SIZE(power), &power[0]); + if (ret != 0) { + dev_err(&dev->dev, "Failed to get regulators: %d\n", ret); + goto err; + } + + codec->dev = &dev->dev; + wm8400_dai.dev = &dev->dev; + + codec->name = "WM8400"; + codec->owner = THIS_MODULE; + codec->read = wm8400_read; + codec->write = wm8400_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8400_set_bias_level; + codec->dai = &wm8400_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8400_REGISTER_COUNT; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + INIT_WORK(&priv->work, wm8400_probe_deferred); + + wm8400_codec_reset(codec); + + reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1); + wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA); + + /* Latch volume update bits */ + reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME); + wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME, + reg & WM8400_IPVU); + reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME); + wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME, + reg & WM8400_IPVU); + + wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); + wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); + + wm8400_codec = codec; + + if (!schedule_work(&priv->work)) { + ret = -EINVAL; + goto err_regulator; + } + + return 0; + +err_regulator: + wm8400_codec = NULL; + regulator_bulk_free(ARRAY_SIZE(power), power); +err: + kfree(priv); + return ret; +} + +static int __exit wm8400_codec_remove(struct platform_device *dev) +{ + struct wm8400_priv *priv = wm8400_codec->private_data; + u16 reg; + + snd_soc_unregister_dai(&wm8400_dai); + snd_soc_unregister_codec(wm8400_codec); + + reg = wm8400_read(wm8400_codec, WM8400_POWER_MANAGEMENT_1); + wm8400_write(wm8400_codec, WM8400_POWER_MANAGEMENT_1, + reg & (~WM8400_CODEC_ENA)); + + regulator_bulk_free(ARRAY_SIZE(power), power); + kfree(priv); + + wm8400_codec = NULL; + + return 0; +} + +static struct platform_driver wm8400_codec_driver = { + .driver = { + .name = "wm8400-codec", + .owner = THIS_MODULE, + }, + .probe = wm8400_codec_probe, + .remove = __exit_p(wm8400_codec_remove), +}; + +static int __init wm8400_codec_init(void) +{ + return platform_driver_register(&wm8400_codec_driver); +} +module_init(wm8400_codec_init); + +static void __exit wm8400_codec_exit(void) +{ + platform_driver_unregister(&wm8400_codec_driver); +} +module_exit(wm8400_codec_exit); + +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8400); + +MODULE_DESCRIPTION("ASoC WM8400 driver"); +MODULE_AUTHOR("Mark Brown"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8400-codec"); diff --git a/sound/soc/codecs/wm8400.h b/sound/soc/codecs/wm8400.h new file mode 100644 index 000000000000..79c5934d4776 --- /dev/null +++ b/sound/soc/codecs/wm8400.h @@ -0,0 +1,62 @@ +/* + * wm8400.h -- audio driver for WM8400 + * + * Copyright 2008 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. + * + */ + +#ifndef _WM8400_CODEC_H +#define _WM8400_CODEC_H + +#define WM8400_MCLK_DIV 0 +#define WM8400_DACCLK_DIV 1 +#define WM8400_ADCCLK_DIV 2 +#define WM8400_BCLK_DIV 3 + +#define WM8400_MCLK_DIV_1 0x400 +#define WM8400_MCLK_DIV_2 0x800 + +#define WM8400_DAC_CLKDIV_1 0x00 +#define WM8400_DAC_CLKDIV_1_5 0x04 +#define WM8400_DAC_CLKDIV_2 0x08 +#define WM8400_DAC_CLKDIV_3 0x0c +#define WM8400_DAC_CLKDIV_4 0x10 +#define WM8400_DAC_CLKDIV_5_5 0x14 +#define WM8400_DAC_CLKDIV_6 0x18 + +#define WM8400_ADC_CLKDIV_1 0x00 +#define WM8400_ADC_CLKDIV_1_5 0x20 +#define WM8400_ADC_CLKDIV_2 0x40 +#define WM8400_ADC_CLKDIV_3 0x60 +#define WM8400_ADC_CLKDIV_4 0x80 +#define WM8400_ADC_CLKDIV_5_5 0xa0 +#define WM8400_ADC_CLKDIV_6 0xc0 + + +#define WM8400_BCLK_DIV_1 (0x0 << 1) +#define WM8400_BCLK_DIV_1_5 (0x1 << 1) +#define WM8400_BCLK_DIV_2 (0x2 << 1) +#define WM8400_BCLK_DIV_3 (0x3 << 1) +#define WM8400_BCLK_DIV_4 (0x4 << 1) +#define WM8400_BCLK_DIV_5_5 (0x5 << 1) +#define WM8400_BCLK_DIV_6 (0x6 << 1) +#define WM8400_BCLK_DIV_8 (0x7 << 1) +#define WM8400_BCLK_DIV_11 (0x8 << 1) +#define WM8400_BCLK_DIV_12 (0x9 << 1) +#define WM8400_BCLK_DIV_16 (0xA << 1) +#define WM8400_BCLK_DIV_22 (0xB << 1) +#define WM8400_BCLK_DIV_24 (0xC << 1) +#define WM8400_BCLK_DIV_32 (0xD << 1) +#define WM8400_BCLK_DIV_44 (0xE << 1) +#define WM8400_BCLK_DIV_48 (0xF << 1) + +extern struct snd_soc_dai wm8400_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8400; + +#endif diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 40f8238df717..6a4cea09c45d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), }; -/* add non dapm controls */ -static int wm8510_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8510_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Speaker Output Mixer */ static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), @@ -352,7 +336,7 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } - pll_factors(freq_out*8, freq_in); + 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); @@ -383,7 +367,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai, wm8510_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f; wm8510_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: @@ -468,7 +452,7 @@ 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->codec; + 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; @@ -570,6 +554,14 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec, #define WM8510_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 wm8510_dai_ops = { + .hw_params = wm8510_pcm_hw_params, + .digital_mute = wm8510_mute, + .set_fmt = wm8510_set_dai_fmt, + .set_clkdiv = wm8510_set_dai_clkdiv, + .set_pll = wm8510_set_dai_pll, +}; + struct snd_soc_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { @@ -584,20 +576,14 @@ struct snd_soc_dai wm8510_dai = { .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, - .ops = { - .hw_params = wm8510_pcm_hw_params, - .digital_mute = wm8510_mute, - .set_fmt = wm8510_set_dai_fmt, - .set_clkdiv = wm8510_set_dai_clkdiv, - .set_pll = wm8510_set_dai_pll, - }, + .ops = &wm8510_dai_ops, }; EXPORT_SYMBOL_GPL(wm8510_dai); static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -606,7 +592,7 @@ static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) static int wm8510_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -628,7 +614,7 @@ static int wm8510_resume(struct platform_device *pdev) */ static int wm8510_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8510"; @@ -656,7 +642,8 @@ static int wm8510_init(struct snd_soc_device *socdev) /* power on device */ codec->bias_level = SND_SOC_BIAS_OFF; wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8510_add_controls(codec); + snd_soc_add_controls(codec, wm8510_snd_controls, + ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -685,7 +672,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8510_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -766,7 +753,7 @@ err_driver: static int __devinit wm8510_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8510_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -832,7 +819,7 @@ static int wm8510_probe(struct platform_device *pdev) if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -862,7 +849,7 @@ static int wm8510_probe(struct platform_device *pdev) static int wm8510_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d004e5845298..442ea6f160fc 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008 Wolfson Microelectronics PLC. + * Copyright 2008, 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 as published by the @@ -35,19 +35,6 @@ #include "wm8580.h" -#define WM8580_VERSION "0.1" - -struct pll_state { - unsigned int in; - unsigned int out; -}; - -/* codec private data */ -struct wm8580_priv { - struct pll_state a; - struct pll_state b; -}; - /* WM8580 register space */ #define WM8580_PLLA1 0x00 #define WM8580_PLLA2 0x01 @@ -102,6 +89,8 @@ struct wm8580_priv { #define WM8580_READBACK 0x34 #define WM8580_RESET 0x35 +#define WM8580_MAX_REGISTER 0x35 + /* PLLB4 (register 7h) */ #define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60 #define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20 @@ -193,6 +182,20 @@ static const u16 wm8580_reg[] = { 0x0000, 0x0000 /*R53*/ }; +struct pll_state { + unsigned int in; + unsigned int out; +}; + +/* codec private data */ +struct wm8580_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8580_MAX_REGISTER + 1]; + struct pll_state a; + struct pll_state b; +}; + + /* * read wm8580 register cache */ @@ -200,7 +203,7 @@ 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)); + BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); return cache[reg]; } @@ -223,7 +226,7 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg, { u8 data[2]; - BUG_ON(reg > ARRAY_SIZE(wm8580_reg)); + BUG_ON(reg >= ARRAY_SIZE(wm8580_reg)); /* Registers are 9 bits wide */ value &= 0x1ff; @@ -330,20 +333,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0), SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0), }; -/* Add non-DAPM controls */ -static int wm8580_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8580_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1), SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1), @@ -553,7 +542,7 @@ 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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id); paifb &= ~WM8580_AIF_LENGTH_MASK; @@ -771,8 +760,22 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: + break; + 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 &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD); + wm8580_write(codec, WM8580_PWRDN1, reg); + + /* Make VMID high impedence */ + reg = wm8580_read(codec, WM8580_ADC_CONTROL1); + reg &= ~0x100; + wm8580_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); @@ -785,6 +788,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec, #define WM8580_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 wm8580_dai_ops_playback = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, + .digital_mute = wm8580_digital_mute, +}; + +static struct snd_soc_dai_ops wm8580_dai_ops_capture = { + .hw_params = wm8580_paif_hw_params, + .set_fmt = wm8580_set_paif_dai_fmt, + .set_clkdiv = wm8580_set_dai_clkdiv, + .set_pll = wm8580_set_dai_pll, +}; + struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", @@ -796,13 +814,7 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - .digital_mute = wm8580_digital_mute, - }, + .ops = &wm8580_dai_ops_playback, }, { .name = "WM8580 PAIFTX", @@ -814,109 +826,168 @@ struct snd_soc_dai wm8580_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = WM8580_FORMATS, }, - .ops = { - .hw_params = wm8580_paif_hw_params, - .set_fmt = wm8580_set_paif_dai_fmt, - .set_clkdiv = wm8580_set_dai_clkdiv, - .set_pll = wm8580_set_dai_pll, - }, + .ops = &wm8580_dai_ops_capture, }, }; EXPORT_SYMBOL_GPL(wm8580_dai); -/* - * initialise the WM8580 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8580_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8580_codec; + +static int wm8580_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; int ret = 0; - codec->name = "WM8580"; - codec->owner = THIS_MODULE; - codec->read = wm8580_read_reg_cache; - codec->write = wm8580_write; - codec->set_bias_level = wm8580_set_bias_level; - codec->dai = wm8580_dai; - codec->num_dai = ARRAY_SIZE(wm8580_dai); - codec->reg_cache_size = ARRAY_SIZE(wm8580_reg); - codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg), - GFP_KERNEL); - - if (codec->reg_cache == NULL) - return -ENOMEM; - - /* Get the codec into a known state */ - wm8580_write(codec, WM8580_RESET, 0); - - /* Power up and get individual control of the DACs */ - wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) & - ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD)); + if (wm8580_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - /* Make VMID high impedence */ - wm8580_write(codec, WM8580_ADC_CONTROL1, - wm8580_read(codec, WM8580_ADC_CONTROL1) & ~0x100); + socdev->card->codec = wm8580_codec; + codec = wm8580_codec; /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, - SNDRV_DEFAULT_STR1); + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "wm8580: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } - wm8580_add_controls(codec); + snd_soc_add_controls(codec, wm8580_snd_controls, + ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); - ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "wm8580: failed to register card\n"); + 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: - kfree(codec->reg_cache); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8580_socdev; +/* power down chip */ +static int wm8580_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + return 0; +} -/* - * WM8580 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ +struct snd_soc_codec_device soc_codec_dev_wm8580 = { + .probe = wm8580_probe, + .remove = wm8580_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); + +static int wm8580_register(struct wm8580_priv *wm8580) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8580->codec; + + if (wm8580_codec) { + dev_err(codec->dev, "Another WM8580 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 = 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; + codec->num_dai = ARRAY_SIZE(wm8580_dai); + codec->reg_cache_size = ARRAY_SIZE(wm8580->reg_cache); + codec->reg_cache = &wm8580->reg_cache; + + memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg)); + + /* Get the codec into a known state */ + ret = wm8580_write(codec, WM8580_RESET, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to reset codec: %d\n", ret); + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++) + wm8580_dai[i].dev = codec->dev; + + wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8580_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(wm8580_dai, ARRAY_SIZE(wm8580_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(wm8580); + return ret; +} + +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); + kfree(wm8580); + wm8580_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int wm8580_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8580_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; + struct wm8580_priv *wm8580; + struct snd_soc_codec *codec; - i2c_set_clientdata(i2c, codec); + wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL); + if (wm8580 == NULL) + return -ENOMEM; + + codec = &wm8580->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8580); codec->control_data = i2c; - ret = wm8580_init(socdev); - if (ret < 0) - dev_err(&i2c->dev, "failed to initialise WM8580\n"); - return ret; + codec->dev = &i2c->dev; + + return wm8580_register(wm8580); } static int wm8580_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8580_priv *wm8580 = i2c_get_clientdata(client); + wm8580_unregister(wm8580); return 0; } @@ -928,129 +999,35 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); static struct i2c_driver wm8580_i2c_driver = { .driver = { - .name = "WM8580 I2C Codec", + .name = "wm8580", .owner = THIS_MODULE, }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, .id_table = wm8580_i2c_id, }; +#endif -static int wm8580_add_i2c_device(struct platform_device *pdev, - const struct wm8580_setup_data *setup) +static int __init wm8580_modinit(void) { - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8580_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, "wm8580", 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; + pr_err("Failed to register WM8580 I2C driver: %d\n", ret); } - - return 0; - -err_driver: - i2c_del_driver(&wm8580_i2c_driver); - return -ENODEV; -} #endif -static int wm8580_probe(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8580_setup_data *setup; - struct snd_soc_codec *codec; - struct wm8580_priv *wm8580; - int ret = 0; - - pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION); - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL); - if (wm8580 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8580; - socdev->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8580_socdev = socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8580_add_i2c_device(pdev, setup); - } -#else - /* Add other interfaces here */ -#endif - return ret; -} - -/* power down chip */ -static int wm8580_remove(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; - - if (codec->control_data) - wm8580_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(&wm8580_i2c_driver); -#endif - kfree(codec->private_data); - kfree(codec); - return 0; } - -struct snd_soc_codec_device soc_codec_dev_wm8580 = { - .probe = wm8580_probe, - .remove = wm8580_remove, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); - -static int __init wm8580_modinit(void) -{ - return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); -} module_init(wm8580_modinit); static void __exit wm8580_exit(void) { - snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8580_i2c_driver); +#endif } module_exit(wm8580_exit); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 09e4422f6f2f..0dfb5ddde6a2 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -28,11 +28,6 @@ #define WM8580_CLKSRC_OSC 4 #define WM8580_CLKSRC_NONE 5 -struct wm8580_setup_data { - int i2c_bus; - unsigned short i2c_address; -}; - #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 80b11983e137..e7ff2121ede9 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -47,7 +47,7 @@ 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)); + BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); return cache[reg]; } @@ -55,7 +55,7 @@ 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)); + BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults)); cache[reg] = value; } @@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), }; -static int wm8728_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8728_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* * DAPM controls. */ @@ -152,7 +137,7 @@ 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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); dac &= ~0x18; @@ -259,6 +244,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec, #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8728_dai_ops = { + .hw_params = wm8728_hw_params, + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, +}; + struct snd_soc_dai wm8728_dai = { .name = "WM8728", .playback = { @@ -268,18 +259,14 @@ struct snd_soc_dai wm8728_dai = { .rates = WM8728_RATES, .formats = WM8728_FORMATS, }, - .ops = { - .hw_params = wm8728_hw_params, - .digital_mute = wm8728_mute, - .set_fmt = wm8728_set_dai_fmt, - } + .ops = &wm8728_dai_ops, }; EXPORT_SYMBOL_GPL(wm8728_dai); static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -289,7 +276,7 @@ static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) static int wm8728_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8728_set_bias_level(codec, codec->suspend_bias_level); @@ -302,7 +289,7 @@ static int wm8728_resume(struct platform_device *pdev) */ static int wm8728_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; codec->name = "WM8728"; @@ -330,7 +317,8 @@ static int wm8728_init(struct snd_soc_device *socdev) /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm8728_add_controls(codec); + snd_soc_add_controls(codec, wm8728_snd_controls, + ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -363,7 +351,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8728_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -444,7 +432,7 @@ err_driver: static int __devinit wm8728_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8728_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -508,7 +496,7 @@ static int wm8728_probe(struct platform_device *pdev) if (codec == NULL) return -ENOMEM; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -541,7 +529,7 @@ static int wm8728_probe(struct platform_device *pdev) static int wm8728_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c444b9f2701e..e043e3f60008 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -29,15 +29,20 @@ #include "wm8731.h" -#define WM8731_VERSION "0.13" - +static struct snd_soc_codec *wm8731_codec; struct snd_soc_codec_device soc_codec_dev_wm8731; /* codec private data */ struct wm8731_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8731_CACHEREGNUM]; 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 * We can't read the WM8731 register space when we are @@ -129,22 +134,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), SOC_ENUM("Playback De-emphasis", wm8731_enum[1]), }; -/* add non dapm controls */ -static int wm8731_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - /* Output Mixer */ static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), @@ -269,7 +258,7 @@ static int wm8731_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->codec; + 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; int i = get_coeff(wm8731->sysclk, params_rate(params)); @@ -299,7 +288,7 @@ static int wm8731_pcm_prepare(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* set active */ wm8731_write(codec, WM8731_ACTIVE, 0x0001); @@ -312,7 +301,7 @@ static void wm8731_shutdown(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* deactivate */ if (!codec->active) { @@ -414,21 +403,19 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, static int wm8731_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; + u16 reg; switch (level) { case SND_SOC_BIAS_ON: - /* vref/mid, osc on, dac unmute */ - wm8731_write(codec, WM8731_PWR, reg); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* everything off except vref/vmid, */ + /* Clear PWROFF, gate CLKOUT, everything else as-is */ + reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; wm8731_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - /* everything off, dac mute, inactive */ wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_write(codec, WM8731_PWR, 0xffff); break; @@ -446,6 +433,15 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8731_dai_ops = { + .prepare = wm8731_pcm_prepare, + .hw_params = wm8731_hw_params, + .shutdown = wm8731_shutdown, + .digital_mute = wm8731_mute, + .set_sysclk = wm8731_set_dai_sysclk, + .set_fmt = wm8731_set_dai_fmt, +}; + struct snd_soc_dai wm8731_dai = { .name = "WM8731", .playback = { @@ -460,21 +456,14 @@ struct snd_soc_dai wm8731_dai = { .channels_max = 2, .rates = WM8731_RATES, .formats = WM8731_FORMATS,}, - .ops = { - .prepare = wm8731_pcm_prepare, - .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, - .digital_mute = wm8731_mute, - .set_sysclk = wm8731_set_dai_sysclk, - .set_fmt = wm8731_set_dai_fmt, - } + .ops = &wm8731_dai_ops, }; EXPORT_SYMBOL_GPL(wm8731_dai); 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->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8731_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -484,7 +473,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) static int wm8731_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -500,54 +489,33 @@ static int wm8731_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8731 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8731_init(struct snd_soc_device *socdev) +static int wm8731_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->codec; - int reg, ret = 0; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; - codec->name = "WM8731"; - codec->owner = THIS_MODULE; - codec->read = wm8731_read_reg_cache; - codec->write = wm8731_write; - codec->set_bias_level = wm8731_set_bias_level; - codec->dai = &wm8731_dai; - codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); - codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + if (wm8731_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } - wm8731_reset(codec); + socdev->card->codec = wm8731_codec; + codec = wm8731_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "wm8731: failed to create pcms\n"); + dev_err(codec->dev, "failed to create pcms: %d\n", ret); goto pcm_err; } - /* power on device */ - wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* set 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); - - wm8731_add_controls(codec); + snd_soc_add_controls(codec, wm8731_snd_controls, + ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { - printk(KERN_ERR "wm8731: failed to register card\n"); + dev_err(codec->dev, "failed to register card: %d\n", ret); goto card_err; } @@ -557,133 +525,109 @@ 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 *wm8731_socdev; - -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -/* - * WM8731 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ - -static int wm8731_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +/* power down chip */ +static int wm8731_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; - - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = wm8731_init(socdev); - if (ret < 0) - pr_err("failed to initialise WM8731\n"); + struct snd_soc_device *socdev = platform_get_drvdata(pdev); - return ret; -} + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); -static int wm8731_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 wm8731_i2c_id[] = { - { "wm8731", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); - -static struct i2c_driver wm8731_i2c_driver = { - .driver = { - .name = "WM8731 I2C Codec", - .owner = THIS_MODULE, - }, - .probe = wm8731_i2c_probe, - .remove = wm8731_i2c_remove, - .id_table = wm8731_i2c_id, +struct snd_soc_codec_device soc_codec_dev_wm8731 = { + .probe = wm8731_probe, + .remove = wm8731_remove, + .suspend = wm8731_suspend, + .resume = wm8731_resume, }; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); -static int wm8731_add_i2c_device(struct platform_device *pdev, - const struct wm8731_setup_data *setup) +static int wm8731_register(struct wm8731_priv *wm8731) { - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; int ret; + struct snd_soc_codec *codec = &wm8731->codec; + u16 reg; - ret = i2c_add_driver(&wm8731_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; + if (wm8731_codec) { + dev_err(codec->dev, "Another WM8731 is registered\n"); + return -EINVAL; } - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); - 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; - } + 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; + codec->num_dai = 1; + codec->reg_cache_size = WM8731_CACHEREGNUM; + codec->reg_cache = &wm8731->reg_cache; - 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; + memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg)); + + ret = wm8731_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; } - return 0; + wm8731_dai.dev = codec->dev; -err_driver: - i2c_del_driver(&wm8731_i2c_driver); - return -ENODEV; -} -#endif + wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -#if defined(CONFIG_SPI_MASTER) -static int __devinit wm8731_spi_probe(struct spi_device *spi) -{ - struct snd_soc_device *socdev = wm8731_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; + /* 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); - codec->control_data = spi; + /* Disable bypass path by default */ + reg = wm8731_read_reg_cache(codec, WM8731_APANA); + wm8731_write(codec, WM8731_APANA, reg & ~0x4); - ret = wm8731_init(socdev); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8731\n"); + wm8731_codec = codec; - return ret; -} + 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(&wm8731_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } -static int __devexit wm8731_spi_remove(struct spi_device *spi) -{ return 0; } -static struct spi_driver wm8731_spi_driver = { - .driver = { - .name = "wm8731", - .bus = &spi_bus_type, - .owner = THIS_MODULE, - }, - .probe = wm8731_spi_probe, - .remove = __devexit_p(wm8731_spi_remove), -}; +static void wm8731_unregister(struct wm8731_priv *wm8731) +{ + wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8731_dai); + snd_soc_unregister_codec(&wm8731->codec); + kfree(wm8731); + wm8731_codec = NULL; +} +#if defined(CONFIG_SPI_MASTER) static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) { struct spi_transfer t; @@ -707,101 +651,121 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len) return len; } -#endif /* CONFIG_SPI_MASTER */ -static int wm8731_probe(struct platform_device *pdev) +static int __devinit wm8731_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8731_setup_data *setup; struct snd_soc_codec *codec; struct wm8731_priv *wm8731; - int ret = 0; - - pr_info("WM8731 Audio Codec %s", WM8731_VERSION); - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); - if (wm8731 == NULL) { - kfree(codec); + if (wm8731 == NULL) return -ENOMEM; - } - codec->private_data = wm8731; - socdev->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + codec = &wm8731->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8731_spi_write; + codec->dev = &spi->dev; - wm8731_socdev = socdev; - ret = -ENODEV; + spi->dev.driver_data = wm8731; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8731_add_i2c_device(pdev, setup); - } -#endif -#if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - codec->hw_write = (hw_write_t)wm8731_spi_write; - ret = spi_register_driver(&wm8731_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } -#endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return wm8731_register(wm8731); } -/* power down chip */ -static int wm8731_remove(struct platform_device *pdev) +static int __devexit wm8731_spi_remove(struct spi_device *spi) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct wm8731_priv *wm8731 = spi->dev.driver_data; - if (codec->control_data) - wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); + wm8731_unregister(wm8731); + + return 0; +} + +static struct spi_driver wm8731_spi_driver = { + .driver = { + .name = "wm8731", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8731_spi_probe, + .remove = __devexit_p(wm8731_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ - 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(&wm8731_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8731_spi_driver); -#endif - kfree(codec->private_data); - kfree(codec); +static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8731_priv *wm8731; + struct snd_soc_codec *codec; + + wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL); + if (wm8731 == NULL) + 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); +} + +static __devexit int wm8731_i2c_remove(struct i2c_client *client) +{ + struct wm8731_priv *wm8731 = i2c_get_clientdata(client); + wm8731_unregister(wm8731); return 0; } -struct snd_soc_codec_device soc_codec_dev_wm8731 = { - .probe = wm8731_probe, - .remove = wm8731_remove, - .suspend = wm8731_suspend, - .resume = wm8731_resume, +static const struct i2c_device_id wm8731_i2c_id[] = { + { "wm8731", 0 }, + { } }; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); +MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); + +static struct i2c_driver wm8731_i2c_driver = { + .driver = { + .name = "WM8731 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8731_i2c_probe, + .remove = __devexit_p(wm8731_i2c_remove), + .id_table = wm8731_i2c_id, +}; +#endif static int __init wm8731_modinit(void) { - return snd_soc_register_dai(&wm8731_dai); + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8731_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8731_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n", + ret); + } +#endif + return 0; } module_init(wm8731_modinit); static void __exit wm8731_exit(void) { - snd_soc_unregister_dai(&wm8731_dai); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8731_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8731_spi_driver); +#endif } module_exit(wm8731_exit); diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 95190e9c0c14..cd7b806e8ad0 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -34,12 +34,6 @@ #define WM8731_SYSCLK 0 #define WM8731_DAI 0 -struct wm8731_setup_data { - int spi; - int i2c_bus; - unsigned short i2c_address; -}; - extern struct snd_soc_dai wm8731_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8731; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 5997fa68e0d5..b64509b01a49 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0), }; -/* add non dapm controls */ -static int wm8750_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -619,7 +604,7 @@ static int wm8750_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->codec; + 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; @@ -694,6 +679,13 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec, #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8750_dai_ops = { + .hw_params = wm8750_pcm_hw_params, + .digital_mute = wm8750_mute, + .set_fmt = wm8750_set_dai_fmt, + .set_sysclk = wm8750_set_dai_sysclk, +}; + struct snd_soc_dai wm8750_dai = { .name = "WM8750", .playback = { @@ -708,12 +700,7 @@ struct snd_soc_dai wm8750_dai = { .channels_max = 2, .rates = WM8750_RATES, .formats = WM8750_FORMATS,}, - .ops = { - .hw_params = wm8750_pcm_hw_params, - .digital_mute = wm8750_mute, - .set_fmt = wm8750_set_dai_fmt, - .set_sysclk = wm8750_set_dai_sysclk, - }, + .ops = &wm8750_dai_ops, }; EXPORT_SYMBOL_GPL(wm8750_dai); @@ -727,7 +714,7 @@ static void wm8750_work(struct work_struct *work) static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -736,7 +723,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) static int wm8750_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -769,7 +756,7 @@ static int wm8750_resume(struct platform_device *pdev) */ static int wm8750_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8750"; @@ -816,7 +803,8 @@ static int wm8750_init(struct snd_soc_device *socdev) reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); - wm8750_add_controls(codec); + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -850,7 +838,7 @@ static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -931,7 +919,7 @@ err_driver: static int __devinit wm8750_spi_probe(struct spi_device *spi) { struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; codec->control_data = spi; @@ -1003,7 +991,7 @@ static int wm8750_probe(struct platform_device *pdev) } codec->private_data = wm8750; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1057,7 +1045,7 @@ static int run_delayed_work(struct delayed_work *dwork) static int wm8750_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 6c21b50c9375..a6e8f3f7f052 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -51,8 +51,6 @@ #include "wm8753.h" -#define WM8753_VERSION "0.16" - static int caps_charge = 2000; module_param(caps_charge, int, 0); MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); @@ -60,12 +58,6 @@ MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode); -/* codec private data */ -struct wm8753_priv { - unsigned int sysclk; - unsigned int pcmclk; -}; - /* * wm8753 register cache * We can't read the WM8753 register space when we @@ -90,6 +82,14 @@ static const u16 wm8753_reg[] = { 0x0000, 0x0000 }; +/* codec private data */ +struct wm8753_priv { + unsigned int sysclk; + unsigned int pcmclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8753_reg)]; +}; + /* * read wm8753 register cache */ @@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1)) + if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1)) return -1; return cache[reg - 1]; } @@ -109,7 +109,7 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u16 *cache = codec->reg_cache; - if (reg < 1 || reg > 0x3f) + if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1)) return; cache[reg - 1] = value; } @@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]), SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; -/* add non dapm controls */ -static int wm8753_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -927,7 +912,7 @@ static int wm8753_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8753_priv *wm8753 = codec->private_data; u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3; u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f; @@ -1161,7 +1146,7 @@ static int wm8753_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8753_priv *wm8753 = codec->private_data; u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0; u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3; @@ -1316,6 +1301,51 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec, * 3. Voice disabled - HIFI over HIFI * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture */ +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1h_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode1v_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = { + .hw_params = wm8753_pcm_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode2_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = { + .hw_params = wm8753_i2s_hw_params, + .digital_mute = wm8753_mute, + .set_fmt = wm8753_mode3_4_set_dai_fmt, + .set_clkdiv = wm8753_set_dai_clkdiv, + .set_pll = wm8753_set_dai_pll, + .set_sysclk = wm8753_set_dai_sysclk, +}; + static const struct snd_soc_dai wm8753_all_dai[] = { /* DAI HiFi mode 1 */ { .name = "WM8753 HiFi", @@ -1332,14 +1362,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1h_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode1, }, /* DAI Voice mode 1 */ { .name = "WM8753 Voice", @@ -1356,14 +1379,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode1v_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode1, }, /* DAI HiFi mode 2 - dummy */ { .name = "WM8753 HiFi", @@ -1384,14 +1400,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_pcm_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode2_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_voice_mode2, }, /* DAI HiFi mode 3 */ { .name = "WM8753 HiFi", @@ -1408,14 +1417,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode3, }, /* DAI Voice mode 3 - dummy */ { .name = "WM8753 Voice", @@ -1436,14 +1438,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_max = 2, .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, - .ops = { - .hw_params = wm8753_i2s_hw_params, - .digital_mute = wm8753_mute, - .set_fmt = wm8753_mode3_4_set_dai_fmt, - .set_clkdiv = wm8753_set_dai_clkdiv, - .set_pll = wm8753_set_dai_pll, - .set_sysclk = wm8753_set_dai_sysclk, - }, + .ops = &wm8753_dai_ops_hifi_mode4, }, /* DAI Voice mode 4 - dummy */ { .name = "WM8753 Voice", @@ -1451,7 +1446,14 @@ static const struct snd_soc_dai wm8753_all_dai[] = { }, }; -struct snd_soc_dai wm8753_dai[2]; +struct snd_soc_dai wm8753_dai[] = { + { + .name = "WM8753 DAI 0", + }, + { + .name = "WM8753 DAI 1", + }, +}; EXPORT_SYMBOL_GPL(wm8753_dai); static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) @@ -1459,30 +1461,35 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) if (mode < 4) { int playback_active, capture_active, codec_active, pop_wait; void *private_data; + struct list_head list; playback_active = wm8753_dai[0].playback.active; capture_active = wm8753_dai[0].capture.active; codec_active = wm8753_dai[0].active; private_data = wm8753_dai[0].private_data; pop_wait = wm8753_dai[0].pop_wait; + list = wm8753_dai[0].list; wm8753_dai[0] = wm8753_all_dai[mode << 1]; wm8753_dai[0].playback.active = playback_active; wm8753_dai[0].capture.active = capture_active; wm8753_dai[0].active = codec_active; wm8753_dai[0].private_data = private_data; wm8753_dai[0].pop_wait = pop_wait; + wm8753_dai[0].list = list; playback_active = wm8753_dai[1].playback.active; capture_active = wm8753_dai[1].capture.active; codec_active = wm8753_dai[1].active; private_data = wm8753_dai[1].private_data; pop_wait = wm8753_dai[1].pop_wait; + list = wm8753_dai[1].list; wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1]; wm8753_dai[1].playback.active = playback_active; wm8753_dai[1].capture.active = capture_active; wm8753_dai[1].active = codec_active; wm8753_dai[1].private_data = private_data; wm8753_dai[1].pop_wait = pop_wait; + wm8753_dai[1].list = list; } wm8753_dai[0].codec = codec; wm8753_dai[1].codec = codec; @@ -1498,7 +1505,7 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* we only need to suspend if we are a valid card */ if (!codec->card) @@ -1511,7 +1518,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) static int wm8753_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -1524,6 +1531,11 @@ static int wm8753_resume(struct platform_device *pdev) for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) { if (i + 1 == WM8753_RESET) continue; + + /* No point in writing hardware default values back */ + if (cache[i] == wm8753_reg[i]) + continue; + data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); @@ -1542,44 +1554,129 @@ static int wm8753_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8753_codec; + +static int wm8753_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8753_codec) { + dev_err(&pdev->dev, "WM8753 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8753_codec; + codec = wm8753_codec; + + wm8753_set_dai_mode(codec, 0); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8753: failed to create pcms\n"); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8753_snd_controls, + ARRAY_SIZE(wm8753_snd_controls)); + wm8753_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8753: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + +pcm_err: + return ret; +} + /* - * initialise the WM8753 driver - * register the mixer and dsp interfaces with the kernel + * This function forces any delayed work to be queued and run. */ -static int wm8753_init(struct snd_soc_device *socdev) +static int run_delayed_work(struct delayed_work *dwork) +{ + int ret; + + /* cancel any work waiting to be queued. */ + ret = cancel_delayed_work(dwork); + + /* if there was any work waiting then we run it now and + * wait for it's completion */ + if (ret) { + schedule_delayed_work(dwork, 0); + flush_scheduled_work(); + } + return ret; +} + +/* power down chip */ +static int wm8753_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_wm8753 = { + .probe = wm8753_probe, + .remove = wm8753_remove, + .suspend = wm8753_suspend, + .resume = wm8753_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); + +static int wm8753_register(struct wm8753_priv *wm8753) { - struct snd_soc_codec *codec = socdev->codec; - int reg, ret = 0; + int ret, i; + struct snd_soc_codec *codec = &wm8753->codec; + u16 reg; + + if (wm8753_codec) { + dev_err(codec->dev, "Multiple WM8753 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); codec->name = "WM8753"; codec->owner = THIS_MODULE; codec->read = wm8753_read_reg_cache; codec->write = wm8753_write; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); - codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL); - - if (codec->reg_cache == NULL) - return -ENOMEM; - - wm8753_set_dai_mode(codec, 0); + codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache); + codec->reg_cache = &wm8753->reg_cache; + codec->private_data = wm8753; - wm8753_reset(codec); + memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache)); + INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + ret = wm8753_reset(codec); if (ret < 0) { - printk(KERN_ERR "wm8753: failed to create pcms\n"); - goto pcm_err; + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; } /* charge output caps */ wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); - codec->bias_level = SND_SOC_BIAS_STANDBY; schedule_delayed_work(&codec->delayed_work, - msecs_to_jiffies(caps_charge)); + msecs_to_jiffies(caps_charge)); /* set the update bits */ reg = wm8753_read_reg_cache(codec, WM8753_LDAC); @@ -1603,59 +1700,70 @@ static int wm8753_init(struct snd_soc_device *socdev) reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); - wm8753_add_controls(codec); - wm8753_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8753: failed to register card\n"); - goto card_err; + wm8753_codec = codec; + + for (i = 0; i < ARRAY_SIZE(wm8753_dai); i++) + wm8753_dai[i].dev = codec->dev; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; } - return ret; + ret = snd_soc_register_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); + return 0; + +err_codec: + run_delayed_work(&codec->delayed_work); + snd_soc_unregister_codec(codec); +err: + kfree(wm8753); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8753_socdev; +static void wm8753_unregister(struct wm8753_priv *wm8753) +{ + wm8753_set_bias_level(&wm8753->codec, SND_SOC_BIAS_OFF); + run_delayed_work(&wm8753->codec.delayed_work); + snd_soc_unregister_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai)); + snd_soc_unregister_codec(&wm8753->codec); + kfree(wm8753); + wm8753_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -/* - * WM8753 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ - static int wm8753_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8753_priv *wm8753; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + if (wm8753 == NULL) + return -ENOMEM; - ret = wm8753_init(socdev); - if (ret < 0) - pr_err("failed to initialise WM8753\n"); + codec = &wm8753->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8753); - return ret; + codec->dev = &i2c->dev; + + return wm8753_register(wm8753); } static int wm8753_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); - return 0; + struct wm8753_priv *wm8753 = i2c_get_clientdata(client); + wm8753_unregister(wm8753); + return 0; } static const struct i2c_device_id wm8753_i2c_id[] = { @@ -1666,86 +1774,16 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id); static struct i2c_driver wm8753_i2c_driver = { .driver = { - .name = "WM8753 I2C Codec", + .name = "wm8753", .owner = THIS_MODULE, }, .probe = wm8753_i2c_probe, .remove = wm8753_i2c_remove, .id_table = wm8753_i2c_id, }; - -static int wm8753_add_i2c_device(struct platform_device *pdev, - const struct wm8753_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8753_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, "wm8753", 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; - -err_driver: - i2c_del_driver(&wm8753_i2c_driver); - return -ENODEV; -} #endif #if defined(CONFIG_SPI_MASTER) -static int __devinit wm8753_spi_probe(struct spi_device *spi) -{ - struct snd_soc_device *socdev = wm8753_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; - - codec->control_data = spi; - - ret = wm8753_init(socdev); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8753\n"); - - return ret; -} - -static int __devexit wm8753_spi_remove(struct spi_device *spi) -{ - return 0; -} - -static struct spi_driver wm8753_spi_driver = { - .driver = { - .name = "wm8753", - .bus = &spi_bus_type, - .owner = THIS_MODULE, - }, - .probe = wm8753_spi_probe, - .remove = __devexit_p(wm8753_spi_remove), -}; - static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) { struct spi_transfer t; @@ -1769,120 +1807,69 @@ static int wm8753_spi_write(struct spi_device *spi, const char *data, int len) return len; } -#endif - -static int wm8753_probe(struct platform_device *pdev) +static int __devinit wm8753_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8753_setup_data *setup; struct snd_soc_codec *codec; struct wm8753_priv *wm8753; - int ret = 0; - - pr_info("WM8753 Audio Codec %s", WM8753_VERSION); - - setup = socdev->codec_data; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); - if (wm8753 == NULL) { - kfree(codec); + if (wm8753 == NULL) return -ENOMEM; - } - codec->private_data = wm8753; - socdev->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8753_socdev = socdev; - INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); + codec = &wm8753->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8753_spi_write; + codec->dev = &spi->dev; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8753_add_i2c_device(pdev, setup); - } -#endif -#if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - codec->hw_write = (hw_write_t)wm8753_spi_write; - ret = spi_register_driver(&wm8753_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } -#endif + spi->dev.driver_data = wm8753; - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return wm8753_register(wm8753); } -/* - * This function forces any delayed work to be queued and run. - */ -static int run_delayed_work(struct delayed_work *dwork) +static int __devexit wm8753_spi_remove(struct spi_device *spi) { - int ret; - - /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(dwork); - - /* if there was any work waiting then we run it now and - * wait for it's completion */ - if (ret) { - schedule_delayed_work(dwork, 0); - flush_scheduled_work(); - } - return ret; + struct wm8753_priv *wm8753 = spi->dev.driver_data; + wm8753_unregister(wm8753); + return 0; } -/* power down chip */ -static int wm8753_remove(struct platform_device *pdev) -{ - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; +static struct spi_driver wm8753_spi_driver = { + .driver = { + .name = "wm8753", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8753_spi_probe, + .remove = __devexit_p(wm8753_spi_remove), +}; +#endif - if (codec->control_data) - wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - run_delayed_work(&codec->delayed_work); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); +static int __init wm8753_modinit(void) +{ + int ret; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&wm8753_i2c_driver); + ret = i2c_add_driver(&wm8753_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8753 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8753_spi_driver); + ret = spi_register_driver(&wm8753_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8753 SPI driver: %d\n", ret); #endif - kfree(codec->private_data); - kfree(codec); - return 0; } - -struct snd_soc_codec_device soc_codec_dev_wm8753 = { - .probe = wm8753_probe, - .remove = wm8753_remove, - .suspend = wm8753_suspend, - .resume = wm8753_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); - -static int __init wm8753_modinit(void) -{ - return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); -} module_init(wm8753_modinit); static void __exit wm8753_exit(void) { - snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8753_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8753_spi_driver); +#endif } module_exit(wm8753_exit); diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h index f55704ce931b..57b2ba244040 100644 --- a/sound/soc/codecs/wm8753.h +++ b/sound/soc/codecs/wm8753.h @@ -77,12 +77,6 @@ #define WM8753_BIASCTL 0x3d #define WM8753_ADCTL2 0x3f -struct wm8753_setup_data { - int spi; - int i2c_bus; - unsigned short i2c_address; -}; - #define WM8753_PLL1 0 #define WM8753_PLL2 1 diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 6767de10ded0..46c5ea1ff921 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1, }; -/* add non dapm controls */ -static int wm8900_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8900_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new wm8900_dapm_loutput2_control = SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0); @@ -736,7 +720,7 @@ static int wm8900_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 reg; reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60; @@ -1104,6 +1088,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute) (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm8900_dai_ops = { + .hw_params = wm8900_hw_params, + .set_clkdiv = wm8900_set_dai_clkdiv, + .set_pll = wm8900_set_dai_pll, + .set_fmt = wm8900_set_dai_fmt, + .digital_mute = wm8900_digital_mute, +}; + struct snd_soc_dai wm8900_dai = { .name = "WM8900 HiFi", .playback = { @@ -1120,13 +1112,7 @@ struct snd_soc_dai wm8900_dai = { .rates = WM8900_RATES, .formats = WM8900_PCM_FORMATS, }, - .ops = { - .hw_params = wm8900_hw_params, - .set_clkdiv = wm8900_set_dai_clkdiv, - .set_pll = wm8900_set_dai_pll, - .set_fmt = wm8900_set_dai_fmt, - .digital_mute = wm8900_digital_mute, - }, + .ops = &wm8900_dai_ops, }; EXPORT_SYMBOL_GPL(wm8900_dai); @@ -1226,7 +1212,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec, static int wm8900_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8900_priv *wm8900 = codec->private_data; int fll_out = wm8900->fll_out; int fll_in = wm8900->fll_in; @@ -1250,7 +1236,7 @@ static int wm8900_suspend(struct platform_device *pdev, pm_message_t state) static int wm8900_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8900_priv *wm8900 = codec->private_data; u16 *cache; int i, ret; @@ -1288,8 +1274,8 @@ static int wm8900_resume(struct platform_device *pdev) static struct snd_soc_codec *wm8900_codec; -static int wm8900_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static __devinit int wm8900_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct wm8900_priv *wm8900; struct snd_soc_codec *codec; @@ -1388,7 +1374,7 @@ err: return ret; } -static int wm8900_i2c_remove(struct i2c_client *client) +static __devexit int wm8900_i2c_remove(struct i2c_client *client) { snd_soc_unregister_dai(&wm8900_dai); snd_soc_unregister_codec(wm8900_codec); @@ -1414,7 +1400,7 @@ static struct i2c_driver wm8900_i2c_driver = { .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, - .remove = wm8900_i2c_remove, + .remove = __devexit_p(wm8900_i2c_remove), .id_table = wm8900_i2c_id, }; @@ -1430,7 +1416,7 @@ static int wm8900_probe(struct platform_device *pdev) } codec = wm8900_codec; - socdev->codec = codec; + socdev->card->codec = codec; /* Register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -1439,7 +1425,8 @@ static int wm8900_probe(struct platform_device *pdev) goto pcm_err; } - wm8900_add_controls(codec); + snd_soc_add_controls(codec, wm8900_snd_controls, + ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); ret = snd_soc_init_card(socdev); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index bde74546db4a..8cf571f1a803 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume", 0, 63, 0, out_tlv), }; -static int wm8903_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8903_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - - return 0; -} - static const struct snd_kcontrol_new linput_mode_mux = SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum); @@ -1276,7 +1261,7 @@ static int wm8903_startup(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8903_priv *wm8903 = codec->private_data; struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; @@ -1318,7 +1303,7 @@ static void wm8903_shutdown(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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8903_priv *wm8903 = codec->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -1338,7 +1323,7 @@ static int wm8903_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->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm8903_priv *wm8903 = codec->private_data; struct i2c_client *i2c = codec->control_data; int fs = params_rate(params); @@ -1512,6 +1497,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8903_dai_ops = { + .startup = wm8903_startup, + .shutdown = wm8903_shutdown, + .hw_params = wm8903_hw_params, + .digital_mute = wm8903_digital_mute, + .set_fmt = wm8903_set_dai_fmt, + .set_sysclk = wm8903_set_dai_sysclk, +}; + struct snd_soc_dai wm8903_dai = { .name = "WM8903", .playback = { @@ -1528,21 +1522,14 @@ struct snd_soc_dai wm8903_dai = { .rates = WM8903_CAPTURE_RATES, .formats = WM8903_FORMATS, }, - .ops = { - .startup = wm8903_startup, - .shutdown = wm8903_shutdown, - .hw_params = wm8903_hw_params, - .digital_mute = wm8903_digital_mute, - .set_fmt = wm8903_set_dai_fmt, - .set_sysclk = wm8903_set_dai_sysclk - } + .ops = &wm8903_dai_ops, }; EXPORT_SYMBOL_GPL(wm8903_dai); static int wm8903_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -1552,7 +1539,7 @@ static int wm8903_suspend(struct platform_device *pdev, pm_message_t state) static int wm8903_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; @@ -1577,8 +1564,8 @@ static int wm8903_resume(struct platform_device *pdev) static struct snd_soc_codec *wm8903_codec; -static int wm8903_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct wm8903_priv *wm8903; struct snd_soc_codec *codec; @@ -1684,7 +1671,7 @@ err: return ret; } -static int wm8903_i2c_remove(struct i2c_client *client) +static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); @@ -1714,7 +1701,7 @@ static struct i2c_driver wm8903_i2c_driver = { .owner = THIS_MODULE, }, .probe = wm8903_i2c_probe, - .remove = wm8903_i2c_remove, + .remove = __devexit_p(wm8903_i2c_remove), .id_table = wm8903_i2c_id, }; @@ -1728,7 +1715,7 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - socdev->codec = wm8903_codec; + socdev->card->codec = wm8903_codec; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); @@ -1737,8 +1724,9 @@ static int wm8903_probe(struct platform_device *pdev) goto err; } - wm8903_add_controls(socdev->codec); - wm8903_add_widgets(socdev->codec); + snd_soc_add_controls(socdev->card->codec, wm8903_snd_controls, + ARRAY_SIZE(wm8903_snd_controls)); + wm8903_add_widgets(socdev->card->codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -1759,7 +1747,7 @@ err: static int wm8903_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 88ead7f8dd98..032dca22dbd3 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), }; -/* add non-DAPM controls */ -static int wm8971_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8971_snd_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * DAPM Controls */ @@ -546,7 +531,7 @@ static int wm8971_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->codec; + 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; @@ -619,6 +604,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +static struct snd_soc_dai_ops wm8971_dai_ops = { + .hw_params = wm8971_pcm_hw_params, + .digital_mute = wm8971_mute, + .set_fmt = wm8971_set_dai_fmt, + .set_sysclk = wm8971_set_dai_sysclk, +}; + struct snd_soc_dai wm8971_dai = { .name = "WM8971", .playback = { @@ -633,12 +625,7 @@ struct snd_soc_dai wm8971_dai = { .channels_max = 2, .rates = WM8971_RATES, .formats = WM8971_FORMATS,}, - .ops = { - .hw_params = wm8971_pcm_hw_params, - .digital_mute = wm8971_mute, - .set_fmt = wm8971_set_dai_fmt, - .set_sysclk = wm8971_set_dai_sysclk, - }, + .ops = &wm8971_dai_ops, }; EXPORT_SYMBOL_GPL(wm8971_dai); @@ -652,7 +639,7 @@ static void wm8971_work(struct work_struct *work) static int wm8971_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -661,7 +648,7 @@ static int wm8971_suspend(struct platform_device *pdev, pm_message_t state) static int wm8971_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -692,7 +679,7 @@ static int wm8971_resume(struct platform_device *pdev) static int wm8971_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg, ret = 0; codec->name = "WM8971"; @@ -745,7 +732,8 @@ static int wm8971_init(struct snd_soc_device *socdev) reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); - wm8971_add_controls(codec); + snd_soc_add_controls(codec, wm8971_snd_controls, + ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -772,7 +760,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8971_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -873,7 +861,7 @@ static int wm8971_probe(struct platform_device *pdev) } codec->private_data = wm8971; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -908,7 +896,7 @@ static int wm8971_probe(struct platform_device *pdev) static int wm8971_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5b5afc144478..c518c3e5aa3f 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -2,8 +2,7 @@ * wm8990.c -- WM8990 ALSA Soc Audio driver * * Copyright 2008 Wolfson Microelectronics PLC. - * Author: Liam Girdwood - * lg@opensource.wolfsonmicro.com or linux@wolfsonmicro.com + * Author: Liam Girdwood <lrg@slimlogic.co.uk> * * 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 @@ -116,7 +115,7 @@ 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)) - 1); + BUG_ON(reg >= ARRAY_SIZE(wm8990_reg)); return cache[reg]; } @@ -129,7 +128,7 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; /* Reset register and reserved registers are uncached */ - if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1) + if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg)) return; cache[reg] = value; @@ -177,7 +176,9 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value & 0xff; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; int ret; u16 val; @@ -417,21 +418,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME, }; -/* add non dapm controls */ -static int wm8990_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8990_snd_controls[i], codec, - NULL)); - if (err < 0) - return err; - } - return 0; -} - /* * _DAPM_ Controls */ @@ -1177,7 +1163,7 @@ 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->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1); audio1 &= ~WM8990_AIF_WL_MASK; @@ -1346,6 +1332,15 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, * 1. ADC/DAC on Primary Interface * 2. ADC on Primary Interface/DAC on secondary */ +static struct snd_soc_dai_ops wm8990_dai_ops = { + .hw_params = wm8990_hw_params, + .digital_mute = wm8990_mute, + .set_fmt = wm8990_set_dai_fmt, + .set_clkdiv = wm8990_set_dai_clkdiv, + .set_pll = wm8990_set_dai_pll, + .set_sysclk = wm8990_set_dai_sysclk, +}; + struct snd_soc_dai wm8990_dai = { /* ADC/DAC on primary */ .name = "WM8990 ADC/DAC Primary", @@ -1362,21 +1357,14 @@ struct snd_soc_dai wm8990_dai = { .channels_max = 2, .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, - .ops = { - .hw_params = wm8990_hw_params, - .digital_mute = wm8990_mute, - .set_fmt = wm8990_set_dai_fmt, - .set_clkdiv = wm8990_set_dai_clkdiv, - .set_pll = wm8990_set_dai_pll, - .set_sysclk = wm8990_set_dai_sysclk, - }, + .ops = &wm8990_dai_ops, }; EXPORT_SYMBOL_GPL(wm8990_dai); static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; /* we only need to suspend if we are a valid card */ if (!codec->card) @@ -1389,7 +1377,7 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state) static int wm8990_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i; u8 data[2]; u16 *cache = codec->reg_cache; @@ -1417,7 +1405,7 @@ static int wm8990_resume(struct platform_device *pdev) */ static int wm8990_init(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 reg; int ret = 0; @@ -1460,7 +1448,8 @@ static int wm8990_init(struct snd_soc_device *socdev) wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - wm8990_add_controls(codec); + snd_soc_add_controls(codec, wm8990_snd_controls, + ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -1494,7 +1483,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8990_socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int ret; i2c_set_clientdata(i2c, codec); @@ -1593,7 +1582,7 @@ static int wm8990_probe(struct platform_device *pdev) } codec->private_data = wm8990; - socdev->codec = codec; + socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); @@ -1619,7 +1608,7 @@ static int wm8990_probe(struct platform_device *pdev) static int wm8990_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec->control_data) wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c new file mode 100644 index 000000000000..3265817c5c26 --- /dev/null +++ b/sound/soc/codecs/wm9705.c @@ -0,0 +1,415 @@ +/* + * wm9705.c -- ALSA Soc WM9705 codec support + * + * Copyright 2008 Ian Molton <spyro@f2s.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; Version 2 of the License only. + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "wm9705.h" + +/* + * WM9705 register cache + */ +static const u16 wm9705_reg[] = { + 0x6150, 0x8000, 0x8000, 0x8000, /* 0x0 */ + 0x0000, 0x8000, 0x8008, 0x8008, /* 0x8 */ + 0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */ + 0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */ + 0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */ + 0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */ + 0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */ + 0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */ + 0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */ + 0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */ +}; + +static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = { + SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), + SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1), + SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), + SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), + SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), + SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1), + SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), + SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), + SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1), + SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1), + SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1), + SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1), + SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1), + SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0), + SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0), + SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1), +}; + +static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; +static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC", + "Line", "Stereo Mix", "Mono Mix", "Phone"}; + +static const struct soc_enum wm9705_enum_mic = + SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic); +static const struct soc_enum wm9705_enum_rec_l = + SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel); +static const struct soc_enum wm9705_enum_rec_r = + SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel); + +/* Headphone Mixer */ +static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1), + SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1), + SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1), + SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1), + SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1), +}; + +/* Mic source */ +static const struct snd_kcontrol_new wm9705_mic_src_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_mic); + +/* Capture source */ +static const struct snd_kcontrol_new wm9705_capture_selectl_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_l); +static const struct snd_kcontrol_new wm9705_capture_selectr_controls = + SOC_DAPM_ENUM("Route", wm9705_enum_rec_r); + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = { + SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0, + &wm9705_mic_src_controls), + SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectl_controls), + SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9705_capture_selectr_controls), + SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0, + &wm9705_hp_mixer_controls[0], + ARRAY_SIZE(wm9705_hp_mixer_controls)), + SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + SND_SOC_DAPM_OUTPUT("MONOOUT"), + SND_SOC_DAPM_INPUT("PHONE"), + SND_SOC_DAPM_INPUT("LINEINL"), + SND_SOC_DAPM_INPUT("LINEINR"), + SND_SOC_DAPM_INPUT("CDINL"), + SND_SOC_DAPM_INPUT("CDINR"), + SND_SOC_DAPM_INPUT("PCBEEP"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), +}; + +/* Audio map + * WM9705 has no switches to disable the route from the inputs to the HP mixer + * so in order to prevent active inputs from forcing the audio outputs to be + * constantly enabled, we use the mutes on those inputs to simulate such + * controls. + */ +static const struct snd_soc_dapm_route audio_map[] = { + /* HP mixer */ + {"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"}, + {"HP Mixer", "CD Playback Switch", "CD PGA"}, + {"HP Mixer", "Mic Playback Switch", "Mic PGA"}, + {"HP Mixer", "Phone Playback Switch", "Phone PGA"}, + {"HP Mixer", "Line Playback Switch", "Line PGA"}, + {"HP Mixer", NULL, "Left DAC"}, + {"HP Mixer", NULL, "Right DAC"}, + + /* mono mixer */ + {"Mono Mixer", NULL, "HP Mixer"}, + + /* outputs */ + {"Headphone PGA", NULL, "HP Mixer"}, + {"HPOUTL", NULL, "Headphone PGA"}, + {"HPOUTR", NULL, "Headphone PGA"}, + {"Line out PGA", NULL, "HP Mixer"}, + {"LOUT", NULL, "Line out PGA"}, + {"ROUT", NULL, "Line out PGA"}, + {"Mono PGA", NULL, "Mono Mixer"}, + {"MONOOUT", NULL, "Mono PGA"}, + + /* inputs */ + {"CD PGA", NULL, "CDINL"}, + {"CD PGA", NULL, "CDINR"}, + {"Line PGA", NULL, "LINEINL"}, + {"Line PGA", NULL, "LINEINR"}, + {"Phone PGA", NULL, "PHONE"}, + {"Mic Source", "Mic 1", "MIC1"}, + {"Mic Source", "Mic 2", "MIC2"}, + {"Mic PGA", NULL, "Mic Source"}, + {"PCBEEP PGA", NULL, "PCBEEP"}, + + /* Left capture selector */ + {"Left Capture Source", "Mic", "Mic Source"}, + {"Left Capture Source", "CD", "CDINL"}, + {"Left Capture Source", "Line", "LINEINL"}, + {"Left Capture Source", "Stereo Mix", "HP Mixer"}, + {"Left Capture Source", "Mono Mix", "HP Mixer"}, + {"Left Capture Source", "Phone", "PHONE"}, + + /* Right capture source */ + {"Right Capture Source", "Mic", "Mic Source"}, + {"Right Capture Source", "CD", "CDINR"}, + {"Right Capture Source", "Line", "LINEINR"}, + {"Right Capture Source", "Stereo Mix", "HP Mixer"}, + {"Right Capture Source", "Mono Mix", "HP Mixer"}, + {"Right Capture Source", "Phone", "PHONE"}, + + {"ADC PGA", NULL, "Left Capture Source"}, + {"ADC PGA", NULL, "Right Capture Source"}, + + /* ADC's */ + {"Left ADC", NULL, "ADC PGA"}, + {"Right ADC", NULL, "ADC PGA"}, +}; + +static int wm9705_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, + ARRAY_SIZE(wm9705_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +/* We use a register cache to enhance read performance. */ +static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + switch (reg) { + case AC97_RESET: + case AC97_VENDOR_ID1: + case AC97_VENDOR_ID2: + return soc_ac97_ops.read(codec->ac97, reg); + default: + reg = reg >> 1; + + if (reg >= (ARRAY_SIZE(wm9705_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg < (ARRAY_SIZE(wm9705_reg))) + cache[reg] = val; + + return 0; +} + +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +#define WM9705_AC97_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) + +static struct snd_soc_dai_ops wm9705_dai_ops = { + .prepare = ac97_prepare, +}; + +struct snd_soc_dai wm9705_dai[] = { + { + .name = "AC97 HiFi", + .ac97_control = 1, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &wm9705_dai_ops, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9705_AC97_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + } +}; +EXPORT_SYMBOL_GPL(wm9705_dai); + +static int wm9705_reset(struct snd_soc_codec *codec) +{ + if (soc_ac97_ops.reset) { + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) == wm9705_reg[0]) + return 0; /* Success */ + } + + return -EIO; +} + +static int wm9705_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "WM9705 SoC Audio Codec\n"); + + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), + GFP_KERNEL); + if (socdev->card->codec == NULL) + return -ENOMEM; + codec = socdev->card->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9705_reg); + codec->reg_cache_step = 2; + + codec->name = "WM9705"; + codec->owner = THIS_MODULE; + codec->dai = wm9705_dai; + codec->num_dai = ARRAY_SIZE(wm9705_dai); + codec->write = ac97_write; + codec->read = ac97_read; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register AC97 codec\n"); + goto codec_err; + } + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + ret = wm9705_reset(codec); + if (ret) + goto reset_err; + + snd_soc_add_controls(codec, wm9705_snd_ac97_controls, + ARRAY_SIZE(wm9705_snd_ac97_controls)); + wm9705_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm9705: failed to register card\n"); + goto pcm_err; + } + + return 0; + +reset_err: + snd_soc_free_pcms(socdev); +pcm_err: + snd_soc_free_ac97_codec(codec); +codec_err: + kfree(codec->reg_cache); +cache_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int wm9705_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->reg_cache); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9705 = { + .probe = wm9705_soc_probe, + .remove = wm9705_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705); + +MODULE_DESCRIPTION("ASoC WM9705 driver"); +MODULE_AUTHOR("Ian Molton"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h new file mode 100644 index 000000000000..d380f110f9e2 --- /dev/null +++ b/sound/soc/codecs/wm9705.h @@ -0,0 +1,14 @@ +/* + * wm9705.h -- WM9705 Soc Audio driver + */ + +#ifndef _WM9705_H +#define _WM9705_H + +#define WM9705_DAI_AC97_HIFI 0 +#define WM9705_DAI_AC97_AUX 1 + +extern struct snd_soc_dai wm9705_dai[2]; +extern struct snd_soc_codec_device soc_codec_dev_wm9705; + +#endif diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index af83d629078a..765cf1e7369e 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), }; -/* add non dapm controls */ -static int wm9712_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9712_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path. @@ -467,7 +452,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, else { reg = reg >> 1; - if (reg > (ARRAY_SIZE(wm9712_reg))) + if (reg >= (ARRAY_SIZE(wm9712_reg))) return -EIO; return cache[reg]; @@ -481,7 +466,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg <= (ARRAY_SIZE(wm9712_reg))) + if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; return 0; @@ -493,7 +478,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int reg; u16 vra; @@ -514,7 +499,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); @@ -532,6 +517,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops wm9712_dai_ops_hifi = { + .prepare = ac97_prepare, +}; + +static struct snd_soc_dai_ops wm9712_dai_ops_aux = { + .prepare = ac97_aux_prepare, +}; + struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", @@ -548,8 +541,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_prepare,}, + .ops = &wm9712_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -559,8 +551,7 @@ struct snd_soc_dai wm9712_dai[] = { .channels_max = 1, .rates = WM9712_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare,}, + .ops = &wm9712_dai_ops_aux, } }; EXPORT_SYMBOL_GPL(wm9712_dai); @@ -607,7 +598,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; @@ -616,7 +607,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev, static int wm9712_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; int i, ret; u16 *cache = codec->reg_cache; @@ -652,10 +643,11 @@ static int wm9712_soc_probe(struct platform_device *pdev) printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), + GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL); @@ -698,7 +690,8 @@ static int wm9712_soc_probe(struct platform_device *pdev) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - wm9712_add_controls(codec); + snd_soc_add_controls(codec, wm9712_snd_ac97_controls, + ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) { @@ -718,15 +711,15 @@ codec_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int wm9712_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index f3ca8aaf0139..523bad077fa0 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -32,7 +32,6 @@ struct wm9713_priv { u32 pll_in; /* PLL input frequency */ - u32 pll_out; /* PLL output frequency */ }; static unsigned int ac97_read(struct snd_soc_codec *codec, @@ -190,21 +189,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; -/* add non dapm controls */ -static int wm9713_add_controls(struct snd_soc_codec *codec) -{ - int err, i; - - for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm9713_snd_ac97_controls[i], - codec, NULL)); - if (err < 0) - return err; - } - return 0; -} - /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -636,7 +620,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec, else { reg = reg >> 1; - if (reg > (ARRAY_SIZE(wm9713_reg))) + if (reg >= (ARRAY_SIZE(wm9713_reg))) return -EIO; return cache[reg]; @@ -650,7 +634,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, if (reg < 0x7c) soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; - if (reg <= (ARRAY_SIZE(wm9713_reg))) + if (reg < (ARRAY_SIZE(wm9713_reg))) cache[reg] = val; return 0; @@ -738,13 +722,13 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, struct _pll_div pll_div; /* turn PLL off ? */ - if (freq_in == 0 || freq_out == 0) { + if (freq_in == 0) { /* disable PLL power and select ext source */ reg = ac97_read(codec, AC97_HANDSET_RATE); ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); reg = ac97_read(codec, AC97_EXTENDED_MID); ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); - wm9713->pll_out = 0; + wm9713->pll_in = 0; return 0; } @@ -788,7 +772,6 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); reg = ac97_read(codec, AC97_HANDSET_RATE); ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); - wm9713->pll_out = freq_out; wm9713->pll_in = freq_in; /* wait 10ms AC97 link frames for the link to stabilise */ @@ -957,13 +940,14 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - u16 status; + u16 status, rate; /* Gracefully shut down the voice interface. */ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; - ac97_write(codec, AC97_HANDSET_RATE, 0x0280); + rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, 0x0F80); + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); ac97_write(codec, AC97_EXTENDED_MID, status); } @@ -1021,6 +1005,27 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ SNDRV_PCM_FORMAT_S24_LE) +static struct snd_soc_dai_ops wm9713_dai_ops_hifi = { + .prepare = ac97_hifi_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_aux = { + .prepare = ac97_aux_prepare, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, +}; + +static struct snd_soc_dai_ops wm9713_dai_ops_voice = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown, + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, +}; + struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", @@ -1037,10 +1042,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_hifi_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_hifi, }, { .name = "AC97 Aux", @@ -1050,10 +1052,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 1, .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .prepare = ac97_aux_prepare, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll,}, + .ops = &wm9713_dai_ops_aux, }, { .name = "WM9713 Voice", @@ -1069,14 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = { .channels_max = 2, .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, - .ops = { - .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, - .set_clkdiv = wm9713_set_dai_clkdiv, - .set_pll = wm9713_set_dai_pll, - .set_fmt = wm9713_set_dai_fmt, - .set_tristate = wm9713_set_dai_tristate, - }, + .ops = &wm9713_dai_ops_voice, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); @@ -1132,7 +1124,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; u16 reg; /* Disable everything except touchpanel - that will be handled @@ -1150,7 +1142,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev, static int wm9713_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; struct wm9713_priv *wm9713 = codec->private_data; int i, ret; u16 *cache = codec->reg_cache; @@ -1164,8 +1156,8 @@ static int wm9713_soc_resume(struct platform_device *pdev) wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* do we need to re-start the PLL ? */ - if (wm9713->pll_out) - wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + if (wm9713->pll_in) + wm9713_set_pll(codec, 0, wm9713->pll_in, 0); /* only synchronise the codec if warm reset failed */ if (ret == 0) { @@ -1191,10 +1183,11 @@ static int wm9713_soc_probe(struct platform_device *pdev) printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); - socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), + GFP_KERNEL); + if (socdev->card->codec == NULL) return -ENOMEM; - codec = socdev->codec; + codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL); @@ -1245,7 +1238,8 @@ static int wm9713_soc_probe(struct platform_device *pdev) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - wm9713_add_controls(codec); + snd_soc_add_controls(codec, wm9713_snd_ac97_controls, + ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) @@ -1265,15 +1259,15 @@ priv_err: kfree(codec->reg_cache); cache_err: - kfree(socdev->codec); - socdev->codec = NULL; + kfree(socdev->card->codec); + socdev->card->codec = NULL; return ret; } static int wm9713_soc_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index b502741692d6..bd7392c9657e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -20,7 +20,7 @@ config SND_DAVINCI_SOC_EVM config SND_DAVINCI_SOC_SFFSDR tristate "SoC Audio support for SFFSDR" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR + depends on SND_DAVINCI_SOC && MACH_SFFSDR select SND_DAVINCI_SOC_I2S select SND_SOC_PCM3008 select SFFSDR_FPGA diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 54851f318568..9b90b347007c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -186,7 +186,8 @@ static int __init evm_init(void) platform_set_drvdata(evm_snd_device, &evm_snd_devdata); evm_snd_devdata.dev = &evm_snd_device->dev; - evm_snd_device->dev.platform_data = &evm_snd_data; + platform_device_add_data(evm_snd_device, &evm_snd_data, + sizeof(evm_snd_data)); ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, ARRAY_SIZE(evm_snd_resources)); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 0fee779e3c76..ffdb9439d3d8 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev, #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, +}; + struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, @@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = { .channels_max = 2, .rates = DAVINCI_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = davinci_i2s_startup, - .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params, - .set_fmt = davinci_i2s_set_dai_fmt, - }, + .ops = &davinci_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(davinci_i2s_dai); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 366049d8578c..7af3b5b3a53d 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -286,7 +286,7 @@ static int davinci_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -struct snd_pcm_ops davinci_pcm_ops = { +static struct snd_pcm_ops davinci_pcm_ops = { .open = davinci_pcm_open, .close = davinci_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 4935d1bcbd8d..40eccfe9e358 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,9 @@ #include <asm/dma.h> #include <asm/mach-types.h> +#ifdef CONFIG_SFFSDR_FPGA #include <asm/plat-sffsdr/sffsdr-fpga.h> +#endif #include <mach/mcbsp.h> #include <mach/edma.h> @@ -34,31 +36,45 @@ #include "davinci-pcm.h" #include "davinci-i2s.h" +/* + * CLKX and CLKR are the inputs for the Sample Rate Generator. + * FSX and FSR are outputs, driven by the sample Rate Generator. + */ +#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ + SND_SOC_DAIFMT_CBM_CFS | \ + SND_SOC_DAIFMT_IB_NF) + static int sffsdr_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + 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; int fs; int ret = 0; - /* Set cpu DAI configuration: - * CLKX and CLKR are the inputs for the Sample Rate Generator. - * FSX and FSR are outputs, driven by the sample Rate Generator. */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_RIGHT_J | - SND_SOC_DAIFMT_CBM_CFS | - SND_SOC_DAIFMT_IB_NF); - if (ret < 0) - return ret; - /* Fsref can be 32000, 44100 or 48000. */ fs = params_rate(params); +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); + if (ret < 0) + return ret; + pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else return sffsdr_fpga_set_codec_fs(fs); +#endif } static struct snd_soc_ops sffsdr_ops = { @@ -127,7 +143,8 @@ static int __init sffsdr_init(void) platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata); sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev; - sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data; + platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data, + sizeof(sffsdr_snd_data)); ret = platform_device_add_resources(sffsdr_snd_device, sffsdr_snd_resources, diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 95c12b26fe37..9fc908283371 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,17 +1,18 @@ config SND_SOC_OF_SIMPLE tristate +# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers +# for the SSI and the Elo DMA controller. You will still need to select +# a platform driver and a codec driver. config SND_SOC_MPC8610 - bool "ALSA SoC support for the MPC8610 SOC" - depends on MPC8610_HPCD - default y if MPC8610 - help - Say Y if you want to add support for codecs attached to the SSI - device on an MPC8610. + tristate + depends on MPC8610 config SND_SOC_MPC8610_HPCD - bool "ALSA SoC support for the Freescale MPC8610 HPCD board" - depends on SND_SOC_MPC8610 + tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" + # I2C is necessary for the CS4270 driver + depends on MPC8610_HPCD && I2C + select SND_SOC_MPC8610 select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 035da4afec34..f85134c86387 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -2,10 +2,13 @@ obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o # MPC8610 HPCD Machine Support -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o +snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o +obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o # MPC8610 Platform Support -obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o +snd-soc-fsl-ssi-objs := fsl_ssi.o +snd-soc-fsl-dma-objs := fsl_dma.o +obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 64993eda5679..b3eb8570cd7b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -142,7 +142,8 @@ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_JOINT_DUPLEX, + SNDRV_PCM_INFO_JOINT_DUPLEX | + SNDRV_PCM_INFO_PAUSE, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, @@ -464,11 +465,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); + dma_private->link[i].next = cpu_to_be64(temp_link); temp_link += sizeof(struct fsl_dma_link_descriptor); } @@ -525,79 +522,9 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) * This function obtains hardware parameters about the opened stream and * programs the DMA controller accordingly. * - * Note that due to a quirk of the SSI's STX register, the target address - * for the DMA operations depends on the sample size. So we don't program - * the dest_addr (for playback -- source_addr for capture) fields in the - * link descriptors here. We do that in fsl_dma_prepare() - */ -static int fsl_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct fsl_dma_private *dma_private = runtime->private_data; - - dma_addr_t temp_addr; /* Pointer to next period */ - - unsigned int i; - - /* Get all the parameters we need */ - size_t buffer_size = params_buffer_bytes(hw_params); - size_t period_size = params_period_bytes(hw_params); - - /* Initialize our DMA tracking variables */ - dma_private->period_size = period_size; - dma_private->num_periods = params_periods(hw_params); - dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; - dma_private->dma_buf_next = dma_private->dma_buf_phys + - (NUM_DMA_LINKS * period_size); - if (dma_private->dma_buf_next >= dma_private->dma_buf_end) - dma_private->dma_buf_next = dma_private->dma_buf_phys; - - /* - * The actual address in STX0 (destination for playback, source for - * capture) is based on the sample size, but we don't know the sample - * size in this function, so we'll have to adjust that later. See - * comments in fsl_dma_prepare(). - * - * The DMA controller does not have a cache, so the CPU does not - * need to tell it to flush its cache. However, the DMA - * controller does need to tell the CPU to flush its cache. - * That's what the SNOOP bit does. - * - * Also, even though the DMA controller supports 36-bit addressing, for - * simplicity we currently support only 32-bit addresses for the audio - * buffer itself. - */ - temp_addr = substream->dma_buffer.addr; - - for (i = 0; i < NUM_DMA_LINKS; i++) { - struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - - link->count = cpu_to_be32(period_size); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - link->source_addr = cpu_to_be32(temp_addr); - else - link->dest_addr = cpu_to_be32(temp_addr); - - temp_addr += period_size; - } - - return 0; -} - -/** - * fsl_dma_prepare - prepare the DMA registers for playback. - * - * This function is called after the specifics of the audio data are known, - * i.e. snd_pcm_runtime is initialized. - * - * In this function, we finish programming the registers of the DMA - * controller that are dependent on the sample size. - * - * One of the drawbacks with big-endian is that when copying integers of - * different sizes to a fixed-sized register, the address to which the - * integer must be copied is dependent on the size of the integer. + * One drawback of big-endian is that when copying integers of different + * sizes to a fixed-sized register, the address to which the integer must be + * copied is dependent on the size of the integer. * * For example, if P is the address of a 32-bit register, and X is a 32-bit * integer, then X should be copied to address P. However, if X is a 16-bit @@ -613,22 +540,58 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * and 8 bytes at a time). So we do not support packed 24-bit samples. * 24-bit data must be padded to 32 bits. */ -static int fsl_dma_prepare(struct snd_pcm_substream *substream) +static int fsl_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; + + /* Number of bits per sample */ + unsigned int sample_size = + snd_pcm_format_physical_width(params_format(hw_params)); + + /* Number of bytes per frame */ + unsigned int frame_size = 2 * (sample_size / 8); + + /* Bus address of SSI STX register */ + dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys; + + /* Size of the DMA buffer, in bytes */ + size_t buffer_size = params_buffer_bytes(hw_params); + + /* Number of bytes per period */ + size_t period_size = params_period_bytes(hw_params); + + /* Pointer to next period */ + dma_addr_t temp_addr = substream->dma_buffer.addr; + + /* Pointer to DMA controller */ struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; - u32 mr; + + u32 mr; /* DMA Mode Register */ + unsigned int i; - dma_addr_t ssi_sxx_phys; /* Bus address of SSI STX register */ - unsigned int frame_size; /* Number of bytes per frame */ - ssi_sxx_phys = dma_private->ssi_sxx_phys; + /* Initialize our DMA tracking variables */ + dma_private->period_size = period_size; + dma_private->num_periods = params_periods(hw_params); + dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size; + dma_private->dma_buf_next = dma_private->dma_buf_phys + + (NUM_DMA_LINKS * period_size); + + if (dma_private->dma_buf_next >= dma_private->dma_buf_end) + /* This happens if the number of periods == NUM_DMA_LINKS */ + dma_private->dma_buf_next = dma_private->dma_buf_phys; mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK | CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK); - switch (runtime->sample_bits) { + /* Due to a quirk of the SSI's STX register, the target address + * for the DMA operations depends on the sample size. So we calculate + * that offset here. While we're at it, also tell the DMA controller + * how much data to transfer per sample. + */ + switch (sample_size) { case 8: mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1; ssi_sxx_phys += 3; @@ -641,12 +604,12 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4; break; default: + /* We should never get here */ dev_err(substream->pcm->card->dev, - "unsupported sample size %u\n", runtime->sample_bits); + "unsupported sample size %u\n", sample_size); return -EINVAL; } - frame_size = runtime->frame_bits / 8; /* * BWC should always be a multiple of the frame size. BWC determines * how many bytes are sent/received before the DMA controller checks the @@ -655,7 +618,6 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) * capture, the receive FIFO is triggered when it contains one frame, so * we want to receive one frame at a time. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) mr |= CCSR_DMA_MR_BWC(2 * frame_size); else @@ -663,16 +625,48 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream) out_be32(&dma_channel->mr, mr); - /* - * Program the address of the DMA transfer to/from the SSI. - */ for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + link->count = cpu_to_be32(period_size); + + /* Even though the DMA controller supports 36-bit addressing, + * for simplicity we allow only 32-bit addresses for the audio + * buffer itself. This was enforced in fsl_dma_new() with the + * DMA mask. + * + * The snoop bit tells the DMA controller whether it should tell + * the ECM to snoop during a read or write to an address. For + * audio, we use DMA to transfer data between memory and an I/O + * device (the SSI's STX0 or SRX0 register). Snooping is only + * needed if there is a cache, so we need to snoop memory + * addresses only. For playback, that means we snoop the source + * but not the destination. For capture, we snoop the + * destination but not the source. + * + * Note that failing to snoop properly is unlikely to cause + * cache incoherency if the period size is larger than the + * size of L1 cache. This is because filling in one period will + * flush out the data for the previous period. So if you + * increased period_bytes_min to a large enough size, you might + * get more performance by not snooping, and you'll still be + * okay. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link->source_addr = cpu_to_be32(temp_addr); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_addr = cpu_to_be32(ssi_sxx_phys); - else + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + } else { link->source_addr = cpu_to_be32(ssi_sxx_phys); + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP); + + link->dest_addr = cpu_to_be32(temp_addr); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + } + + temp_addr += period_size; } return 0; @@ -808,7 +802,6 @@ static struct snd_pcm_ops fsl_dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = fsl_dma_hw_params, .hw_free = fsl_dma_hw_free, - .prepare = fsl_dma_prepare, .pointer = fsl_dma_pointer, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6d6eb71dc1d..169bca295b78 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -72,6 +72,7 @@ * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened + * @asynchronous: 0=synchronous mode, 1=asynchronous mode * @cpu_dai: the CPU DAI for this device * @dev_attr: the sysfs device attribute structure * @stats: SSI statistics @@ -86,6 +87,7 @@ struct fsl_ssi_private { struct device *dev; unsigned int playback; unsigned int capture; + int asynchronous; struct snd_soc_dai cpu_dai; struct device_attribute dev_attr; @@ -301,9 +303,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * * FIXME: Little-endian samples require a different shift dir */ - clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK, - CCSR_SSI_SCR_TFR_CLK_DIS | - CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN); + clrsetbits_be32(&ssi->scr, + CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN, + CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE + | (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN)); out_be32(&ssi->stcr, CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 | @@ -382,10 +385,15 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, first_runtime->rate, first_runtime->rate); - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - first_runtime->sample_bits, - first_runtime->sample_bits); + /* If we're in synchronous mode, then we need to constrain + * the sample size as well. We don't support independent sample + * rates in asynchronous mode. + */ + if (!ssi_private->asynchronous) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); ssi_private->second_stream = substream; } @@ -400,7 +408,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, } /** - * fsl_ssi_prepare: prepare the SSI. + * fsl_ssi_hw_params - program the sample size * * Most of the SSI registers have been programmed in the startup function, * but the word length must be programmed here. Unfortunately, programming @@ -412,23 +420,27 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; - - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + struct fsl_ssi_private *ssi_private = cpu_dai->private_data; if (substream == ssi_private->first_stream) { - u32 wl; + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + unsigned int sample_size = + snd_pcm_format_width(params_format(hw_params)); + u32 wl = CCSR_SSI_SxCCR_WL(sample_size); /* The SSI should always be disabled at this points (SSIEN=0) */ - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); /* In synchronous mode, the SSI uses STCCR for capture */ - clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + !ssi_private->asynchronous) + clrsetbits_be32(&ssi->stccr, + CCSR_SSI_SxCCR_WL_MASK, wl); + else + clrsetbits_be32(&ssi->srccr, + CCSR_SSI_SxCCR_WL_MASK, wl); } return 0; @@ -452,28 +464,33 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + long timeout = jiffies + 10; + setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - /* - * I think we need this delay to allow time for the SSI - * to put data into its FIFO. Without it, ALSA starts - * to complain about overruns. + /* Wait until the SSI has filled its FIFO. Without this + * delay, ALSA complains about overruns. When the FIFO + * is full, the DMA controller initiates its first + * transfer. Until then, however, the DMA's DAR + * register is zero, which translates to an + * out-of-bounds pointer. This makes ALSA think an + * overrun has occurred. */ - mdelay(1); + while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && + (jiffies < timeout)); + if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) + return -EIO; } break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) clrbits32(&ssi->scr, CCSR_SSI_SCR_TE); @@ -563,6 +580,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * fsl_ssi_dai_template: template CPU DAI for the SSI */ +static struct snd_soc_dai_ops fsl_ssi_dai_ops = { + .startup = fsl_ssi_startup, + .hw_params = fsl_ssi_hw_params, + .shutdown = fsl_ssi_shutdown, + .trigger = fsl_ssi_trigger, + .set_sysclk = fsl_ssi_set_sysclk, + .set_fmt = fsl_ssi_set_fmt, +}; + static struct snd_soc_dai fsl_ssi_dai_template = { .playback = { /* The SSI does not support monaural audio. */ @@ -577,14 +603,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, - .ops = { - .startup = fsl_ssi_startup, - .prepare = fsl_ssi_prepare, - .shutdown = fsl_ssi_shutdown, - .trigger = fsl_ssi_trigger, - .set_sysclk = fsl_ssi_set_sysclk, - .set_fmt = fsl_ssi_set_fmt, - }, + .ops = &fsl_ssi_dai_ops, }; /** @@ -654,6 +673,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info) ssi_private->ssi_phys = ssi_info->ssi_phys; ssi_private->irq = ssi_info->irq; ssi_private->dev = ssi_info->dev; + ssi_private->asynchronous = ssi_info->asynchronous; ssi_private->dev->driver_data = fsl_ssi_dai; @@ -704,6 +724,14 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai) } EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai); +static int __init fsl_ssi_init(void) +{ + printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n"); + + return 0; +} +module_init(fsl_ssi_init); + MODULE_AUTHOR("Timur Tabi <timur@freescale.com>"); MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 83b44d700e33..eade01feaab6 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -208,6 +208,7 @@ struct ccsr_ssi { * ssi_phys: physical address of the SSI registers * irq: IRQ of this SSI * dev: struct device, used to create the sysfs statistics file + * asynchronous: 0=synchronous mode, 1=asynchronous mode */ struct fsl_ssi_info { unsigned int id; @@ -215,6 +216,7 @@ struct fsl_ssi_info { dma_addr_t ssi_phys; unsigned int irq; struct device *dev; + int asynchronous; }; struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info); diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 9eb1ce185bd0..3aa729df27b5 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -468,6 +468,16 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) /** * psc_i2s_dai_template: template CPU Digital Audio Interface */ +static struct snd_soc_dai_ops psc_i2s_dai_ops = { + .startup = psc_i2s_startup, + .hw_params = psc_i2s_hw_params, + .hw_free = psc_i2s_hw_free, + .shutdown = psc_i2s_shutdown, + .trigger = psc_i2s_trigger, + .set_sysclk = psc_i2s_set_sysclk, + .set_fmt = psc_i2s_set_fmt, +}; + static struct snd_soc_dai psc_i2s_dai_template = { .playback = { .channels_min = 2, @@ -481,15 +491,7 @@ static struct snd_soc_dai psc_i2s_dai_template = { .rates = PSC_I2S_RATES, .formats = PSC_I2S_FORMATS, }, - .ops = { - .startup = psc_i2s_startup, - .hw_params = psc_i2s_hw_params, - .hw_free = psc_i2s_hw_free, - .shutdown = psc_i2s_shutdown, - .trigger = psc_i2s_trigger, - .set_sysclk = psc_i2s_set_sysclk, - .set_fmt = psc_i2s_set_fmt, - }, + .ops = &psc_i2s_dai_ops, }; /* --------------------------------------------------------------------- diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index bcec3f60bad9..ef67d1cdffe7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -183,16 +183,6 @@ static struct snd_soc_ops mpc8610_hpcd_ops = { }; /** - * mpc8610_hpcd_machine: ASoC machine data - */ -static struct snd_soc_card mpc8610_hpcd_machine = { - .probe = mpc8610_hpcd_machine_probe, - .remove = mpc8610_hpcd_machine_remove, - .name = "MPC8610 HPCD", - .num_links = 1, -}; - -/** * mpc8610_hpcd_probe: OF probe function for the fabric driver * * This function gets called when an SSI node is found in the device tree. @@ -363,6 +353,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } ssi_info.irq = machine_data->ssi_irq; + /* Do we want to use asynchronous mode? */ + ssi_info.asynchronous = + of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0; + if (ssi_info.asynchronous) + dev_info(&ofdev->dev, "using asynchronous mode\n"); /* Map the global utilities registers. */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); @@ -455,7 +450,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, machine_data->dai.codec_dai = &cs4270_dai; /* The codec_dai we want */ machine_data->dai.ops = &mpc8610_hpcd_ops; - mpc8610_hpcd_machine.dai_link = &machine_data->dai; + machine_data->machine.probe = mpc8610_hpcd_machine_probe; + machine_data->machine.remove = mpc8610_hpcd_machine_remove; + machine_data->machine.name = "MPC8610 HPCD"; + machine_data->machine.num_links = 1; + machine_data->machine.dai_link = &machine_data->dai; /* Allocate a new audio platform device structure */ sound_device = platform_device_alloc("soc-audio", -1); @@ -465,7 +464,7 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, goto error; } - machine_data->sound_devdata.card = &mpc8610_hpcd_machine; + machine_data->sound_devdata.card = &machine_data->machine; machine_data->sound_devdata.codec_dev = &soc_codec_device_cs4270; machine_data->machine.platform = &fsl_soc_platform; diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4f7f04014585..675732e724d5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -8,7 +8,7 @@ config SND_OMAP_SOC_MCBSP config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" - depends on SND_OMAP_SOC && MACH_NOKIA_N810 + depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C select SND_OMAP_SOC_MCBSP select OMAP_MUX select SND_SOC_TLV320AIC3X @@ -17,7 +17,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" - depends on SND_OMAP_SOC && MACH_OMAP_OSK + depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC23 help @@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA select SND_SOC_TWL4030 help Say Y if you want to add support for SoC audio on the OMAP3 Pandora. + +config SND_OMAP_SOC_OMAP3_BEAGLE + tristate "SoC Audio support for OMAP3 Beagle" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the Beagleboard. + + diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 76fedd96e365..0c9e4ac37660 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o +snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o @@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(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 diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 25593fee9121..a6d1178ce128 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -40,6 +40,13 @@ #define N810_HEADSET_AMP_GPIO 10 #define N810_SPEAKER_AMP_GPIO 101 +enum { + N810_JACK_DISABLED, + N810_JACK_HP, + N810_JACK_HS, + N810_JACK_MIC, +}; + static struct clk *sys_clkout2; static struct clk *sys_clkout2_src; static struct clk *func96m_clk; @@ -50,15 +57,32 @@ static int n810_dmic_func; static void n810_ext_control(struct snd_soc_codec *codec) { + int hp = 0, line1l = 0; + + switch (n810_jack_func) { + case N810_JACK_HS: + line1l = 1; + case N810_JACK_HP: + hp = 1; + break; + case N810_JACK_MIC: + line1l = 1; + break; + } + if (n810_spk_func) snd_soc_dapm_enable_pin(codec, "Ext Spk"); else snd_soc_dapm_disable_pin(codec, "Ext Spk"); - if (n810_jack_func) + if (hp) snd_soc_dapm_enable_pin(codec, "Headphone Jack"); else snd_soc_dapm_disable_pin(codec, "Headphone Jack"); + if (line1l) + snd_soc_dapm_enable_pin(codec, "LINE1L"); + else + snd_soc_dapm_disable_pin(codec, "LINE1L"); if (n810_dmic_func) snd_soc_dapm_enable_pin(codec, "DMic"); @@ -72,7 +96,7 @@ static int n810_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); @@ -229,7 +253,7 @@ static const struct snd_soc_dapm_route audio_map[] = { }; static const char *spk_function[] = {"Off", "On"}; -static const char *jack_function[] = {"Off", "Headphone"}; +static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"}; static const char *input_function[] = {"ADC", "Digital Mic"}; static const struct soc_enum n810_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), @@ -248,20 +272,23 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = { static int n810_aic33_init(struct snd_soc_codec *codec) { - int i, err; + int err; /* Not connected */ snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); snd_soc_dapm_nc_pin(codec, "HPLCOM"); snd_soc_dapm_nc_pin(codec, "HPRCOM"); + snd_soc_dapm_nc_pin(codec, "MIC3L"); + snd_soc_dapm_nc_pin(codec, "MIC3R"); + snd_soc_dapm_nc_pin(codec, "LINE1R"); + snd_soc_dapm_nc_pin(codec, "LINE2L"); + snd_soc_dapm_nc_pin(codec, "LINE2R"); /* Add N810 specific controls */ - for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&aic33_n810_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, aic33_n810_controls, + ARRAY_SIZE(aic33_n810_controls)); + if (err < 0) + return err; /* Add N810 specific widgets */ snd_soc_dapm_new_controls(codec, aic33_dapm_widgets, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index ec5e18a78758..d6882be33452 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -302,6 +302,10 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->spcr1 |= RINTM(3); regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; + if (cpu_is_omap2430() || cpu_is_omap34xx()) { + regs->xccr = DXENDLY(1) | XDMAEN; + regs->rccr = RFULL_CYCLE | RDMAEN; + } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -457,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return err; } +static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { + .startup = omap_mcbsp_dai_startup, + .shutdown = omap_mcbsp_dai_shutdown, + .trigger = omap_mcbsp_dai_trigger, + .hw_params = omap_mcbsp_dai_hw_params, + .set_fmt = omap_mcbsp_dai_set_dai_fmt, + .set_clkdiv = omap_mcbsp_dai_set_clkdiv, + .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, +}; + #define OMAP_MCBSP_DAI_BUILDER(link_id) \ { \ .name = "omap-mcbsp-dai-"#link_id, \ @@ -473,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ - .ops = { \ - .startup = omap_mcbsp_dai_startup, \ - .shutdown = omap_mcbsp_dai_shutdown, \ - .trigger = omap_mcbsp_dai_trigger, \ - .hw_params = omap_mcbsp_dai_hw_params, \ - .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ - .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ - .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ - }, \ + .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ } diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b0362dfd5b71..8e1431cb46bb 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -175,9 +175,10 @@ 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; + unsigned long flags; int ret = 0; - spin_lock_irq(&prtd->lock); + spin_lock_irqsave(&prtd->lock, flags); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -195,7 +196,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd) default: ret = -EINVAL; } - spin_unlock_irq(&prtd->lock); + spin_unlock_irqrestore(&prtd->lock, flags); return ret; } @@ -264,7 +265,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -struct snd_pcm_ops omap_pcm_ops = { +static struct snd_pcm_ops omap_pcm_ops = { .open = omap_pcm_open, .close = omap_pcm_close, .ioctl = snd_pcm_lib_ioctl, diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fcc2f5d9a878..fe282d4ef422 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -143,7 +143,7 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (internal)", NULL), SND_SOC_DAPM_MIC("Mic (external)", NULL), SND_SOC_DAPM_LINE("Line In", NULL), }; @@ -155,16 +155,33 @@ static const struct snd_soc_dapm_route omap3pandora_out_map[] = { }; static const struct snd_soc_dapm_route omap3pandora_in_map[] = { - {"INL", NULL, "Line In"}, - {"INR", NULL, "Line In"}, - {"INL", NULL, "Mic (Internal)"}, - {"INR", NULL, "Mic (external)"}, + {"AUXL", NULL, "Line In"}, + {"AUXR", NULL, "Line In"}, + + {"MAINMIC", NULL, "Mic Bias 1"}, + {"Mic Bias 1", NULL, "Mic (internal)"}, + + {"SUBMIC", NULL, "Mic Bias 2"}, + {"Mic Bias 2", NULL, "Mic (external)"}, }; static int omap3pandora_out_init(struct snd_soc_codec *codec) { int ret; + /* All TWL4030 output pins are floating */ + 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, "HSOL"); + snd_soc_dapm_nc_pin(codec, "HSOR"); + snd_soc_dapm_nc_pin(codec, "CARKITL"); + snd_soc_dapm_nc_pin(codec, "CARKITR"); + snd_soc_dapm_nc_pin(codec, "HFL"); + snd_soc_dapm_nc_pin(codec, "HFR"); + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, ARRAY_SIZE(omap3pandora_out_dapm_widgets)); if (ret < 0) @@ -180,18 +197,11 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) { int ret; - /* All TWL4030 output pins are floating */ - 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, "HSOL"), - snd_soc_dapm_nc_pin(codec, "HSOR"), - snd_soc_dapm_nc_pin(codec, "CARKITL"), - snd_soc_dapm_nc_pin(codec, "CARKITR"), - snd_soc_dapm_nc_pin(codec, "HFL"), - snd_soc_dapm_nc_pin(codec, "HFR"), + /* Not comnnected */ + snd_soc_dapm_nc_pin(codec, "HSMIC"); + snd_soc_dapm_nc_pin(codec, "CARKITMIC"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC0"); + snd_soc_dapm_nc_pin(codec, "DIGIMIC1"); ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, ARRAY_SIZE(omap3pandora_in_dapm_widgets)); @@ -251,10 +261,9 @@ static int __init omap3pandora_soc_init(void) { int ret; - if (!machine_is_omap3_pandora()) { - pr_debug(PREFIX "Not OMAP3 Pandora\n"); + if (!machine_is_omap3_pandora()) return -ENODEV; - } + pr_info("OMAP3 Pandora SoC init\n"); ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index cd41a948df7b..a952a4eb3361 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -186,13 +186,6 @@ static int __init osk_soc_init(void) return -ENODEV; } - if (clk_get_usecount(tlv320aic23_mclk) > 0) { - /* MCLK is already in use */ - printk(KERN_WARNING - "MCLK in use at %d Hz. We change it to %d Hz\n", - (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); - } - /* * Configure 12 MHz output on MCLK. */ @@ -205,9 +198,8 @@ static int __init osk_soc_init(void) } } - printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n", - (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK, - clk_get_usecount(tlv320aic23_mclk)); + printk(KERN_INFO "MCLK = %d [%d]\n", + (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK); return 0; err1: diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index ad97836818b1..10f1c867f11d 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -28,6 +28,7 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -38,6 +39,8 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +static struct snd_soc_card snd_soc_sdp3430; + static int sdp3430_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -81,17 +84,126 @@ static struct snd_soc_ops sdp3430_ops = { .hw_params = sdp3430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +/* Headset jack detection gpios */ +static struct snd_soc_jack_gpio hs_jack_gpios[] = { + { + .gpio = (OMAP_MAX_GPIO_LINES + 2), + .name = "hsdet-gpio", + .report = SND_JACK_HEADSET, + .debounce_time = 200, + }, +}; + +/* SDP3430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp3430_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), +}; + +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 Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, +}; + +static int sdp3430_twl4030_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add SDP3430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets, + ARRAY_SIZE(sdp3430_twl4030_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP3430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP3430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + /* TWL4030 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AUXL"); + snd_soc_dapm_nc_pin(codec, "AUXR"); + 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); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(&snd_soc_sdp3430, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + return ret; +} + /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai, + .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; /* Audio machine driver */ -static struct snd_soc_machine snd_soc_machine_sdp3430 = { +static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", .platform = &omap_soc_platform, .dai_link = &sdp3430_dai, @@ -100,7 +212,7 @@ static struct snd_soc_machine snd_soc_machine_sdp3430 = { /* Audio subsystem */ static struct snd_soc_device sdp3430_snd_devdata = { - .machine = &snd_soc_machine_sdp3430, + .card = &snd_soc_sdp3430, .codec_dev = &soc_codec_dev_twl4030, }; @@ -142,6 +254,9 @@ module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { + snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + platform_device_unregister(sdp3430_snd_device); } module_exit(sdp3430_soc_exit); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index f82e10699471..5998ab366e83 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -61,6 +61,24 @@ config SND_PXA2XX_SOC_TOSA Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C6000x models (Tosa). +config SND_PXA2XX_SOC_E740 + tristate "SoC AC97 Audio support for e740" + depends on SND_PXA2XX_SOC && MACH_E740 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e740 PDA + +config SND_PXA2XX_SOC_E750 + tristate "SoC AC97 Audio support for e750" + depends on SND_PXA2XX_SOC && MACH_E750 + select SND_SOC_WM9705 + select SND_PXA2XX_SOC_AC97 + help + Say Y if you want to add support for SoC audio on the + toshiba e750 PDA + config SND_PXA2XX_SOC_E800 tristate "SoC AC97 Audio support for e800" depends on SND_PXA2XX_SOC && MACH_E800 @@ -97,3 +115,12 @@ config SND_SOC_ZYLONITE help Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. + +config SND_PXA2XX_SOC_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 08a9f2797729..8ed881c5e5cc 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -13,17 +13,23 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o +snd-soc-e740-objs := e740_wm9705.o +snd-soc-e750-objs := e750_wm9705.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o snd-soc-zylonite-objs := zylonite.o +snd-soc-mioa701-objs := mioa701_wm9713.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o +obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o +obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o +obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1ba25a559524..02263e5d8f03 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -16,6 +16,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/timer.h> +#include <linux/i2c.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/gpio.h> @@ -100,7 +101,7 @@ static void corgi_ext_control(struct snd_soc_codec *codec) static int corgi_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ corgi_ext_control(codec); @@ -275,18 +276,16 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { */ static int corgi_wm8731_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "LLINEIN"); snd_soc_dapm_nc_pin(codec, "RLINEIN"); /* Add corgi specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8731_corgi_controls, + ARRAY_SIZE(wm8731_corgi_controls)); + if (err < 0) + return err; /* Add corgi specific widgets */ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, @@ -317,19 +316,44 @@ static struct snd_soc_card snd_soc_corgi = { .num_links = 1, }; -/* corgi audio private data */ -static struct wm8731_setup_data corgi_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; - /* corgi audio subsystem */ static struct snd_soc_device corgi_snd_devdata = { .card = &snd_soc_corgi, .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &corgi_wm8731_setup, }; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8731 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8731_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} + static struct platform_device *corgi_snd_device; static int __init corgi_init(void) @@ -340,6 +364,10 @@ static int __init corgi_init(void) machine_is_husky())) return -ENODEV; + ret = wm8731_i2c_register(); + if (ret != 0) + return ret; + corgi_snd_device = platform_device_alloc("soc-audio", -1); if (!corgi_snd_device) return -ENOMEM; diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c new file mode 100644 index 000000000000..7cd2f89d7b10 --- /dev/null +++ b/sound/soc/pxa/e740_wm9705.c @@ -0,0 +1,211 @@ +/* + * e740-wm9705.c -- SoC audio for e740 + * + * Copyright 2007 (c) Ian Molton <spyro@f2s.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; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + + +#define E740_AUDIO_OUT 1 +#define E740_AUDIO_IN 2 + +static int e740_audio_power; + +static void e740_sync_audio_power(int status) +{ + gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status); + gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0); + gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0); +} + +static int e740_mic_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_IN; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_IN; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static int e740_output_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + e740_audio_power |= E740_AUDIO_OUT; + else if (event & SND_SOC_DAPM_POST_PMD) + e740_audio_power &= ~E740_AUDIO_OUT; + + e740_sync_audio_power(e740_audio_power); + + return 0; +} + +static const struct snd_soc_dapm_widget e740_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_output_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Output Amp", NULL, "LOUT"}, + {"Output Amp", NULL, "ROUT"}, + {"Output Amp", NULL, "MONOOUT"}, + + {"Speaker", NULL, "Output Amp"}, + {"Headphone Jack", NULL, "Output Amp"}, + + {"MIC1", NULL, "Mic Amp"}, + {"Mic Amp", NULL, "Mic (Internal)"}, +}; + +static int e740_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "HPOUTL"); + snd_soc_dapm_nc_pin(codec, "HPOUTR"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e740_dapm_widgets, + ARRAY_SIZE(e740_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e740_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e740_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e740 = { + .name = "Toshiba e740", + .platform = &pxa2xx_soc_platform, + .dai_link = e740_dai, + .num_links = ARRAY_SIZE(e740_dai), +}; + +static struct snd_soc_device e740_snd_devdata = { + .card = &e740, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e740_snd_device; + +static int __init e740_init(void) +{ + int ret; + + if (!machine_is_e740()) + return -ENODEV; + + ret = gpio_request(GPIO_E740_MIC_ON, "Mic amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E740_AMP_ON, "Output amp"); + if (ret) + goto free_mic_amp_gpio; + + ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power"); + if (ret) + goto free_op_amp_gpio; + + /* Disable audio */ + ret = gpio_direction_output(GPIO_E740_MIC_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_AMP_ON, 0); + if (ret) + goto free_apwr_gpio; + ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1); + if (ret) + goto free_apwr_gpio; + + e740_snd_device = platform_device_alloc("soc-audio", -1); + if (!e740_snd_device) { + ret = -ENOMEM; + goto free_apwr_gpio; + } + + platform_set_drvdata(e740_snd_device, &e740_snd_devdata); + e740_snd_devdata.dev = &e740_snd_device->dev; + ret = platform_device_add(e740_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e740_snd_device); +free_apwr_gpio: + gpio_free(GPIO_E740_WM9705_nAVDD2); +free_op_amp_gpio: + gpio_free(GPIO_E740_AMP_ON); +free_mic_amp_gpio: + gpio_free(GPIO_E740_MIC_ON); + + return ret; +} + +static void __exit e740_exit(void) +{ + platform_device_unregister(e740_snd_device); +} + +module_init(e740_init); +module_exit(e740_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e740"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c new file mode 100644 index 000000000000..8dceccc5e059 --- /dev/null +++ b/sound/soc/pxa/e750_wm9705.c @@ -0,0 +1,187 @@ +/* + * e750-wm9705.c -- SoC audio for e750 + * + * Copyright 2007 (c) Ian Molton <spyro@f2s.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; version 2 ONLY. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <mach/audio.h> +#include <mach/eseries-gpio.h> + +#include <asm/mach-types.h> + +#include "../codecs/wm9705.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int e750_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1); + + return 0; +} + +static int e750_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E750_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Amp", NULL, "HPOUTL"}, + {"Headphone Amp", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal)"}, +}; + +static int e750_ac97_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_nc_pin(codec, "LOUT"); + snd_soc_dapm_nc_pin(codec, "ROUT"); + snd_soc_dapm_nc_pin(codec, "PHONE"); + snd_soc_dapm_nc_pin(codec, "LINEINL"); + snd_soc_dapm_nc_pin(codec, "LINEINR"); + snd_soc_dapm_nc_pin(codec, "CDINL"); + snd_soc_dapm_nc_pin(codec, "CDINR"); + snd_soc_dapm_nc_pin(codec, "PCBEEP"); + snd_soc_dapm_nc_pin(codec, "MIC2"); + + snd_soc_dapm_new_controls(codec, e750_dapm_widgets, + ARRAY_SIZE(e750_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e750_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI], + .init = e750_ac97_init, + /* use ops to check startup state */ + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX], + }, +}; + +static struct snd_soc_card e750 = { + .name = "Toshiba e750", + .platform = &pxa2xx_soc_platform, + .dai_link = e750_dai, + .num_links = ARRAY_SIZE(e750_dai), +}; + +static struct snd_soc_device e750_snd_devdata = { + .card = &e750, + .codec_dev = &soc_codec_dev_wm9705, +}; + +static struct platform_device *e750_snd_device; + +static int __init e750_init(void) +{ + int ret; + + if (!machine_is_e750()) + return -ENODEV; + + ret = gpio_request(GPIO_E750_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + e750_snd_device = platform_device_alloc("soc-audio", -1); + if (!e750_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } + + platform_set_drvdata(e750_snd_device, &e750_snd_devdata); + e750_snd_devdata.dev = &e750_snd_device->dev; + ret = platform_device_add(e750_snd_device); + + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e750_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E750_SPK_AMP_OFF); +free_hp_amp_gpio: + gpio_free(GPIO_E750_HP_AMP_OFF); + + return ret; +} + +static void __exit e750_exit(void) +{ + platform_device_unregister(e750_snd_device); + gpio_free(GPIO_E750_SPK_AMP_OFF); + gpio_free(GPIO_E750_HP_AMP_OFF); +} + +module_init(e750_init); +module_exit(e750_exit); + +/* Module information */ +MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); +MODULE_DESCRIPTION("ALSA SoC driver for e750"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 2e3386dfa0f0..bc019cdce429 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -1,8 +1,6 @@ /* * e800-wm9712.c -- SoC audio for e800 * - * Based on tosa.c - * * Copyright 2007 (c) Ian Molton <spyro@f2s.com> * * This program is free software; you can redistribute it and/or modify it @@ -13,7 +11,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/device.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> @@ -21,23 +19,85 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/audio.h> +#include <mach/eseries-gpio.h> #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -static struct snd_soc_card e800; +static int e800_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 1); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_SPK_AMP_ON, 0); -static struct snd_soc_dai_link e800_dai[] = { + return 0; +} + +static int e800_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (event & SND_SOC_DAPM_PRE_PMU) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 0); + else if (event & SND_SOC_DAPM_POST_PMD) + gpio_set_value(GPIO_E800_HP_AMP_OFF, 1); + + return 0; +} + +static const struct snd_soc_dapm_widget e800_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic (Internal1)", NULL), + SND_SOC_DAPM_MIC("Mic (Internal2)", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0, + e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HPOUTL"}, + {"Headphone Jack", NULL, "HPOUTR"}, + {"Headphone Jack", NULL, "Headphone Amp"}, + + {"Speaker Amp", NULL, "MONOOUT"}, + {"Speaker", NULL, "Speaker Amp"}, + + {"MIC1", NULL, "Mic (Internal1)"}, + {"MIC2", NULL, "Mic (Internal2)"}, +}; + +static int e800_ac97_init(struct snd_soc_codec *codec) { - .name = "AC97 Aux", - .stream_name = "AC97 Aux", - .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], - .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], -}, + snd_soc_dapm_new_controls(codec, e800_dapm_widgets, + ARRAY_SIZE(e800_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link e800_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], + .init = e800_ac97_init, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], + }, }; static struct snd_soc_card e800 = { @@ -61,6 +121,22 @@ static int __init e800_init(void) if (!machine_is_e800()) return -ENODEV; + ret = gpio_request(GPIO_E800_HP_AMP_OFF, "Headphone amp"); + if (ret) + return ret; + + ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp"); + if (ret) + goto free_hp_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1); + if (ret) + goto free_spk_amp_gpio; + + ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1); + if (ret) + goto free_spk_amp_gpio; + e800_snd_device = platform_device_alloc("soc-audio", -1); if (!e800_snd_device) return -ENOMEM; @@ -69,8 +145,15 @@ static int __init e800_init(void) e800_snd_devdata.dev = &e800_snd_device->dev; ret = platform_device_add(e800_snd_device); - if (ret) - platform_device_put(e800_snd_device); + if (!ret) + return 0; + +/* Fail gracefully */ + platform_device_put(e800_snd_device); +free_spk_amp_gpio: + gpio_free(GPIO_E800_SPK_AMP_ON); +free_hp_amp_gpio: + gpio_free(GPIO_E800_HP_AMP_OFF); return ret; } @@ -78,6 +161,8 @@ static int __init e800_init(void) static void __exit e800_exit(void) { platform_device_unregister(e800_snd_device); + gpio_free(GPIO_E800_SPK_AMP_ON); + gpio_free(GPIO_E800_HP_AMP_OFF); } module_init(e800_init); @@ -86,4 +171,4 @@ module_exit(e800_exit); /* Module information */ MODULE_AUTHOR("Ian Molton <spyro@f2s.com>"); MODULE_DESCRIPTION("ALSA SoC driver for e800"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c new file mode 100644 index 000000000000..19eda8bbfdaf --- /dev/null +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -0,0 +1,250 @@ +/* + * Handles the Mitac mioa701 SoC system + * + * Copyright (C) 2008 Robert Jarzmik + * + * 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 in version 2 of the License. + * + * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * This is a little schema of the sound interconnections : + * + * Sagem X200 Wolfson WM9713 + * +--------+ +-------------------+ Rear Speaker + * | | | | /-+ + * | +--->----->---+MONOIN SPKL+--->----+-+ | + * | GSM | | | | | | + * | +--->----->---+PCBEEP SPKR+--->----+-+ | + * | CHIP | | | \-+ + * | +---<-----<---+MONO | + * | | | | Front Speaker + * +--------+ | | /-+ + * | HPL+--->----+-+ | + * | | | | | + * | OUT3+--->----+-+ | + * | | \-+ + * | | + * | | Front Micro + * | | + + * | MIC1+-----<--+o+ + * | | + + * +-------------------+ --- + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> + +#include <asm/mach-types.h> +#include <mach/audio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/ac97_codec.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" +#include "../codecs/wm9713.h" + +#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x) + +#define AC97_GPIO_PULL 0x58 + +/* Use GPIO8 for rear speaker amplifier */ +static int rear_amp_power(struct snd_soc_codec *codec, int power) +{ + unsigned short reg; + + if (power) { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15)); + } else { + reg = snd_soc_read(codec, AC97_GPIO_CFG); + snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100); + reg = snd_soc_read(codec, AC97_GPIO_PULL); + snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15)); + } + + return 0; +} + +static int rear_amp_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kctl, int event) +{ + struct snd_soc_codec *codec = widget->codec; + + return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); +} + +/* mioa701 machine dapm widgets */ +static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Front Speaker", NULL), + SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event), + SND_SOC_DAPM_MIC("Headset", NULL), + 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("Front Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Call Mic */ + {"Mic Bias", NULL, "Front Mic"}, + {"MIC1", NULL, "Mic Bias"}, + + /* Headset Mic */ + {"LINEL", NULL, "Headset Mic"}, + {"LINER", NULL, "Headset Mic"}, + + /* GSM Module */ + {"MONOIN", NULL, "GSM Line Out"}, + {"PCBEEP", NULL, "GSM Line Out"}, + {"GSM Line In", NULL, "MONO"}, + + /* headphone connected to HPL, HPR */ + {"Headset", NULL, "HPL"}, + {"Headset", NULL, "HPR"}, + + /* front speaker connected to HPL, OUT3 */ + {"Front Speaker", NULL, "HPL"}, + {"Front Speaker", NULL, "OUT3"}, + + /* rear speaker connected to SPKL, SPKR */ + {"Rear Speaker", NULL, "SPKL"}, + {"Rear Speaker", NULL, "SPKR"}, +}; + +static int mioa701_wm9713_init(struct snd_soc_codec *codec) +{ + unsigned short reg; + + /* Add mioa701 specific widgets */ + snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets)); + + /* Set up mioa701 specific audio path audio_mapnects */ + snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map)); + + /* Prepare GPIO8 for rear speaker amplifier */ + reg = codec->read(codec, AC97_GPIO_CFG); + codec->write(codec, AC97_GPIO_CFG, reg | 0x0100); + + /* Prepare MIC input */ + reg = codec->read(codec, AC97_3D_CONTROL); + codec->write(codec, AC97_3D_CONTROL, reg | 0xc000); + + snd_soc_dapm_enable_pin(codec, "Front Speaker"); + snd_soc_dapm_enable_pin(codec, "Rear Speaker"); + snd_soc_dapm_enable_pin(codec, "Front Mic"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_ops mioa701_ops; + +static struct snd_soc_dai_link mioa701_dai[] = { + { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], + .init = mioa701_wm9713_init, + .ops = &mioa701_ops, + }, + { + .name = "AC97 Aux", + .stream_name = "AC97 Aux", + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], + .ops = &mioa701_ops, + }, +}; + +static struct snd_soc_card mioa701 = { + .name = "MioA701", + .platform = &pxa2xx_soc_platform, + .dai_link = mioa701_dai, + .num_links = ARRAY_SIZE(mioa701_dai), +}; + +static struct snd_soc_device mioa701_snd_devdata = { + .card = &mioa701, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *mioa701_snd_device; + +static int mioa701_wm9713_probe(struct platform_device *pdev) +{ + int ret; + + if (!machine_is_mioa701()) + return -ENODEV; + + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + "lead to overheating and possible destruction of your device." + "Do not use without a good knowledge of mio's board design!\n"); + + mioa701_snd_device = platform_device_alloc("soc-audio", -1); + if (!mioa701_snd_device) + return -ENOMEM; + + platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata); + mioa701_snd_devdata.dev = &mioa701_snd_device->dev; + + ret = platform_device_add(mioa701_snd_device); + if (!ret) + return 0; + + platform_device_put(mioa701_snd_device); + return ret; +} + +static int __devexit mioa701_wm9713_remove(struct platform_device *pdev) +{ + platform_device_unregister(mioa701_snd_device); + return 0; +} + +static struct platform_driver mioa701_wm9713_driver = { + .probe = mioa701_wm9713_probe, + .remove = __devexit_p(mioa701_wm9713_remove), + .driver = { + .name = "mioa701-wm9713", + .owner = THIS_MODULE, + }, +}; + +static int __init mioa701_asoc_init(void) +{ + return platform_driver_register(&mioa701_wm9713_driver); +} + +static void __exit mioa701_asoc_exit(void) +{ + platform_driver_unregister(&mioa701_wm9713_driver); +} + +module_init(mioa701_asoc_init); +module_exit(mioa701_asoc_exit); + +/* Module information */ +MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); +MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4a9cf3083af0..48a73f64500b 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -55,7 +55,7 @@ static void palm27x_ext_control(struct snd_soc_codec *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->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ palm27x_ext_control(codec); @@ -146,19 +146,16 @@ static const struct snd_kcontrol_new palm27x_controls[] = { static int palm27x_ac97_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "OUT3"); snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add palm27x specific controls */ - for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&palm27x_controls[i], - codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, palm27x_controls, + ARRAY_SIZE(palm27x_controls)); + if (err < 0) + return err; /* add palm27x specific widgets */ snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets, diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 6e9827189fff..ef7c6c8dc8f1 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/timer.h> +#include <linux/i2c.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -77,7 +78,7 @@ static void poodle_ext_control(struct snd_soc_codec *codec) static int poodle_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ poodle_ext_control(codec); @@ -240,19 +241,17 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = { */ static int poodle_wm8731_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "LLINEIN"); snd_soc_dapm_nc_pin(codec, "RLINEIN"); snd_soc_dapm_enable_pin(codec, "MICIN"); /* Add poodle specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8731_poodle_controls, + ARRAY_SIZE(wm8731_poodle_controls)); + if (err < 0) + return err; /* Add poodle specific widgets */ snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets, @@ -283,17 +282,42 @@ static struct snd_soc_card snd_soc_poodle = { .num_links = 1, }; -/* poodle audio private data */ -static struct wm8731_setup_data poodle_wm8731_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8731 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8731_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8731", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} /* poodle audio subsystem */ static struct snd_soc_device poodle_snd_devdata = { .card = &snd_soc_poodle, .codec_dev = &soc_codec_dev_wm8731, - .codec_data = &poodle_wm8731_setup, }; static struct platform_device *poodle_snd_device; @@ -305,6 +329,10 @@ static int __init poodle_init(void) if (!machine_is_poodle()) return -ENODEV; + ret = wm8731_i2c_register(); + if (ret != 0) + return ret; + locomo_gpio_set_dir(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_AMP_ON, 0); /* should we mute HP at startup - burning power ?*/ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 73cb6b4c2f2d..b0bf40973d5b 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -1,4 +1,3 @@ -#define DEBUG /* * pxa-ssp.c -- ALSA Soc Audio Layer * @@ -21,6 +20,8 @@ #include <linux/clk.h> #include <linux/io.h> +#include <asm/irq.h> + #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -221,9 +222,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ); - if (ret < 0) - return ret; + priv->dev.port = cpu_dai->id + 1; + priv->dev.irq = NO_IRQ; + clk_enable(priv->dev.ssp->clk); ssp_disable(&priv->dev); } return ret; @@ -238,7 +239,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { ssp_disable(&priv->dev); - ssp_exit(&priv->dev); + clk_disable(priv->dev.ssp->clk); } } @@ -298,7 +299,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int val; u32 sscr0 = ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n", @@ -326,7 +327,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; ssp_set_scr(&priv->dev, 1); - sscr0 |= SSCR0_ADC; + sscr0 |= SSCR0_ACS; break; default: return -ENODEV; @@ -520,9 +521,20 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, u32 sscr1; u32 sspsp; + /* check if we need to change anything at all */ + if (priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + dev_err(&ssp->pdev->dev, + "can't change hardware dai format: stream is in use"); + return -EINVAL; + } + /* reset port settings */ sscr0 = ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC); + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -545,18 +557,18 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_MOD | SSCR0_PSP; + 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_FSRT; + sspsp |= SSPSP_SFRMP; break; case SND_SOC_DAIFMT_NB_IF: - sspsp |= SSPSP_SFRMP | SSPSP_FSRT; break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SFRMP; + sspsp |= SSPSP_SCMODE(3); break; default: return -EINVAL; @@ -642,34 +654,65 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif sscr0 |= SSCR0_DataSize(16); - if (params_channels(params) > 1) - sscr0 |= SSCR0_EDSS; break; case SNDRV_PCM_FORMAT_S24_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - /* we must be in network mode (2 slots) for 24 bit stereo */ break; case SNDRV_PCM_FORMAT_S32_LE: sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - /* we must be in network mode (2 slots) for 32 bit stereo */ break; } ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - /* Cleared when the DAI format is set */ - sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width); + sspsp = ssp_read_reg(ssp, SSPSP); + + if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) && + (width == 16)) { + /* This is a special case where the bitclk is 64fs + * and we're not dealing with 2*32 bits of audio + * samples. + * + * The SSP values used for that are all found out by + * trying and failing a lot; some of the registers + * needed for that mode are only available on PXA3xx. + */ + +#ifdef CONFIG_PXA3xx + if (!cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_SFRMWDTH(width * 2); + sspsp |= SSPSP_SFRMDLY(width * 4); + sspsp |= SSPSP_EDMYSTOP(3); + sspsp |= SSPSP_DMYSTOP(3); + sspsp |= SSPSP_DMYSTRT(1); +#else + return -EINVAL; +#endif + } else { + /* The frame width is the width the LRCLK is + * asserted for; the delay is expressed in + * half cycle units. We need the extra cycle + * because the data starts clocking out one BCLK + * after LRCLK changes polarity. + */ + sspsp |= SSPSP_SFRMWDTH(width + 1); + sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_DMYSTRT(1); + } + ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; } - /* We always use a network mode so we always require TDM slots + /* When we use a network mode, we always require TDM slots * - complain loudly and fail if they've not been set up yet. */ - if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) { + if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) { dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); return -EINVAL; } @@ -751,7 +794,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id, "SoC audio"); + priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); if (priv->dev.ssp == NULL) { ret = -ENODEV; goto err_priv; @@ -782,6 +825,19 @@ static void pxa_ssp_remove(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +static struct snd_soc_dai_ops pxa_ssp_dai_ops = { + .startup = pxa_ssp_startup, + .shutdown = pxa_ssp_shutdown, + .trigger = pxa_ssp_trigger, + .hw_params = pxa_ssp_hw_params, + .set_sysclk = pxa_ssp_set_dai_sysclk, + .set_clkdiv = pxa_ssp_set_dai_clkdiv, + .set_pll = pxa_ssp_set_dai_pll, + .set_fmt = pxa_ssp_set_dai_fmt, + .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, + .set_tristate = pxa_ssp_set_dai_tristate, +}; + struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp1", @@ -802,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp2", .id = 1, @@ -833,18 +878,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp3", @@ -865,18 +899,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, { .name = "pxa2xx-ssp4", @@ -897,18 +920,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, - .ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, - }, + .ops = &pxa_ssp_dai_ops, }, }; EXPORT_SYMBOL_GPL(pxa_ssp_dai); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 812c2b4d3e07..01c21c6cdbbc 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -106,13 +106,13 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) static int pxa2xx_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - return pxa2xx_ac97_hw_probe(pdev); + return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); } static void pxa2xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { - pxa2xx_ac97_hw_remove(pdev); + pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); } static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, @@ -164,6 +164,18 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { + .hw_params = pxa2xx_ac97_hw_params, +}; + +static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { + .hw_params = pxa2xx_ac97_hw_aux_params, +}; + +static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { + .hw_params = pxa2xx_ac97_hw_mic_params, +}; + /* * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. @@ -189,8 +201,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 2, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_params,}, + .ops = &pxa_ac97_hifi_dai_ops, }, { .name = "pxa2xx-ac97-aux", @@ -208,8 +219,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_aux_params,}, + .ops = &pxa_ac97_aux_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -221,23 +231,52 @@ struct snd_soc_dai pxa_ac97_dai[] = { .channels_max = 1, .rates = PXA2XX_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = pxa2xx_ac97_hw_mic_params,}, + .ops = &pxa_ac97_mic_dai_ops, }, }; EXPORT_SYMBOL_GPL(pxa_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); -static int __init pxa_ac97_init(void) +static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev) { + int i; + + for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++) + pxa_ac97_dai[i].dev = &pdev->dev; + + /* 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 + * and running. + */ return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); } + +static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); + + return 0; +} + +static struct platform_driver pxa2xx_ac97_driver = { + .probe = pxa2xx_ac97_dev_probe, + .remove = __devexit_p(pxa2xx_ac97_dev_remove), + .driver = { + .name = "pxa2xx-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa_ac97_init(void) +{ + return platform_driver_register(&pxa2xx_ac97_driver); +} module_init(pxa_ac97_init); static void __exit pxa_ac97_exit(void) { - snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); + platform_driver_unregister(&pxa2xx_ac97_driver); } module_exit(pxa_ac97_exit); diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 517991fb1099..e6c24408c5f9 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -25,20 +25,11 @@ #include <mach/hardware.h> #include <mach/pxa-regs.h> -#include <mach/pxa2xx-gpio.h> #include <mach/audio.h> #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" -struct pxa2xx_gpio { - u32 sys; - u32 rx; - u32 tx; - u32 clk; - u32 frm; -}; - /* * I2S Controller Register and Bit Definitions */ @@ -106,21 +97,6 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { DCMD_BURST32 | DCMD_WIDTH4, }; -static struct pxa2xx_gpio gpio_bus[] = { - { /* I2S SoC Slave */ - .rx = GPIO29_SDATA_IN_I2S_MD, - .tx = GPIO30_SDATA_OUT_I2S_MD, - .clk = GPIO28_BITCLK_IN_I2S_MD, - .frm = GPIO31_SYNC_I2S_MD, - }, - { /* I2S SoC Master */ - .rx = GPIO29_SDATA_IN_I2S_MD, - .tx = GPIO30_SDATA_OUT_I2S_MD, - .clk = GPIO28_BITCLK_OUT_I2S_MD, - .frm = GPIO31_SYNC_I2S_MD, - }, -}; - static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -181,9 +157,6 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (clk_id != PXA2XX_I2S_SYSCLK) return -ENODEV; - if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT) - pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys); - return 0; } @@ -194,10 +167,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); - pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); - pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); - pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); pxa_i2s_wait(); @@ -335,6 +304,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .startup = pxa2xx_i2s_startup, + .shutdown = pxa2xx_i2s_shutdown, + .trigger = pxa2xx_i2s_trigger, + .hw_params = pxa2xx_i2s_hw_params, + .set_fmt = pxa2xx_i2s_set_dai_fmt, + .set_sysclk = pxa2xx_i2s_set_dai_sysclk, +}; + struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, @@ -350,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = { .channels_max = 2, .rates = PXA2XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .startup = pxa2xx_i2s_startup, - .shutdown = pxa2xx_i2s_shutdown, - .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params, - .set_fmt = pxa2xx_i2s_set_dai_fmt, - .set_sysclk = pxa2xx_i2s_set_dai_sysclk, - }, + .ops = &pxa_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(pxa_i2s_dai); @@ -398,11 +369,6 @@ static struct platform_driver pxa2xx_i2s_driver = { static int __init pxa2xx_i2s_init(void) { - if (cpu_is_pxa27x()) - gpio_bus[1].sys = GPIO113_I2S_SYSCLK_MD; - else - gpio_bus[1].sys = GPIO32_SYSCLK_I2S_MD; - clk_i2s = ERR_PTR(-ENOENT); return platform_driver_register(&pxa2xx_i2s_driver); } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index a3b9e6bdf979..6ca9f53080c6 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -109,7 +109,7 @@ static void spitz_ext_control(struct snd_soc_codec *codec) static int spitz_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ spitz_ext_control(codec); @@ -278,7 +278,7 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = { */ static int spitz_wm8750_init(struct snd_soc_codec *codec) { - int i, err; + int err; /* NC codec pins */ snd_soc_dapm_nc_pin(codec, "RINPUT1"); @@ -290,12 +290,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) snd_soc_dapm_nc_pin(codec, "MONO1"); /* Add spitz specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8750_spitz_controls, + ARRAY_SIZE(wm8750_spitz_controls)); + if (err < 0) + return err; /* Add spitz specific widgets */ snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index c77194f74c9b..fc781374b1bf 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -82,7 +82,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; + struct snd_soc_codec *codec = rtd->socdev->card->codec; /* check the jack status at stream startup */ tosa_ext_control(codec); @@ -188,18 +188,16 @@ static const struct snd_kcontrol_new tosa_controls[] = { static int tosa_ac97_init(struct snd_soc_codec *codec) { - int i, err; + int err; snd_soc_dapm_nc_pin(codec, "OUT3"); snd_soc_dapm_nc_pin(codec, "MONOOUT"); /* add tosa specific controls */ - for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&tosa_controls[i],codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, tosa_controls, + ARRAY_SIZE(tosa_controls)); + if (err < 0) + return err; /* add tosa specific widgets */ snd_soc_dapm_new_controls(codec, tosa_dapm_widgets, diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index f8e9ecd589d3..9a386b4c4ed1 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/device.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <sound/core.h> #include <sound/pcm.h> @@ -26,6 +27,17 @@ #include "pxa2xx-ac97.h" #include "pxa-ssp.h" +/* + * There is a physical switch SW15 on the board which changes the MCLK + * for the WM9713 between the standard AC97 master clock and the + * output of the CLK_POUT signal from the PXA. + */ +static int clk_pout; +module_param(clk_pout, int, 0); +MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board)."); + +static struct clk *pout; + static struct snd_soc_card zylonite; static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = { @@ -61,10 +73,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_codec *codec) { - /* Currently we only support use of the AC97 clock here. If - * CLK_POUT is selected by SW15 then the clock API will need - * to be used to request and enable it here. - */ + if (clk_pout) + snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0); snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); @@ -86,40 +96,35 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, 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; - unsigned int acds = 0; unsigned int wm9713_div = 0; int ret = 0; + int rate = params_rate(params); + int width = snd_pcm_format_physical_width(params_format(params)); - switch (params_rate(params)) { + /* Only support ratios that we can generate neatly from the AC97 + * based master clock - in particular, this excludes 44.1kHz. + * In most applications the voice DAC will be used for telephony + * data so multiples of 8kHz will be the common case. + */ + switch (rate) { case 8000: wm9713_div = 12; - pll_out = 2048000; break; case 16000: wm9713_div = 6; - pll_out = 4096000; break; case 48000: - default: wm9713_div = 2; - pll_out = 12288000; - acds = 1; break; + default: + /* Don't support OSS emulation */ + return -EINVAL; } - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - 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; + /* Add 1 to the width for the leading clock cycle */ + pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, - params_channels(params), - params_channels(params)); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; @@ -127,19 +132,22 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds); + if (clk_pout) + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, + WM9713_PCMDIV(wm9713_div)); + else + ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, + WM9713_PCMDIV(wm9713_div)); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; - /* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs - * to be set instead. - */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV, - WM9713_PCMDIV(wm9713_div)); + 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; @@ -173,8 +181,72 @@ static struct snd_soc_dai_link zylonite_dai[] = { }, }; +static int zylonite_probe(struct platform_device *pdev) +{ + int ret; + + if (clk_pout) { + pout = clk_get(NULL, "CLK_POUT"); + if (IS_ERR(pout)) { + dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n", + PTR_ERR(pout)); + return PTR_ERR(pout); + } + + ret = clk_enable(pout); + if (ret != 0) { + dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + ret); + clk_put(pout); + return ret; + } + + dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n", + clk_get_rate(pout)); + } + + return 0; +} + +static int zylonite_remove(struct platform_device *pdev) +{ + if (clk_pout) { + clk_disable(pout); + clk_put(pout); + } + + return 0; +} + +static int zylonite_suspend_post(struct platform_device *pdev, + pm_message_t state) +{ + if (clk_pout) + clk_disable(pout); + + return 0; +} + +static int zylonite_resume_pre(struct platform_device *pdev) +{ + int ret = 0; + + if (clk_pout) { + ret = clk_enable(pout); + if (ret != 0) + dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n", + ret); + } + + return ret; +} + static struct snd_soc_card zylonite = { .name = "Zylonite", + .probe = &zylonite_probe, + .remove = &zylonite_remove, + .suspend_post = &zylonite_suspend_post, + .resume_pre = &zylonite_resume_pre, .platform = &pxa2xx_soc_platform, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index fcd03acf10f6..2f3a21eee051 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,19 +1,31 @@ config SND_S3C24XX_SOC - tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 + tristate "SoC Audio for the Samsung S3CXXXX chips" + depends on ARCH_S3C2410 || ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to - the S3C24XX AC97, I2S or SSP interface. You will also need - to select the audio interfaces to support below. + the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will + also need to select the audio interfaces to support below. config SND_S3C24XX_SOC_I2S tristate + select S3C2410_DMA + +config SND_S3C_I2SV2_SOC + tristate config SND_S3C2412_SOC_I2S tristate + select SND_S3C_I2SV2_SOC + select S3C2410_DMA + +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA config SND_S3C2443_SOC_AC97 tristate + select S3C2410_DMA select AC97_BUS select SND_SOC_AC97_BUS @@ -26,6 +38,14 @@ 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_JIVE_WM8750 + tristate "SoC I2S Audio support for Jive" + depends on SND_S3C24XX_SOC && MACH_JIVE + select SND_SOC_WM8750 + select SND_S3C2412_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the Jive. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 @@ -48,4 +68,5 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" depends on SND_S3C24XX_SOC select SND_S3C24XX_SOC_I2S + select SND_SOC_L3 select SND_SOC_UDA134X diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 96b3f3f617d4..07a93a2ebe5f 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -2,19 +2,25 @@ snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o +snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(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-smdk2443-wm9710-objs := smdk2443_wm9710.o snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.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-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c new file mode 100644 index 000000000000..32063790d95b --- /dev/null +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -0,0 +1,201 @@ +/* sound/soc/s3c24xx/jive_wm8750.c + * + * Copyright 2007,2008 Simtec Electronics + * + * Based on sound/soc/pxa/spitz.c + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * 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/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "s3c24xx-pcm.h" +#include "s3c2412-i2s.h" + +#include "../codecs/wm8750.h" + +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LOUT1" }, + { "Headphone Jack", NULL, "ROUT1" }, + { "Internal Speaker", NULL, "LOUT2" }, + { "Internal Speaker", NULL, "ROUT2" }, + { "LINPUT1", NULL, "Line Input" }, + { "RINPUT1", NULL, "Line Input" }, +}; + +static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Internal Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static int jive_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; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + s3c_i2sv2_calc_rate(&div, NULL, params_rate(params), + s3c2412_get_iisclk()); + + /* 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); + 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; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops jive_ops = { + .hw_params = jive_hw_params, +}; + +static int jive_wm8750_init(struct snd_soc_codec *codec) +{ + int err; + + /* These endpoints are not being used. */ + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO"); + + /* Add jive specific widgets */ + err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); + if (err) { + printk(KERN_ERR "%s: failed to add widgets (%d)\n", + __func__, err); + return err; + } + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link jive_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .cpu_dai = &s3c2412_i2s_dai, + .codec_dai = &wm8750_dai, + .init = jive_wm8750_init, + .ops = &jive_ops, +}; + +/* jive audio machine driver */ +static struct snd_soc_machine snd_soc_machine_jive = { + .name = "Jive", + .dai_link = &jive_dai, + .num_links = 1, +}; + +/* jive audio private data */ +static struct wm8750_setup_data jive_wm8750_setup = { +}; + +/* jive audio subsystem */ +static struct snd_soc_device jive_snd_devdata = { + .machine = &snd_soc_machine_jive, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_wm8750_spi, + .codec_data = &jive_wm8750_setup, +}; + +static struct platform_device *jive_snd_device; + +static int __init jive_init(void) +{ + int ret; + + if (!machine_is_jive()) + return 0; + + printk("JIVE WM8750 Audio support\n"); + + jive_snd_device = platform_device_alloc("soc-audio", -1); + if (!jive_snd_device) + return -ENOMEM; + + platform_set_drvdata(jive_snd_device, &jive_snd_devdata); + jive_snd_devdata.dev = &jive_snd_device->dev; + ret = platform_device_add(jive_snd_device); + + if (ret) + platform_device_put(jive_snd_device); + + return ret; +} + +static void __exit jive_exit(void) +{ + platform_device_unregister(jive_snd_device); +} + +module_init(jive_init); +module_exit(jive_exit); + +MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 45bb12e8ea44..289fadf60b10 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -29,25 +29,17 @@ #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h> -#include <mach/audio.h> +#include <plat/audio.h> #include <linux/io.h> #include <mach/spi-gpio.h> -#include <asm/plat-s3c24xx/regs-iis.h> +#include <plat/regs-iis.h> #include "../codecs/wm8753.h" #include "lm4857.h" #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -/* Debugging stuff */ -#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0 -#if S3C24XX_SOC_NEO1973_WM8753_DEBUG -#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x) -#else -#define DBG(x...) -#endif - /* define the scenarios */ #define NEO_AUDIO_OFF 0 #define NEO_GSM_CALL_AUDIO_HANDSET 1 @@ -72,7 +64,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iis_clkrate = s3c24xx_i2s_get_clockrate(); @@ -158,7 +150,7 @@ static int neo1973_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; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); @@ -181,7 +173,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iis_clkrate = s3c24xx_i2s_get_clockrate(); @@ -224,7 +216,7 @@ static int neo1973_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; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* disable the PLL */ return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); @@ -246,7 +238,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (neo1973_scenario) { case NEO_AUDIO_OFF: @@ -330,7 +322,7 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (neo1973_scenario == ucontrol->value.integer.value[0]) return 0; @@ -344,7 +336,7 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (i2c_master_send(i2c, lm4857_regs, 4) != 4) printk(KERN_ERR "lm4857: i2c write failed\n"); @@ -357,7 +349,7 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; return 0; @@ -385,7 +377,7 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol, { u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (value) value -= 5; @@ -399,7 +391,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (value) value += 5; @@ -506,9 +498,9 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { */ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { - int i, err; + int err; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* set up NC codec pins */ snd_soc_dapm_nc_pin(codec, "LOUT2"); @@ -526,13 +518,10 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) set_scenario_endpoints(codec, NEO_AUDIO_OFF); /* add neo1973 specific controls */ - for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { - err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8753_neo1973_controls[i], - codec, NULL)); - if (err < 0) - return err; - } + err = snd_soc_add_controls(codec, wm8753_neo1973_controls, + ARRAY_SIZE(8753_neo1973_controls)); + if (err < 0) + return err; /* set up neo1973 specific audio routes */ err = snd_soc_dapm_add_routes(codec, dapm_routes, @@ -585,21 +574,15 @@ static struct snd_soc_card neo1973 = { .num_links = ARRAY_SIZE(neo1973_dai), }; -static struct wm8753_setup_data neo1973_wm8753_setup = { - .i2c_bus = 0, - .i2c_address = 0x1a, -}; - static struct snd_soc_device neo1973_snd_devdata = { .card = &neo1973, .codec_dev = &soc_codec_dev_wm8753, - .codec_data = &neo1973_wm8753_setup, }; static int lm4857_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); i2c = client; @@ -609,7 +592,7 @@ static int lm4857_i2c_probe(struct i2c_client *client, static int lm4857_i2c_remove(struct i2c_client *client) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); i2c = NULL; @@ -620,7 +603,7 @@ static u8 lm4857_state; static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); dev_dbg(&dev->dev, "lm4857_suspend\n"); lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; @@ -633,7 +616,7 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) static int lm4857_resume(struct i2c_client *dev) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (lm4857_state) { lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); @@ -644,7 +627,7 @@ static int lm4857_resume(struct i2c_client *dev) static void lm4857_shutdown(struct i2c_client *dev) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); dev_dbg(&dev->dev, "lm4857_shutdown\n"); lm4857_regs[LM4857_CTRL] &= 0xf0; @@ -675,7 +658,7 @@ static int __init neo1973_init(void) { int ret; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (!machine_is_neo1973_gta01()) { printk(KERN_INFO @@ -706,7 +689,7 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c new file mode 100644 index 000000000000..295a4c910262 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -0,0 +1,638 @@ +/* sound/soc/s3c24xx/s3c-i2c-v2.c + * + * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory@wolfsonmicro.com + * linux@wolfsonmicro.com + * + * Copyright (c) 2008, 2007, 2004-2005 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 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/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/regs-s3c2412-iis.h> + +#include <plat/audio.h> +#include <mach/dma.h> + +#include "s3c-i2s-v2.h" + +#define S3C2412_I2S_DEBUG_CON 0 + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +#define bit_set(v, b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + + +/* Turn on or off the transmission path. */ +void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} +EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl); + +void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} +EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl); + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) +{ + u32 iiscon; + unsigned long timeout = jiffies + msecs_to_jiffies(5); + + pr_debug("Entered %s\n", __func__); + + while (1) { + iiscon = readl(i2s->regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + if (timeout < jiffies) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + } + + return 0; +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK +#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE +#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL +#endif + +#if defined(CONFIG_PLAT_S3C64XX) +/* From Rev1.1 datasheet, we have two master and two slave modes: + * IMS[11:10]: + * 00 = master mode, fed from PCLK + * 01 = master mode, fed from CLKAUDIO + * 10 = slave mode, using PCLK + * 11 = slave mode, using I2SCLK + */ +#define IISMOD_MASTER_MASK (1 << 11) +#define IISMOD_SLAVE (1 << 11) +#define IISMOD_MASTER (0x0) +#endif + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 0; + iismod &= ~IISMOD_MASTER_MASK; + iismod |= IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 1; + iismod &= ~IISMOD_MASTER_MASK; + iismod |= IISMOD_MASTER; + break; + default: + pr_debug("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + pr_debug("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->cpu_dai->dma_data = i2s->dma_playback; + else + dai->cpu_dai->dma_data = i2s->dma_capture; + + /* Working copies of register */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, i2s->regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!i2s->master) { + ret = s3c2412_snd_lrsync(i2s); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 1); + else + s3c2412_snd_txctrl(i2s, 1); + + local_irq_restore(irqs); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 0); + else + s3c2412_snd_txctrl(i2s, 0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 reg; + + pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C_I2SV2_DIV_BCLK: + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_RCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +/* default table of all avaialable root fs divisors */ +static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; + +int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + if (fstab == NULL) + fstab = iis_fs_tab; + + for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { + fsdiv = iis_fs_tab[fs]; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = abs(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} +EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); + +int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base) +{ + struct device *dev = &pdev->dev; + + i2s->dev = dev; + + /* record our i2s structure for later use in the callbacks */ + dai->private_data = i2s; + + i2s->regs = ioremap(base, 0x100); + if (i2s->regs == NULL) { + dev_err(dev, "cannot ioremap registers\n"); + return -ENXIO; + } + + i2s->iis_pclk = clk_get(dev, "iis"); + if (i2s->iis_pclk == NULL) { + dev_err(dev, "failed to get iis_clock\n"); + iounmap(i2s->regs); + return -ENOENT; + } + + clk_enable(i2s->iis_pclk); + + s3c2412_snd_txctrl(i2s, 0); + s3c2412_snd_rxctrl(i2s, 0); + + return 0; +} + +EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); + +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warning("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warning("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warning("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + +int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) +{ + dai->ops.trigger = s3c2412_i2s_trigger; + dai->ops.hw_params = s3c2412_i2s_hw_params; + dai->ops.set_fmt = s3c2412_i2s_set_fmt; + dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv; + + dai->suspend = s3c2412_i2s_suspend; + dai->resume = s3c2412_i2s_resume; + + return snd_soc_register_dai(dai); +} + +EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h new file mode 100644 index 000000000000..f66854a77fb2 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -0,0 +1,90 @@ +/* sound/soc/s3c24xx/s3c-i2s-v2.h + * + * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver + * + * Copyright (c) 2007 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +/* This code is the core support for the I2S block found in a number of + * Samsung SoC devices which is unofficially named I2S-V2. Currently the + * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S + * channels via configurable GPIO. + */ + +#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H +#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ + +#define S3C_I2SV2_DIV_BCLK (1) +#define S3C_I2SV2_DIV_RCLK (2) +#define S3C_I2SV2_DIV_PRESCALER (3) + +/** + * struct s3c_i2sv2_info - S3C I2S-V2 information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device registe block. + * @master: True if the I2S core is the I2S bit clock master. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + * @suspend_iismod: PM save for the IISMOD register. + * @suspend_iiscon: PM save for the IISCON register. + * @suspend_iispsr: PM save for the IISPSR register. + * + * This is the private codec state for the hardware associated with an + * I2S channel such as the register mappings and clock sources. + */ +struct s3c_i2sv2_info { + struct device *dev; + void __iomem *regs; + + struct clk *iis_pclk; + struct clk *iis_cclk; + struct clk *iis_clk; + + unsigned char master; + + struct s3c24xx_pcm_dma_params *dma_playback; + struct s3c24xx_pcm_dma_params *dma_capture; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; +}; + +struct s3c_i2sv2_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +/** + * s3c_i2sv2_probe - probe for i2s device helper + * @pdev: The platform device supplied to the original probe. + * @dai: The ASoC DAI structure supplied to the original probe. + * @i2s: Our local i2s structure to fill in. + * @base: The base address for the registers. + */ +extern int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base); + +/** + * s3c_i2sv2_register_dai - register dai with soc core + * @dai: The snd_soc_dai structure to register + * + * Fill in any missing fields and then register the given dai with the + * soc core. + */ +extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); + +#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index f3fc0aba0aaf..1ca3cdaa8213 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -22,6 +22,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/kernel.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -30,26 +31,16 @@ #include <sound/soc.h> #include <mach/hardware.h> -#include <linux/io.h> -#include <asm/dma.h> - -#include <asm/plat-s3c24xx/regs-s3c2412-iis.h> +#include <plat/regs-s3c2412-iis.h> -#include <mach/regs-gpio.h> -#include <mach/audio.h> +#include <plat/regs-gpio.h> +#include <plat/audio.h> #include <mach/dma.h> #include "s3c24xx-pcm.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 -#define S3C2412_I2S_DEBUG_CON 0 - -#if S3C2412_I2S_DEBUG -#define DBG(x...) printk(KERN_INFO x) -#else -#define DBG(x...) do { } while (0) -#endif static struct s3c2410_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" @@ -73,431 +64,7 @@ static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { .dma_size = 4, }; -struct s3c2412_i2s_info { - struct device *dev; - void __iomem *regs; - struct clk *iis_clk; - struct clk *iis_pclk; - struct clk *iis_cclk; - - u32 suspend_iismod; - u32 suspend_iiscon; - u32 suspend_iispsr; -}; - -static struct s3c2412_i2s_info s3c2412_i2s; - -#define bit_set(v, b) (((v) & (b)) ? 1 : 0) - -#if S3C2412_I2S_DEBUG_CON -static void dbg_showcon(const char *fn, u32 con) -{ - printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, - bit_set(con, S3C2412_IISCON_LRINDEX), - bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_TXFIFO_FULL), - bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); - - printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", - fn, - bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_TXCH_PAUSE), - bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); - printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, - bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); -} -#else -static inline void dbg_showcon(const char *fn, u32 con) -{ -} -#endif - -/* Turn on or off the transmission path. */ -static void s3c2412_snd_txctrl(int on) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - DBG("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_TXDMA_PAUSE; - con &= ~S3C2412_IISCON_TXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXONLY: - case S3C2412_IISMOD_MODE_TXRX: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_RXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } else { - /* Note, we do not have any indication that the FIFO problems - * tha the S3C2410/2440 had apply here, so we should be able - * to disable the DMA and TX without resetting the FIFOS. - */ - - con |= S3C2412_IISCON_TXDMA_PAUSE; - con |= S3C2412_IISCON_TXCH_PAUSE; - con &= ~S3C2412_IISCON_TXDMA_ACTIVE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_RXONLY; - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - con &= ~S3C2412_IISCON_IIS_ACTIVE; - break; - - default: - dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } - - fic = readl(regs + S3C2412_IISFIC); - dbg_showcon(__func__, con); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -static void s3c2412_snd_rxctrl(int on) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - DBG("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_RXDMA_PAUSE; - con &= ~S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - case S3C2412_IISMOD_MODE_RXONLY: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } else { - /* See txctrl notes on FIFOs. */ - - con &= ~S3C2412_IISCON_RXDMA_ACTIVE; - con |= S3C2412_IISCON_RXDMA_PAUSE; - con |= S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_RXONLY: - con &= ~S3C2412_IISCON_IIS_ACTIVE; - mod &= ~S3C2412_IISMOD_MODE_MASK; - break; - - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXONLY; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } - - fic = readl(regs + S3C2412_IISFIC); - DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c2412_snd_lrsync(void) -{ - u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); - - DBG("Entered %s\n", __func__); - - while (1) { - iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON); - if (iiscon & S3C2412_IISCON_LRINDEX) - break; - - if (timeout < jiffies) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } - } - - return 0; -} - -/* - * Check whether CPU is the master or slave - */ -static inline int s3c2412_snd_is_clkmaster(void) -{ - u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - - DBG("Entered %s\n", __func__); - - iismod &= S3C2412_IISMOD_MASTER_MASK; - return !(iismod == S3C2412_IISMOD_SLAVE); -} - -/* - * Set S3C2412 I2S DAI format - */ -static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 iismod; - - - DBG("Entered %s\n", __func__); - - iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - iismod &= ~S3C2412_IISMOD_MASTER_MASK; - iismod |= S3C2412_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - iismod &= ~S3C2412_IISMOD_MASTER_MASK; - iismod |= S3C2412_IISMOD_MASTER_INTERNAL; - break; - default: - DBG("unknwon master/slave format\n"); - return -EINVAL; - } - - iismod &= ~S3C2412_IISMOD_SDF_MASK; - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_RIGHT_J: - iismod |= S3C2412_IISMOD_SDF_MSB; - break; - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2412_IISMOD_SDF_LSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod |= S3C2412_IISMOD_SDF_IIS; - break; - default: - DBG("Unknown data format\n"); - return -EINVAL; - } - - writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c2412_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; - u32 iismod; - - DBG("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out; - else - rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in; - - /* Working copies of register */ - iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("%s: r: IISMOD: %x\n", __func__, iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } - - writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("%s: w: IISMOD: %x\n", __func__, iismod); - return 0; -} - -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - unsigned long irqs; - int ret = 0; - - DBG("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* On start, ensure that the FIFOs are cleared and reset. */ - - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, - s3c2412_i2s.regs + S3C2412_IISFIC); - - /* clear again, just in case */ - writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC); - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!s3c2412_snd_is_clkmaster()) { - ret = s3c2412_snd_lrsync(); - if (ret) - goto exit_err; - } - - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(1); - else - s3c2412_snd_txctrl(1); - - local_irq_restore(irqs); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(0); - else - s3c2412_snd_txctrl(0); - - local_irq_restore(irqs); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* default table of all avaialable root fs divisors */ -static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 }; - -int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) -{ - unsigned long clkrate = clk_get_rate(clk); - unsigned int div; - unsigned int fsclk; - unsigned int actual; - unsigned int fs; - unsigned int fsdiv; - signed int deviation = 0; - unsigned int best_fs = 0; - unsigned int best_div = 0; - unsigned int best_rate = 0; - unsigned int best_deviation = INT_MAX; - - - if (fstab == NULL) - fstab = s3c2412_iis_fs; - - for (fs = 0;; fs++) { - fsdiv = s3c2412_iis_fs[fs]; - - if (fsdiv == 0) - break; - - fsclk = clkrate / fsdiv; - div = fsclk / rate; - - if ((fsclk % rate) > (rate / 2)) - div++; - - if (div <= 1) - continue; - - actual = clkrate / (fsdiv * div); - deviation = actual - rate; - - printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", - fsdiv, div, actual, deviation); - - deviation = abs(deviation); - - if (deviation < best_deviation) { - best_fs = fsdiv; - best_div = div; - best_rate = actual; - best_deviation = deviation; - } - - if (deviation == 0) - break; - } - - printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", - best_fs, best_div, best_rate); - - info->fs_div = best_fs; - info->clk_div = best_div; - - return 0; -} -EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); +static struct s3c_i2sv2_info s3c2412_i2s; /* * Set S3C2412 Clock source @@ -507,15 +74,17 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, + pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, freq, dir); switch (clk_id) { case S3C2412_CLKSRC_PCLK: + s3c2412_i2s.master = 1; iismod &= ~S3C2412_IISMOD_MASTER_MASK; iismod |= S3C2412_IISMOD_MASTER_INTERNAL; break; case S3C2412_CLKSRC_I2SCLK: + s3c2412_i2s.master = 0; iismod &= ~S3C2412_IISMOD_MASTER_MASK; iismod |= S3C2412_IISMOD_MASTER_EXTERNAL; break; @@ -527,74 +96,6 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -/* - * Set S3C2412 Clock dividers - */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - u32 reg; - - DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); - - switch (div_id) { - case S3C2412_DIV_BCLK: - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_BCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - - DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C2412_DIV_RCLK: - if (div > 3) { - /* convert value to bit field */ - - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; - - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; - - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; - - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; - - default: - return -EINVAL; - } - } - - reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_RCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C2412_DIV_PRESCALER: - if (div >= 0) { - writel((div << 8) | S3C2412_IISPSR_PSREN, - i2s->regs + S3C2412_IISPSR); - } else { - writel(0x0, i2s->regs + S3C2412_IISPSR); - } - DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); - break; - - default: - return -EINVAL; - } - - return 0; -} struct clk *s3c2412_get_iisclk(void) { @@ -606,34 +107,30 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); static int s3c2412_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - DBG("Entered %s\n", __func__); + int ret; - s3c2412_i2s.dev = &pdev->dev; + pr_debug("Entered %s\n", __func__); - s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c2412_i2s.regs == NULL) - return -ENXIO; + ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); + if (ret) + return ret; - s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis"); - if (s3c2412_i2s.iis_pclk == NULL) { - DBG("failed to get iis_clock\n"); - iounmap(s3c2412_i2s.regs); - return -ENODEV; - } + s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; + s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); if (s3c2412_i2s.iis_cclk == NULL) { - DBG("failed to get i2sclk clock\n"); + pr_debug("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); return -ENODEV; } - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + /* Set MPLL as the source for IIS CLK */ - clk_enable(s3c2412_i2s.iis_pclk); + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); clk_enable(s3c2412_i2s.iis_cclk); - s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk; + s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; /* Configure the I2S pins in correct mode */ s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); @@ -642,78 +139,22 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - s3c2412_snd_txctrl(0); - s3c2412_snd_rxctrl(0); - return 0; } -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warning("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warning("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warning("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c2412_i2s_info *i2s = &s3c2412_i2s; - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif /* CONFIG_PM */ - #define S3C2412_I2S_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) +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .set_sysclk = s3c2412_i2s_set_sysclk, +}; + struct snd_soc_dai s3c2412_i2s_dai = { - .name = "s3c2412-i2s", - .id = 0, - .probe = s3c2412_i2s_probe, - .suspend = s3c2412_i2s_suspend, - .resume = s3c2412_i2s_resume, + .name = "s3c2412-i2s", + .id = 0, + .probe = s3c2412_i2s_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -726,19 +167,13 @@ struct snd_soc_dai s3c2412_i2s_dai = { .rates = S3C2412_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = { - .trigger = s3c2412_i2s_trigger, - .hw_params = s3c2412_i2s_hw_params, - .set_fmt = s3c2412_i2s_set_fmt, - .set_clkdiv = s3c2412_i2s_set_clkdiv, - .set_sysclk = s3c2412_i2s_set_sysclk, - }, + .ops = &s3c2412_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); static int __init s3c2412_i2s_init(void) { - return snd_soc_register_dai(&s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); } module_init(s3c2412_i2s_init); @@ -748,7 +183,6 @@ static void __exit s3c2412_i2s_exit(void) } module_exit(s3c2412_i2s_exit); - /* Module information */ MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index aac08a25e541..92848e54be16 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -15,9 +15,11 @@ #ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H #define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ -#define S3C2412_DIV_BCLK (1) -#define S3C2412_DIV_RCLK (2) -#define S3C2412_DIV_PRESCALER (3) +#include "s3c-i2s-v2.h" + +#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER #define S3C2412_CLKSRC_PCLK (0) #define S3C2412_CLKSRC_I2SCLK (1) @@ -26,13 +28,4 @@ extern struct clk *s3c2412_get_iisclk(void); extern struct snd_soc_dai s3c2412_i2s_dai; -struct s3c2412_rate_calc { - unsigned int clk_div; /* for prescaler */ - unsigned int fs_div; /* for root frame clock */ -}; - -extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk); - #endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 5822d2dd49ba..3698f707c44d 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -31,7 +31,7 @@ #include <plat/regs-ac97.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <mach/audio.h> +#include <plat/audio.h> #include <asm/dma.h> #include <mach/dma.h> @@ -355,6 +355,16 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = { + .hw_params = s3c2443_ac97_hw_params, + .trigger = s3c2443_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = { + .hw_params = s3c2443_ac97_hw_mic_params, + .trigger = s3c2443_ac97_mic_trigger, +}; + struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", @@ -374,9 +384,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 2, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_params, - .trigger = s3c2443_ac97_trigger}, + .ops = &s3c2443_ac97_dai_ops, }, { .name = "pxa2xx-ac97-mic", @@ -388,9 +396,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { .channels_max = 1, .rates = s3c2443_AC97_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .hw_params = s3c2443_ac97_hw_mic_params, - .trigger = s3c2443_ac97_mic_trigger,}, + .ops = &s3c2443_ac97_mic_dai_ops, }, }; EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 6f4d439b57aa..cc066964dad6 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -4,7 +4,7 @@ * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com * - * (c) 2004-2005 Simtec Electronics + * Copyright 2004-2005 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks <ben@simtec.co.uk> * @@ -30,22 +30,15 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <mach/audio.h> +#include <plat/audio.h> #include <asm/dma.h> #include <mach/dma.h> -#include <asm/plat-s3c24xx/regs-iis.h> +#include <plat/regs-iis.h> #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -#define S3C24XX_I2S_DEBUG 0 -#if S3C24XX_I2S_DEBUG -#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) -#else -#define DBG(x...) -#endif - static struct s3c2410_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; @@ -84,13 +77,13 @@ static void s3c24xx_snd_txctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); if (on) { iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; @@ -120,7 +113,7 @@ static void s3c24xx_snd_txctrl(int on) writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); } - DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); } static void s3c24xx_snd_rxctrl(int on) @@ -129,13 +122,13 @@ static void s3c24xx_snd_rxctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); if (on) { iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; @@ -165,7 +158,7 @@ static void s3c24xx_snd_rxctrl(int on) writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); } - DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon); + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); } /* @@ -177,7 +170,7 @@ static int s3c24xx_snd_lrsync(void) u32 iiscon; int timeout = 50; /* 5ms */ - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); while (1) { iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -197,7 +190,7 @@ static int s3c24xx_snd_lrsync(void) */ static inline int s3c24xx_snd_is_clkmaster(void) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; } @@ -210,10 +203,10 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, { u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params r: IISMOD: %lx \n", iismod); + pr_debug("hw_params r: IISMOD: %x \n", iismod); switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -238,7 +231,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params w: IISMOD: %lx \n", iismod); + pr_debug("hw_params w: IISMOD: %x \n", iismod); return 0; } @@ -249,7 +242,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; @@ -258,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params r: IISMOD: %lx\n", iismod); + pr_debug("hw_params r: IISMOD: %x\n", iismod); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -276,7 +269,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, } writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("hw_params w: IISMOD: %lx\n", iismod); + pr_debug("hw_params w: IISMOD: %x\n", iismod); return 0; } @@ -285,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -327,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); iismod &= ~S3C2440_IISMOD_MPLL; @@ -353,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, { u32 reg; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); switch (div_id) { case S3C24XX_DIV_BCLK: @@ -389,7 +382,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); if (s3c24xx_i2s.regs == NULL) @@ -397,7 +390,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); if (s3c24xx_i2s.iis_clk == NULL) { - DBG("failed to get iis_clock\n"); + pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); return -ENODEV; } @@ -421,7 +414,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -435,7 +428,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) { - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); @@ -456,6 +449,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, @@ -472,13 +473,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .channels_max = 2, .rates = S3C24XX_I2S_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, - }, + .ops = &s3c24xx_i2s_dai_ops, }; EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7c64d31d067e..a9d68fa2b34a 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -4,7 +4,7 @@ * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com * - * (c) 2004-2005 Simtec Electronics + * Copyright 2004-2005 Simtec Electronics * http://armlinux.simtec.co.uk/ * Ben Dooks <ben@simtec.co.uk> * @@ -29,17 +29,10 @@ #include <asm/dma.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> +#include <plat/audio.h> #include "s3c24xx-pcm.h" -#define S3C24XX_PCM_DEBUG 0 -#if S3C24XX_PCM_DEBUG -#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x) -#else -#define DBG(x...) -#endif - static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -84,16 +77,16 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) dma_addr_t pos = prtd->dma_pos; int ret; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); while (prtd->dma_loaded < prtd->dma_limit) { unsigned long len = prtd->dma_period; - DBG("dma_loaded: %d\n", prtd->dma_loaded); + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; - DBG(KERN_DEBUG "%s: corrected dma len %ld\n", + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", __func__, len); } @@ -119,7 +112,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, struct snd_pcm_substream *substream = dev_id; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) return; @@ -148,7 +141,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, unsigned long totbytes = params_buffer_bytes(params); int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -161,14 +154,14 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, /* prepare DMA */ prtd->params = dma; - DBG("params %p, client %p, channel %d\n", prtd->params, + pr_debug("params %p, client %p, channel %d\n", prtd->params, prtd->params->client, prtd->params->channel); ret = s3c2410_dma_request(prtd->params->channel, prtd->params->client, NULL); if (ret < 0) { - DBG(KERN_ERR "failed to get dma channel\n"); + printk(KERN_ERR "failed to get dma channel\n"); return ret; } } @@ -196,7 +189,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* TODO - do we need to ensure DMA flushed */ snd_pcm_set_runtime_buffer(substream, NULL); @@ -214,7 +207,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -259,7 +252,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); spin_lock(&prtd->lock); @@ -297,7 +290,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) unsigned long res; dma_addr_t src, dst; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); spin_lock(&prtd->lock); s3c2410_dma_getposition(prtd->params->channel, &src, &dst); @@ -309,7 +302,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) spin_unlock(&prtd->lock); - DBG("Pointer %x %x\n", src, dst); + pr_debug("Pointer %x %x\n", src, dst); /* we seem to be getting the odd error from the pcm library due * to out-of-bounds pointers. this is maybe due to the dma engine @@ -330,7 +323,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); @@ -349,10 +342,10 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (!prtd) - DBG("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); kfree(prtd); @@ -364,7 +357,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); return dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, @@ -390,7 +383,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -409,7 +402,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_dma_buffer *buf; int stream; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -433,7 +426,7 @@ static int s3c24xx_pcm_new(struct snd_card *card, { int ret = 0; - DBG("Entered %s\n", __func__); + pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) card->dev->dma_mask = &s3c24xx_pcm_dmamask; diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index a0a4d1832a14..8e79a416db57 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -22,7 +22,7 @@ #include <sound/s3c24xx_uda134x.h> #include <sound/uda134x.h> -#include <asm/plat-s3c24xx/regs-iis.h> +#include <plat/regs-iis.h> #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c new file mode 100644 index 000000000000..33c5de7e255f --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -0,0 +1,222 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben@simtec.co.uk> + * http://armlinux.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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/regs-s3c2412-iis.h> +#include <plat/gpio-bank-d.h> +#include <plat/gpio-bank-e.h> +#include <plat/gpio-cfg.h> +#include <plat/audio.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c24xx-pcm.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { + [0] = { + .channel = DMACH_I2S0_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_OUT, + .client = &s3c64xx_dma_client_out, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD, + .dma_size = 4, + }, +}; + +static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { + [0] = { + .channel = DMACH_I2S0_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD, + .dma_size = 4, + }, + [1] = { + .channel = DMACH_I2S1_IN, + .client = &s3c64xx_dma_client_in, + .dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD, + .dma_size = 4, + }, +}; + +static struct s3c_i2sv2_info s3c64xx_i2s[2]; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + switch (clk_id) { + case S3C64XX_CLKSRC_PCLK: + iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + break; + + case S3C64XX_CLKSRC_MUX: + iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + + return 0; +} + + +unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + return clk_get_rate(i2s->iis_cclk); +} +EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate); + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct device *dev = &pdev->dev; + struct s3c_i2sv2_info *i2s; + int ret; + + dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id); + + if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) { + dev_err(dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + + ret = s3c_i2sv2_probe(pdev, dai, i2s, + pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0); + if (ret) + return ret; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + i2s->iis_cclk = clk_get(dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(dev, "failed to get audio-bus"); + iounmap(i2s->regs); + return -ENODEV; + } + + /* configure GPIO for i2s port */ + switch (pdev->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +#define S3C64XX_I2S_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 S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { + .set_sysclk = s3c64xx_i2s_set_sysclk, +}; + +struct snd_soc_dai s3c64xx_i2s_dai = { + .name = "s3c64xx-i2s", + .id = 0, + .probe = s3c64xx_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C64XX_I2S_RATES, + .formats = S3C64XX_I2S_FMTS, + }, + .ops = &s3c64xx_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static int __init s3c64xx_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c64xx_i2s_dai); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c64xx_i2s_dai); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); + + + diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h new file mode 100644 index 000000000000..b7ffe3c38b66 --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -0,0 +1,31 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben@simtec.co.uk> + * http://armlinux.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. + */ + +#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK (0) +#define S3C64XX_CLKSRC_MUX (1) + +extern struct snd_soc_dai s3c64xx_i2s_dai; + +extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai); + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index eab31838badf..41db75af3c69 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -267,6 +267,10 @@ static int hac_hw_params(struct snd_pcm_substream *substream, #define AC97_FMTS \ SNDRV_PCM_FMTBIT_S16_LE +static struct snd_soc_dai_ops hac_dai_ops = { + .hw_params = hac_hw_params, +}; + struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", @@ -284,9 +288,7 @@ struct snd_soc_dai sh4_hac_dai[] = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .hw_params = hac_hw_params, - }, + .ops = &hac_dai_ops, }, #ifdef CONFIG_CPU_SUBTYPE_SH7760 { @@ -305,9 +307,7 @@ struct snd_soc_dai sh4_hac_dai[] = { .channels_min = 2, .channels_max = 2, }, - .ops = { - .hw_params = hac_hw_params, - }, + .ops = &hac_dai_ops, }, #endif diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index d1e5390fddeb..56fa0872abbb 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) +static struct snd_soc_dai_ops ssi_dai_ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, +}; + struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", @@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #ifdef CONFIG_CPU_SUBTYPE_SH7760 { @@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { .channels_min = 2, .channels_max = 8, }, - .ops = { - .startup = ssi_startup, - .shutdown = ssi_shutdown, - .trigger = ssi_trigger, - .hw_params = ssi_hw_params, - .set_sysclk = ssi_set_sysclk, - .set_clkdiv = ssi_set_clkdiv, - .set_fmt = ssi_set_fmt, - }, + .ops = &ssi_dai_ops, }, #endif }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 318dfdd54d7f..6e710f705a74 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream, cpu_dai); + if (cpu_dai->ops->startup) { + ret = cpu_dai->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream, codec_dai); + if (codec_dai->ops->startup) { + ret = codec_dai->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -234,7 +234,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->capture.active = codec_dai->capture.active = 1; cpu_dai->active = codec_dai->active = 1; cpu_dai->runtime = runtime; - socdev->codec->active++; + card->codec->active++; mutex_unlock(&pcm_mutex); return 0; @@ -247,8 +247,8 @@ codec_dai_err: platform->pcm_ops->close(substream); platform_err: - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -264,7 +264,7 @@ static void close_delayed_work(struct work_struct *work) struct snd_soc_card *card = container_of(work, struct snd_soc_card, delayed_work.work); struct snd_soc_device *socdev = card->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; struct snd_soc_dai *codec_dai; int i; @@ -319,7 +319,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; mutex_lock(&pcm_mutex); @@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream, cpu_dai); + if (cpu_dai->ops->shutdown) + cpu_dai->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream, codec_dai); + if (codec_dai->ops->shutdown) + codec_dai->ops->shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -387,7 +387,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; int ret = 0; mutex_lock(&pcm_mutex); @@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream, codec_dai); + if (codec_dai->ops->prepare) { + ret = codec_dai->ops->prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; } } - if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream, cpu_dai); + if (cpu_dai->ops->prepare) { + ret = cpu_dai->ops->prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params, codec_dai); + if (codec_dai->ops->hw_params) { + ret = codec_dai->ops->hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); + if (cpu_dai->ops->hw_params) { + ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -526,12 +526,12 @@ out: return ret; platform_err: - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -553,7 +553,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *cpu_dai = machine->cpu_dai; struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; mutex_lock(&pcm_mutex); @@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->pcm_ops->hw_free(substream); /* now free hw params for the DAI's */ - if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream, codec_dai); + if (codec_dai->ops->hw_free) + codec_dai->ops->hw_free(substream, codec_dai); - if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream, cpu_dai); + if (cpu_dai->ops->hw_free) + cpu_dai->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai = machine->codec_dai; int ret; - if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd, codec_dai); + if (codec_dai->ops->trigger) { + ret = codec_dai->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); + if (cpu_dai->ops->trigger) { + ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -629,7 +629,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) struct snd_soc_card *card = socdev->card; struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; int i; /* Due to the resume being scheduled into a workqueue we could @@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) /* mute any active DAC's */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 1); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 1); } /* suspend all pcms */ @@ -705,7 +705,7 @@ static void soc_resume_deferred(struct work_struct *work) struct snd_soc_device *socdev = card->socdev; struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = card->codec; struct platform_device *pdev = to_platform_device(socdev->dev); int i; @@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *dai = card->dai_link[i].codec_dai; - if (dai->ops.digital_mute && dai->playback.active) - dai->ops.digital_mute(dai, 0); + if (dai->ops->digital_mute && dai->playback.active) + dai->ops->digital_mute(dai, 0); } for (i = 0; i < card->num_links; i++) { @@ -982,8 +982,8 @@ static struct platform_driver soc_driver = { static int soc_new_pcm(struct snd_soc_device *socdev, struct snd_soc_dai_link *dai_link, int num) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = card->codec; struct snd_soc_platform *platform = card->platform; struct snd_soc_dai *codec_dai = dai_link->codec_dai; struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; @@ -998,7 +998,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev, rtd->dai = dai_link; rtd->socdev = socdev; - codec_dai->codec = socdev->codec; + codec_dai->codec = card->codec; /* check client and interface hw capabilities */ sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, @@ -1048,9 +1048,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, } /* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf) +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { - struct snd_soc_codec *codec = devdata->codec; int i, step = 1, count = 0; if (!codec->reg_cache_size) @@ -1090,7 +1089,7 @@ static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata, buf); + return soc_codec_reg_show(devdata->card->codec, buf); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -1107,12 +1106,10 @@ static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, { ssize_t ret; struct snd_soc_codec *codec = file->private_data; - struct device *card_dev = codec->card->dev; - struct snd_soc_device *devdata = card_dev->driver_data; char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); if (!buf) return -ENOMEM; - ret = soc_codec_reg_show(devdata, buf); + ret = soc_codec_reg_show(codec, buf); if (ret >= 0) ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); kfree(buf); @@ -1309,8 +1306,8 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits); */ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = card->codec; int ret, i; mutex_lock(&codec->mutex); @@ -1355,8 +1352,8 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms); */ int snd_soc_init_card(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = card->codec; int ret = 0, i, ac97 = 0, err = 0; for (i = 0; i < card->num_links; i++) { @@ -1385,7 +1382,10 @@ int snd_soc_init_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS - if (ac97) { + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { ret = soc_ac97_dev_register(codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); @@ -1404,7 +1404,7 @@ int snd_soc_init_card(struct snd_soc_device *socdev) if (err < 0) printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - soc_init_codec_debugfs(socdev->codec); + soc_init_codec_debugfs(codec); mutex_unlock(&codec->mutex); out: @@ -1421,18 +1421,19 @@ EXPORT_SYMBOL_GPL(snd_soc_init_card); */ void snd_soc_free_pcms(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; #ifdef CONFIG_SND_SOC_AC97_BUS struct snd_soc_dai *codec_dai; int i; #endif mutex_lock(&codec->mutex); - soc_cleanup_codec_debugfs(socdev->codec); + soc_cleanup_codec_debugfs(codec); #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - if (codec_dai->ac97_control && codec->ac97) { + if (codec_dai->ac97_control && codec->ac97 && + strcmp(codec->name, "AC97") != 0) { soc_ac97_dev_unregister(codec); goto free_card; } @@ -1495,6 +1496,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, EXPORT_SYMBOL_GPL(snd_soc_cnew); /** + * snd_soc_add_controls - add an array of controls to a codec. + * Convienience function to add a list of controls. Many codecs were + * duplicating this code. + * + * @codec: codec to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_controls(struct snd_soc_codec *codec, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = codec->card; + int err, i; + + for (i = 0; i < num_controls; i++) { + const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL)); + if (err < 0) { + dev_err(codec->dev, "%s: Failed to add %s\n", + codec->name, control->name); + return err; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_add_controls); + +/** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control * @uinfo: control element information @@ -1585,37 +1617,6 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** - * snd_soc_info_value_enum_double - semi enumerated double mixer info callback - * @kcontrol: mixer control - * @uinfo: control element information - * - * Callback to provide information about a double semi enumerated - * mixer control. - * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. - * - * Returns 0 for success. - */ -int snd_soc_info_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->max; - - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_info_value_enum_double); - -/** * snd_soc_get_value_enum_double - semi enumerated double mixer get callback * @kcontrol: mixer control * @ucontrol: control element information @@ -1631,8 +1632,7 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short reg_val, val, mux; reg_val = snd_soc_read(codec, e->reg); @@ -1671,8 +1671,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val; unsigned short mask; @@ -2053,8 +2052,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops.set_sysclk) - return dai->ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops->set_sysclk) + return dai->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -2073,8 +2072,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops.set_clkdiv) - return dai->ops.set_clkdiv(dai, div_id, div); + if (dai->ops->set_clkdiv) + return dai->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -2092,8 +2091,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops.set_pll) - return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops->set_pll) + return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -2108,8 +2107,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops.set_fmt) - return dai->ops.set_fmt(dai, fmt); + if (dai->ops->set_fmt) + return dai->ops->set_fmt(dai, fmt); else return -EINVAL; } @@ -2127,8 +2126,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->ops.set_sysclk) - return dai->ops.set_tdm_slot(dai, mask, slots); + if (dai->ops->set_sysclk) + return dai->ops->set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -2143,8 +2142,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops.set_sysclk) - return dai->ops.set_tristate(dai, tristate); + if (dai->ops->set_sysclk) + return dai->ops->set_tristate(dai, tristate); else return -EINVAL; } @@ -2159,8 +2158,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops.digital_mute) - return dai->ops.digital_mute(dai, mute); + if (dai->ops->digital_mute) + return dai->ops->digital_mute(dai, mute); else return -EINVAL; } @@ -2213,6 +2212,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) return 0; } +static struct snd_soc_dai_ops null_dai_ops = { +}; + /** * snd_soc_register_dai - Register a DAI with the ASoC core * @@ -2227,6 +2229,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai) if (!dai->dev) printk(KERN_WARNING "No device for DAI %s\n", dai->name); + if (!dai->ops) + dai->ops = &null_dai_ops; + INIT_LIST_HEAD(&dai->list); mutex_lock(&client_mutex); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ad0d801677c1..735903a74675 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -54,14 +54,15 @@ static int dapm_up_seq[] = { snd_soc_dapm_pre, 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_pga, snd_soc_dapm_adc, snd_soc_dapm_hp, - snd_soc_dapm_spk, snd_soc_dapm_post + 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 }; + 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, snd_soc_dapm_dac, snd_soc_dapm_mic, - snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux, - snd_soc_dapm_post + 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_post }; static int dapm_status = 1; @@ -101,7 +102,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, { switch (w->id) { case snd_soc_dapm_switch: - case snd_soc_dapm_mixer: { + case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: { int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrols[i].private_value; @@ -137,7 +139,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, } break; case snd_soc_dapm_value_mux: { - struct soc_value_enum *e = (struct soc_value_enum *) + struct soc_enum *e = (struct soc_enum *) w->kcontrols[i].private_value; int val, item; @@ -200,30 +202,6 @@ static int dapm_connect_mux(struct snd_soc_codec *codec, return -ENODEV; } -/* connect value_mux widget to it's interconnecting audio paths */ -static int dapm_connect_value_mux(struct snd_soc_codec *codec, - struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, - struct snd_soc_dapm_path *path, const char *control_name, - const struct snd_kcontrol_new *kcontrol) -{ - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; - int i; - - for (i = 0; i < e->max; i++) { - if (!(strcmp(control_name, e->texts[i]))) { - list_add(&path->list, &codec->dapm_paths); - list_add(&path->list_sink, &dest->sources); - list_add(&path->list_source, &src->sinks); - path->name = (char *)e->texts[i]; - dapm_set_path_status(dest, path, 0); - return 0; - } - } - - return -ENODEV; -} - /* connect mixer widget to it's interconnecting audio paths */ static int dapm_connect_mixer(struct snd_soc_codec *codec, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, @@ -347,15 +325,32 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, if (path->name != (char*)w->kcontrols[i].name) continue; - /* add dapm control with long name */ - name_len = 2 + strlen(w->name) - + strlen(w->kcontrols[i].name); + /* add dapm control with long name. + * for dapm_mixer this is the concatenation of the + * mixer and kcontrol name. + * for dapm_mixer_named_ctl this is simply the + * kcontrol name. + */ + name_len = strlen(w->kcontrols[i].name) + 1; + if (w->id != snd_soc_dapm_mixer_named_ctl) + name_len += 1 + strlen(w->name); + path->long_name = kmalloc(name_len, GFP_KERNEL); + if (path->long_name == NULL) return -ENOMEM; - snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + switch (w->id) { + default: + snprintf(path->long_name, name_len, "%s %s", + w->name, w->kcontrols[i].name); + break; + case snd_soc_dapm_mixer_named_ctl: + snprintf(path->long_name, name_len, "%s", + w->kcontrols[i].name); + break; + } + path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, @@ -527,6 +522,137 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, EXPORT_SYMBOL_GPL(dapm_reg_event); /* + * 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) +{ + int in, out, power_change, power, ret; + + /* vmid - no action */ + if (w->id == snd_soc_dapm_vmid) + return 0; + + /* active ADC */ + if (w->id == snd_soc_dapm_adc && w->active) { + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + w->power = (in != 0) ? 1 : 0; + dapm_update_bits(w); + return 0; + } + + /* active DAC */ + if (w->id == snd_soc_dapm_dac && w->active) { + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + w->power = (out != 0) ? 1 : 0; + dapm_update_bits(w); + return 0; + } + + /* pre and post event widgets */ + if (w->id == snd_soc_dapm_pre) { + if (!w->event) + return 0; + + if (event == SND_SOC_DAPM_STREAM_START) { + 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); + if (ret < 0) + return ret; + } + return 0; + } + if (w->id == snd_soc_dapm_post) { + if (!w->event) + return 0; + + if (event == SND_SOC_DAPM_STREAM_START) { + 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); + if (ret < 0) + return ret; + } + return 0; + } + + /* all other widgets */ + in = is_connected_input_ep(w); + dapm_clear_walk(w->codec); + out = is_connected_output_ep(w); + dapm_clear_walk(w->codec); + power = (out != 0 && in != 0) ? 1 : 0; + power_change = (w->power == power) ? 0 : 1; + w->power = power; + + if (!power_change) + return 0; + + /* call any power change event handlers */ + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !power) + dapm_set_pga(w, power); + + dapm_update_bits(w); + + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && power) + dapm_set_pga(w, power); + + /* power up post event */ + if (power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; + } + + return 0; +} + +/* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- * @@ -538,7 +664,7 @@ EXPORT_SYMBOL_GPL(dapm_reg_event); static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { struct snd_soc_dapm_widget *w; - int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power; + int i, c = 1, *seq = NULL, ret = 0; /* do we have a sequenced stream event */ if (event == SND_SOC_DAPM_STREAM_START) { @@ -549,135 +675,20 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) seq = dapm_down_seq; } - for(i = 0; i < c; i++) { + for (i = 0; i < c; i++) { list_for_each_entry(w, &codec->dapm_widgets, list) { /* is widget in stream order */ if (seq && seq[i] && w->id != seq[i]) continue; - /* vmid - no action */ - if (w->id == snd_soc_dapm_vmid) - continue; - - /* active ADC */ - if (w->id == snd_soc_dapm_adc && w->active) { - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - w->power = (in != 0) ? 1 : 0; - dapm_update_bits(w); - continue; - } - - /* active DAC */ - if (w->id == snd_soc_dapm_dac && w->active) { - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - w->power = (out != 0) ? 1 : 0; - dapm_update_bits(w); - continue; - } - - /* pre and post event widgets */ - if (w->id == snd_soc_dapm_pre) { - if (!w->event) - continue; - - if (event == SND_SOC_DAPM_STREAM_START) { - 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); - if (ret < 0) - return ret; - } - continue; - } - if (w->id == snd_soc_dapm_post) { - if (!w->event) - continue; - - if (event == SND_SOC_DAPM_STREAM_START) { - 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); - if (ret < 0) - return ret; - } - continue; - } - - /* all other widgets */ - in = is_connected_input_ep(w); - dapm_clear_walk(w->codec); - out = is_connected_output_ep(w); - dapm_clear_walk(w->codec); - power = (out != 0 && in != 0) ? 1 : 0; - power_change = (w->power == power) ? 0: 1; - w->power = power; - - if (!power_change) - continue; - - /* call any power change event handlers */ - if (w->event) - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", - w->name, w->event_flags); - - /* power up pre event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - - /* power down pre event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - - /* Lower PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && !power) - dapm_set_pga(w, power); - - dapm_update_bits(w); - - /* Raise PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && power) - dapm_set_pga(w, power); - - /* power up post event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - - /* power down post event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } + ret = dapm_power_widget(codec, event, w); + if (ret != 0) + return ret; } } - return ret; + return 0; } #ifdef DEBUG @@ -711,6 +722,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) case snd_soc_dapm_adc: case snd_soc_dapm_pga: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: if (w->name) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); @@ -744,7 +756,8 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_soc_dapm_path *path; int found = 0; - if (widget->id != snd_soc_dapm_mux) + if (widget->id != snd_soc_dapm_mux && + widget->id != snd_soc_dapm_value_mux) return -ENODEV; if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) @@ -774,45 +787,6 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } -/* test and update the power status of a value_mux widget */ -static int dapm_value_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int mux, int val, struct soc_value_enum *e) -{ - struct snd_soc_dapm_path *path; - int found = 0; - - if (widget->id != snd_soc_dapm_value_mux) - return -ENODEV; - - if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) - return 0; - - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->codec->dapm_paths, list) { - if (path->kcontrol != kcontrol) - continue; - - if (!path->name || !e->texts[mux]) - continue; - - found = 1; - /* we now need to match the string in the enum to the path */ - if (!(strcmp(path->name, e->texts[mux]))) - path->connect = 1; /* new connection */ - else - path->connect = 0; /* old connection must be - powered down */ - } - - if (found) { - dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); - dump_dapm(widget->codec, "mux power update"); - } - - return 0; -} - /* test and update the power status of a mixer or switch widget */ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int reg, @@ -822,6 +796,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, int found = 0; if (widget->id != snd_soc_dapm_mixer && + widget->id != snd_soc_dapm_mixer_named_ctl && widget->id != snd_soc_dapm_switch) return -ENODEV; @@ -857,7 +832,7 @@ static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_soc_device *devdata = dev_get_drvdata(dev); - struct snd_soc_codec *codec = devdata->codec; + struct snd_soc_codec *codec = devdata->card->codec; struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -875,6 +850,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_adc: case snd_soc_dapm_pga: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -938,7 +914,7 @@ static void dapm_free_widgets(struct snd_soc_codec *codec) } static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec, - char *pin, int status) + const char *pin, int status) { struct snd_soc_dapm_widget *w; @@ -1045,19 +1021,15 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->connect = 1; return 0; case snd_soc_dapm_mux: + case snd_soc_dapm_value_mux: ret = dapm_connect_mux(codec, wsource, wsink, path, control, &wsink->kcontrols[0]); if (ret != 0) goto err; break; - case snd_soc_dapm_value_mux: - ret = dapm_connect_value_mux(codec, wsource, wsink, path, - control, &wsink->kcontrols[0]); - if (ret != 0) - goto err; - break; case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: ret = dapm_connect_mixer(codec, wsource, wsink, path, control); if (ret != 0) goto err; @@ -1135,6 +1107,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) switch(w->id) { case snd_soc_dapm_switch: case snd_soc_dapm_mixer: + case snd_soc_dapm_mixer_named_ctl: dapm_new_mixer(codec, w); break; case snd_soc_dapm_mux: @@ -1382,8 +1355,7 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short reg_val, val, mux; reg_val = snd_soc_read(widget->codec, e->reg); @@ -1423,8 +1395,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); - struct soc_value_enum *e = (struct soc_value_enum *) - kcontrol->private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned short val, mux; unsigned short mask; int ret = 0; @@ -1443,7 +1414,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_value_mux_update_power(widget, kcontrol, mask, mux, val, e); + dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); if (widget->event) { if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ret = widget->event(widget, @@ -1465,6 +1436,76 @@ out: EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); /** + * snd_soc_dapm_info_pin_switch - Info for a pin switch + * + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information about a pin switch control. + */ +int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_info_pin_switch); + +/** + * snd_soc_dapm_get_pin_switch - Get information for a pin switch + * + * @kcontrol: mixer control + * @ucontrol: Value + */ +int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + const char *pin = (const char *)kcontrol->private_value; + + mutex_lock(&codec->mutex); + + ucontrol->value.integer.value[0] = + snd_soc_dapm_get_pin_status(codec, pin); + + mutex_unlock(&codec->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_switch); + +/** + * snd_soc_dapm_put_pin_switch - Set information for a pin switch + * + * @kcontrol: mixer control + * @ucontrol: Value + */ +int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + const char *pin = (const char *)kcontrol->private_value; + + mutex_lock(&codec->mutex); + + if (ucontrol->value.integer.value[0]) + snd_soc_dapm_enable_pin(codec, pin); + else + snd_soc_dapm_disable_pin(codec, pin); + + snd_soc_dapm_sync(codec); + + mutex_unlock(&codec->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); + +/** * snd_soc_dapm_new_control - create new dapm control * @codec: audio codec * @widget: widget template @@ -1596,8 +1637,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, enum snd_soc_bias_level level) { - struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; + struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; if (card->set_bias_level) @@ -1618,7 +1659,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 1); } @@ -1633,7 +1674,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 0); } @@ -1653,7 +1694,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin) { return snd_soc_dapm_set_pin(codec, pin, 0); } @@ -1668,7 +1709,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); * * Returns 1 for connected otherwise 0. */ -int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin) +int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin) { struct snd_soc_dapm_widget *w; @@ -1689,7 +1730,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status); */ void snd_soc_dapm_free(struct snd_soc_device *socdev) { - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = socdev->card->codec; snd_soc_dapm_sys_remove(socdev->dev); dapm_free_widgets(codec); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c new file mode 100644 index 000000000000..28346fb2e70c --- /dev/null +++ b/sound/soc/soc-jack.c @@ -0,0 +1,267 @@ +/* + * soc-jack.c -- ALSA SoC jack handling + * + * Copyright 2008 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 <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/workqueue.h> +#include <linux/delay.h> + +/** + * snd_soc_jack_new - Create a new jack + * @card: ASoC card + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack) +{ + jack->card = card; + INIT_LIST_HEAD(&jack->pins); + + return snd_jack_new(card->codec->card, id, type, &jack->jack); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_new); + +/** + * snd_soc_jack_report - Report the current status for a jack + * + * @jack: the jack + * @status: a bitmask of enum snd_jack_type values that are currently detected. + * @mask: a bitmask of enum snd_jack_type values that being reported. + * + * If configured using snd_soc_jack_add_pins() then the associated + * DAPM pins will be enabled or disabled as appropriate and DAPM + * synchronised. + * + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). + */ +void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) +{ + struct snd_soc_codec *codec = jack->card->codec; + struct snd_soc_jack_pin *pin; + int enable; + int oldstatus; + + if (!jack) { + WARN_ON_ONCE(!jack); + return; + } + + mutex_lock(&codec->mutex); + + oldstatus = jack->status; + + jack->status &= ~mask; + jack->status |= status; + + /* The DAPM sync is expensive enough to be worth skipping */ + if (jack->status == oldstatus) + goto out; + + list_for_each_entry(pin, &jack->pins, list) { + enable = pin->mask & status; + + if (pin->invert) + enable = !enable; + + if (enable) + snd_soc_dapm_enable_pin(codec, pin->pin); + else + snd_soc_dapm_disable_pin(codec, pin->pin); + } + + snd_soc_dapm_sync(codec); + + snd_jack_report(jack->jack, status); + +out: + mutex_unlock(&codec->mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_jack_report); + +/** + * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack + * + * @jack: ASoC jack + * @count: Number of pins + * @pins: Array of pins + * + * After this function has been called the DAPM pins specified in the + * pins array will have their status updated to reflect the current + * state of the jack whenever the jack status is updated. + */ +int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_pin *pins) +{ + int i; + + for (i = 0; i < count; i++) { + if (!pins[i].pin) { + printk(KERN_ERR "No name for pin %d\n", i); + return -EINVAL; + } + if (!pins[i].mask) { + printk(KERN_ERR "No mask for pin %d (%s)\n", i, + pins[i].pin); + return -EINVAL; + } + + INIT_LIST_HEAD(&pins[i].list); + list_add(&(pins[i].list), &jack->pins); + } + + /* Update to reflect the last reported status; canned jack + * implementations are likely to set their state before the + * card has an opportunity to associate pins. + */ + snd_soc_jack_report(jack, 0, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); + +#ifdef CONFIG_GPIOLIB +/* gpio detect */ +static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +{ + struct snd_soc_jack *jack = gpio->jack; + int enable; + int report; + + if (gpio->debounce_time > 0) + mdelay(gpio->debounce_time); + + enable = gpio_get_value(gpio->gpio); + if (gpio->invert) + enable = !enable; + + if (enable) + report = gpio->report; + else + report = 0; + + snd_soc_jack_report(jack, report, gpio->report); +} + +/* irq handler for gpio pin */ +static irqreturn_t gpio_handler(int irq, void *data) +{ + struct snd_soc_jack_gpio *gpio = data; + + schedule_work(&gpio->work); + + return IRQ_HANDLED; +} + +/* gpio work */ +static void gpio_work(struct work_struct *work) +{ + struct snd_soc_jack_gpio *gpio; + + gpio = container_of(work, struct snd_soc_jack_gpio, work); + snd_soc_jack_gpio_detect(gpio); +} + +/** + * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * This function will request gpio, set data direction and request irq + * for each gpio in the array. + */ +int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i, ret; + + for (i = 0; i < count; i++) { + if (!gpio_is_valid(gpios[i].gpio)) { + printk(KERN_ERR "Invalid gpio %d\n", + gpios[i].gpio); + ret = -EINVAL; + goto undo; + } + if (!gpios[i].name) { + printk(KERN_ERR "No name for gpio %d\n", + gpios[i].gpio); + ret = -EINVAL; + goto undo; + } + + ret = gpio_request(gpios[i].gpio, gpios[i].name); + if (ret) + goto undo; + + ret = gpio_direction_input(gpios[i].gpio); + if (ret) + goto err; + + ret = request_irq(gpio_to_irq(gpios[i].gpio), + gpio_handler, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, + jack->card->dev->driver->name, + &gpios[i]); + if (ret) + goto err; + + INIT_WORK(&gpios[i].work, gpio_work); + gpios[i].jack = jack; + } + + return 0; + +err: + gpio_free(gpios[i].gpio); +undo: + snd_soc_jack_free_gpios(jack, i, gpios); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios); + +/** + * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack + * + * @jack: ASoC jack + * @count: number of pins + * @gpios: array of gpio pins + * + * Release gpio and irq resources for gpio pins associated with an ASoC jack. + */ +void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_gpio *gpios) +{ + int i; + + for (i = 0; i < count; i++) { + free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]); + gpio_free(gpios[i].gpio); + gpios[i].jack = NULL; + } +} +EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios); +#endif /* CONFIG_GPIOLIB */ diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index ba38912614b4..574af56ba8a6 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -954,7 +954,8 @@ static int __devinit snd_amd7930_create(struct snd_card *card, amd->regs = of_ioremap(&op->resource[0], 0, resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { - snd_printk("amd7930-%d: Unable to map chip registers.\n", dev); + snd_printk(KERN_ERR + "amd7930-%d: Unable to map chip registers.\n", dev); return -EIO; } @@ -962,7 +963,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, if (request_irq(irq, snd_amd7930_interrupt, IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) { - snd_printk("amd7930-%d: Unable to grab IRQ %d\n", + snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", dev, irq); snd_amd7930_free(amd); return -EBUSY; diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index 0a5391436add..ff0b2a8fd25b 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -24,25 +24,6 @@ #include <asm/uaccess.h> #include "emux_voice.h" -/* - * open the hwdep device - */ -static int -snd_emux_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - - -/* - * close the device - */ -static int -snd_emux_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - #define TMP_CLIENT_ID 0x1001 @@ -146,8 +127,6 @@ snd_emux_init_hwdep(struct snd_emux *emu) emu->hwdep = hw; strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME); hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE; - hw->ops.open = snd_emux_hwdep_open; - hw->ops.release = snd_emux_hwdep_release; hw->ops.ioctl = snd_emux_hwdep_ioctl; hw->exclusive = 1; hw->private_data = emu; diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index 5c47b6c09264..87e42206c4ef 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -132,7 +132,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) p = snd_emux_create_port(emu, tmpname, 32, 1, &callback); if (p == NULL) { - snd_printk("can't create port\n"); + snd_printk(KERN_ERR "can't create port\n"); snd_emux_dec_count(emu); mutex_unlock(&emu->register_mutex); return -ENOMEM; diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 335aa2ce2574..ca5f7effb4df 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -74,15 +74,15 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) emu->client = snd_seq_create_kernel_client(card, index, "%s WaveTable", emu->name); if (emu->client < 0) { - snd_printk("can't create client\n"); + snd_printk(KERN_ERR "can't create client\n"); return -ENODEV; } if (emu->num_ports < 0) { - snd_printk("seqports must be greater than zero\n"); + snd_printk(KERN_WARNING "seqports must be greater than zero\n"); emu->num_ports = 1; } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) { - snd_printk("too many ports." + snd_printk(KERN_WARNING "too many ports." "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS); emu->num_ports = SNDRV_EMUX_MAX_PORTS; } @@ -100,7 +100,7 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) p = snd_emux_create_port(emu, tmpname, MIDI_CHANNELS, 0, &pinfo); if (p == NULL) { - snd_printk("can't create port\n"); + snd_printk(KERN_ERR "can't create port\n"); return -ENOMEM; } @@ -147,12 +147,12 @@ snd_emux_create_port(struct snd_emux *emu, char *name, /* Allocate structures for this channel */ if ((p = kzalloc(sizeof(*p), GFP_KERNEL)) == NULL) { - snd_printk("no memory\n"); + snd_printk(KERN_ERR "no memory\n"); return NULL; } p->chset.channels = kcalloc(max_channels, sizeof(struct snd_midi_channel), GFP_KERNEL); if (p->chset.channels == NULL) { - snd_printk("no memory\n"); + snd_printk(KERN_ERR "no memory\n"); kfree(p); return NULL; } @@ -376,12 +376,12 @@ int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card) goto __error; } emu->vmidi[i] = rmidi; - //snd_printk("virmidi %d ok\n", i); + /* snd_printk(KERN_DEBUG "virmidi %d ok\n", i); */ } return 0; __error: - //snd_printk("error init..\n"); + /* snd_printk(KERN_DEBUG "error init..\n"); */ snd_emux_delete_virmidi(emu); return -ENOMEM; } diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 2cc6f6f79065..3e921b386fd5 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -956,7 +956,8 @@ void snd_emux_lock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_OFF) emu->voices[voice].state = SNDRV_EMUX_ST_LOCKED; else - snd_printk("invalid voice for lock %d (state = %x)\n", + snd_printk(KERN_WARNING + "invalid voice for lock %d (state = %x)\n", voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } @@ -973,7 +974,8 @@ void snd_emux_unlock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_LOCKED) emu->voices[voice].state = SNDRV_EMUX_ST_OFF; else - snd_printk("invalid voice for unlock %d (state = %x)\n", + snd_printk(KERN_WARNING + "invalid voice for unlock %d (state = %x)\n", voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 36d53bd317ed..63c8f45c0c22 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -133,7 +133,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk("patch record too small %ld\n", count); + snd_printk(KERN_ERR "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -143,15 +143,16 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, data += sizeof(patch); if (patch.key != SNDRV_OSS_SOUNDFONT_PATCH) { - snd_printk("'The wrong kind of patch' %x\n", patch.key); + snd_printk(KERN_ERR "The wrong kind of patch %x\n", patch.key); return -EINVAL; } if (count < patch.len) { - snd_printk("Patch too short %ld, need %d\n", count, patch.len); + snd_printk(KERN_ERR "Patch too short %ld, need %d\n", + count, patch.len); return -EINVAL; } if (patch.len < 0) { - snd_printk("poor length %d\n", patch.len); + snd_printk(KERN_ERR "poor length %d\n", patch.len); return -EINVAL; } @@ -195,7 +196,8 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, case SNDRV_SFNT_REMOVE_INFO: /* patch must be opened */ if (!sflist->currsf) { - snd_printk("soundfont: remove_info: patch not opened\n"); + snd_printk(KERN_ERR "soundfont: remove_info: " + "patch not opened\n"); rc = -EINVAL; } else { int bank, instr; @@ -531,7 +533,7 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) return -EINVAL; if (count < (long)sizeof(hdr)) { - printk("Soundfont error: invalid patch zone length\n"); + printk(KERN_ERR "Soundfont error: invalid patch zone length\n"); return -EINVAL; } if (copy_from_user((char*)&hdr, data, sizeof(hdr))) @@ -541,12 +543,14 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) count -= sizeof(hdr); if (hdr.nvoices <= 0 || hdr.nvoices >= 100) { - printk("Soundfont error: Illegal voice number %d\n", hdr.nvoices); + printk(KERN_ERR "Soundfont error: Illegal voice number %d\n", + hdr.nvoices); return -EINVAL; } if (count < (long)sizeof(struct soundfont_voice_info) * hdr.nvoices) { - printk("Soundfont Error: patch length(%ld) is smaller than nvoices(%d)\n", + printk(KERN_ERR "Soundfont Error: " + "patch length(%ld) is smaller than nvoices(%d)\n", count, hdr.nvoices); return -EINVAL; } @@ -952,7 +956,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk("patch record too small %ld\n", count); + snd_printk(KERN_ERR "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -1034,7 +1038,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, /* panning position; -128 - 127 => 0-127 */ zone->v.pan = (patch.panning + 128) / 2; #if 0 - snd_printk("gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", + snd_printk(KERN_DEBUG + "gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", (int)patch.base_freq, zone->v.rate_offset, zone->v.root, zone->v.tune, zone->v.low, zone->v.high); #endif @@ -1068,7 +1073,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, zone->v.parm.volrelease = 0x8000 | snd_sf_calc_parm_decay(release); zone->v.attenuation = calc_gus_attenuation(patch.env_offset[0]); #if 0 - snd_printk("gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", + snd_printk(KERN_DEBUG + "gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", zone->v.parm.volatkhld, zone->v.parm.voldcysus, zone->v.parm.volrelease, diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 55a9075cb097..09aed2363cc9 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,7 +42,7 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.9"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index f9fbdbae269d..ab56e738c5fc 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -75,6 +75,7 @@ struct snd_usb_caiaqdev { wait_queue_head_t ep1_wait_queue; wait_queue_head_t prepare_wait_queue; int spec_received, audio_parm_answer; + int midi_out_active; char vendor_name[CAIAQ_USB_STR_LEN]; char product_name[CAIAQ_USB_STR_LEN]; diff --git a/sound/usb/caiaq/caiaq-midi.c b/sound/usb/caiaq/caiaq-midi.c index 30b57f97c6e4..f19fd360c936 100644 --- a/sound/usb/caiaq/caiaq-midi.c +++ b/sound/usb/caiaq/caiaq-midi.c @@ -59,6 +59,11 @@ static int snd_usb_caiaq_midi_output_open(struct snd_rawmidi_substream *substrea static int snd_usb_caiaq_midi_output_close(struct snd_rawmidi_substream *substream) { + struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; + if (dev->midi_out_active) { + usb_kill_urb(&dev->midi_out_urb); + dev->midi_out_active = 0; + } return 0; } @@ -69,7 +74,8 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, dev->midi_out_buf[0] = EP1_CMD_MIDI_WRITE; dev->midi_out_buf[1] = 0; /* port */ - len = snd_rawmidi_transmit_peek(substream, dev->midi_out_buf+3, EP1_BUFSIZE-3); + len = snd_rawmidi_transmit(substream, dev->midi_out_buf + 3, + EP1_BUFSIZE - 3); if (len <= 0) return; @@ -79,24 +85,24 @@ static void snd_usb_caiaq_midi_send(struct snd_usb_caiaqdev *dev, ret = usb_submit_urb(&dev->midi_out_urb, GFP_ATOMIC); if (ret < 0) - log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed, %d\n", - substream, ret); + log("snd_usb_caiaq_midi_send(%p): usb_submit_urb() failed," + "ret=%d, len=%d\n", + substream, ret, len); + else + dev->midi_out_active = 1; } static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) { struct snd_usb_caiaqdev *dev = substream->rmidi->private_data; - if (dev->midi_out_substream != NULL) - return; - - if (!up) { + if (up) { + dev->midi_out_substream = substream; + if (!dev->midi_out_active) + snd_usb_caiaq_midi_send(dev, substream); + } else { dev->midi_out_substream = NULL; - return; } - - dev->midi_out_substream = substream; - snd_usb_caiaq_midi_send(dev, substream); } @@ -161,16 +167,14 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device) void snd_usb_caiaq_midi_output_done(struct urb* urb) { struct snd_usb_caiaqdev *dev = urb->context; - char *buf = urb->transfer_buffer; + dev->midi_out_active = 0; if (urb->status != 0) return; if (!dev->midi_out_substream) return; - snd_rawmidi_transmit_ack(dev->midi_out_substream, buf[2]); - dev->midi_out_substream = NULL; snd_usb_caiaq_midi_send(dev, dev->midi_out_substream); } diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index eec32e1a3020..8f3cdb37a0ec 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2524,7 +2524,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * build the rate table and bitmap flags */ int r, idx; - unsigned int nonzero_rates = 0; fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); if (fp->rate_table == NULL) { @@ -2532,24 +2531,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform return -1; } - fp->nr_rates = nr_rates; - fp->rate_min = fp->rate_max = combine_triple(&fmt[8]); + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { unsigned int rate = combine_triple(&fmt[idx]); + if (!rate) + continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && - chip->usb_id == USB_ID(0x0d8c, 0x0201) && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; - fp->rate_table[r] = rate; - nonzero_rates |= rate; - if (rate < fp->rate_min) + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) fp->rate_min = rate; - else if (rate > fp->rate_max) + if (!fp->rate_max || rate > fp->rate_max) fp->rate_max = rate; fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; } - if (!nonzero_rates) { + if (!fp->nr_rates) { hwc_debug("All rates were zero. Skipping format!\n"); return -1; } @@ -2966,6 +2968,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 320641ab5be7..26bad373fe65 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, } ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.out_interval = 0; ep_info.out_cables = endpoint->out_cables & 0x5555; err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]); if (err < 0) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 00397c8a765b..2bde79216fa5 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -78,7 +78,6 @@ struct usb_mixer_interface { /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; - unsigned long rc_hwdep_open; u32 rc_code; wait_queue_head_t rc_waitq; struct urb *rc_urb; @@ -1797,24 +1796,6 @@ static void snd_usb_soundblaster_remote_complete(struct urb *urb) wake_up(&mixer->rc_waitq); } -static int snd_usb_sbrc_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - if (test_and_set_bit(0, &mixer->rc_hwdep_open)) - return -EBUSY; - return 0; -} - -static int snd_usb_sbrc_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - clear_bit(0, &mixer->rc_hwdep_open); - smp_mb__after_clear_bit(); - return 0; -} - static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf, long count, loff_t *offset) { @@ -1867,9 +1848,8 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC; hwdep->private_data = mixer; hwdep->ops.read = snd_usb_sbrc_hwdep_read; - hwdep->ops.open = snd_usb_sbrc_hwdep_open; - hwdep->ops.release = snd_usb_sbrc_hwdep_release; hwdep->ops.poll = snd_usb_sbrc_hwdep_poll; + hwdep->exclusive = 1; mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL); if (!mixer->rc_urb) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 92115755d98e..5d8ef09b9dcc 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -128,6 +128,14 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + USB_DEVICE(0x046d, 0x0990), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Logitech, Inc.", + .product_name = "QuickCam Pro 9000", + .ifnum = QUIRK_NO_INTERFACE + } +}, /* * Yamaha devices diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 1558a5c4094f..4af8740db717 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -30,9 +30,6 @@ #include "usbusx2y.h" #include "usX2Yhwdep.h" -int usX2Y_hwdep_pcm_new(struct snd_card *card); - - static int snd_us428ctls_vm_fault(struct vm_area_struct *area, struct vm_fault *vmf) { @@ -106,16 +103,6 @@ static unsigned int snd_us428ctls_poll(struct snd_hwdep *hw, struct file *file, } -static int snd_usX2Y_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int snd_usX2Y_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -267,8 +254,6 @@ int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device) hw->iface = SNDRV_HWDEP_IFACE_USX2Y; hw->private_data = usX2Y(card); - hw->ops.open = snd_usX2Y_hwdep_open; - hw->ops.release = snd_usX2Y_hwdep_release; hw->ops.dsp_status = snd_usX2Y_hwdep_dsp_status; hw->ops.dsp_load = snd_usX2Y_hwdep_dsp_load; hw->ops.mmap = snd_us428ctls_mmap; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index af8b84954054..5ce0da23ee96 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -227,9 +227,9 @@ static void i_usX2Y_In04Int(struct urb *urb) if (usX2Y->US04) { if (0 == usX2Y->US04->submitted) - do + do { err = usb_submit_urb(usX2Y->US04->urb[usX2Y->US04->submitted++], GFP_ATOMIC); - while (!err && usX2Y->US04->submitted < usX2Y->US04->len); + } while (!err && usX2Y->US04->submitted < usX2Y->US04->len); } else if (us428ctls && us428ctls->p4outLast >= 0 && us428ctls->p4outLast < N_us428_p4out_BUFS) { if (us428ctls->p4outLast != us428ctls->p4outSent) { diff --git a/sound/usb/usx2y/usx2yhwdeppcm.h b/sound/usb/usx2y/usx2yhwdeppcm.h index c3382fdc386b..9c4fb84b2aa0 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.h +++ b/sound/usb/usx2y/usx2yhwdeppcm.h @@ -18,3 +18,5 @@ struct snd_usX2Y_hwdep_pcm_shm { volatile unsigned captured_iso_frames; int capture_iso_start; }; + +int usX2Y_hwdep_pcm_new(struct snd_card *card); |