diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2008-12-28 22:41:32 +0300 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2008-12-28 22:41:32 +0300 |
commit | cb10ea549fdc0ab2dd8988adab5bf40b4fa642f3 (patch) | |
tree | 6bc11e0af9f0639a5eedd055401086c8c771f21e /include | |
parent | 81d6e59dabb1ae0c782e9eb7e3d88f699d25b314 (diff) | |
parent | 5ce442fe2c9423ec5451222aee6f9b2127bb8311 (diff) | |
download | linux-cb10ea549fdc0ab2dd8988adab5bf40b4fa642f3.tar.xz |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (367 commits)
ALSA: ASoC: fix a typo in omp-pcm.c
ASoC: Fix DSP formats in SSM2602 audio codec
ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers
ALSA: hda: fix incorrect mixer index values for 92hd83xx
ALSA: hda: dinput_mux check
ALSA: hda - Add quirk for another HP dv7
ALSA: ASoC - Add missing __devexit annotation to wm8350.c
ALSA: ASoc: DaVinci: davinci-evm use dsp_b mode
ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_dai
ALSA: ASoC: tlv320aic3x add dsp_a
ALSA: ASoC: DaVinci: document I2S limitations
ALSA: ASoC: DaVinci: davinci-i2s clean up
ALSA: ASoC: DaVinci: davinci-i2s clean up
ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarity
ALSA: ASoC: DaVinci: davinvi-evm, make requests explicit
ALSA: ca0106 - disable 44.1kHz capture
ALSA: ca0106 - Add missing card->private_data initialization
ALSA: ca0106 - Check ac97 availability at PM
ALSA: hda - Power up always when no jack detection is available
ALSA: hda - Fix unused variable warnings in patch_sigmatel.c
...
Diffstat (limited to 'include')
-rw-r--r-- | include/linux/input.h | 2 | ||||
-rw-r--r-- | include/linux/mfd/wm8350/audio.h | 38 | ||||
-rw-r--r-- | include/sound/ac97_codec.h | 2 | ||||
-rw-r--r-- | include/sound/asound.h | 1 | ||||
-rw-r--r-- | include/sound/core.h | 28 | ||||
-rw-r--r-- | include/sound/info.h | 106 | ||||
-rw-r--r-- | include/sound/jack.h | 2 | ||||
-rw-r--r-- | include/sound/l3.h | 18 | ||||
-rw-r--r-- | include/sound/s3c24xx_uda134x.h | 14 | ||||
-rw-r--r-- | include/sound/soc-dai.h | 231 | ||||
-rw-r--r-- | include/sound/soc-dapm.h | 2 | ||||
-rw-r--r-- | include/sound/soc.h | 206 | ||||
-rw-r--r-- | include/sound/uda134x.h | 26 | ||||
-rw-r--r-- | include/sound/version.h | 2 |
14 files changed, 449 insertions, 229 deletions
diff --git a/include/linux/input.h b/include/linux/input.h index 5341e8251f8c..9a6355f74db2 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -659,6 +659,8 @@ struct input_absinfo { #define SW_RADIO SW_RFKILL_ALL /* deprecated */ #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ +#define SW_LINEOUT_INSERT 0x06 /* set = inserted */ +#define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index 217bb22ebb8e..af95a1d2f3a1 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -1,7 +1,7 @@ /* * audio.h -- Audio Driver for Wolfson WM8350 PMIC * - * Copyright 2007 Wolfson Microelectronics PLC + * Copyright 2007, 2008 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 @@ -70,9 +70,9 @@ #define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */ #define WM8350_VMID_OFF 0 -#define WM8350_VMID_500K 1 -#define WM8350_VMID_100K 2 -#define WM8350_VMID_10K 3 +#define WM8350_VMID_300K 1 +#define WM8350_VMID_50K 2 +#define WM8350_VMID_5K 3 /* * R40 (0x28) - Clock Control 1 @@ -591,8 +591,38 @@ #define WM8350_IRQ_CODEC_MICSCD 41 #define WM8350_IRQ_CODEC_MICD 42 +/* + * WM8350 Platform data. + * + * This must be initialised per platform for best audio performance. + * Please see WM8350 datasheet for information. + */ +struct wm8350_audio_platform_data { + int vmid_discharge_msecs; /* VMID --> OFF discharge time */ + int drain_msecs; /* OFF drain time */ + int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */ + int vmid_charge_msecs; /* vmid power up time */ + u32 vmid_s_curve:2; /* vmid enable s curve speed */ + u32 dis_out4:2; /* out4 discharge speed */ + u32 dis_out3:2; /* out3 discharge speed */ + u32 dis_out2:2; /* out2 discharge speed */ + u32 dis_out1:2; /* out1 discharge speed */ + u32 vroi_out4:1; /* out4 tie off */ + u32 vroi_out3:1; /* out3 tie off */ + u32 vroi_out2:1; /* out2 tie off */ + u32 vroi_out1:1; /* out1 tie off */ + u32 vroi_enable:1; /* enable tie off */ + u32 codec_current_on:2; /* current level ON */ + u32 codec_current_standby:2; /* current level STANDBY */ + u32 codec_current_charge:2; /* codec current @ vmid charge */ +}; + +struct snd_soc_codec; + struct wm8350_codec { struct platform_device *pdev; + struct snd_soc_codec *codec; + struct wm8350_audio_platform_data *platform_data; }; #endif diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 9c309daf492b..251fc1cd5002 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -281,10 +281,12 @@ /* specific - Analog Devices */ #define AC97_AD_TEST 0x5a /* test register */ #define AC97_AD_TEST2 0x5c /* undocumented test register 2 */ +#define AC97_AD_HPFD_SHIFT 12 /* High Pass Filter Disable */ #define AC97_AD_CODEC_CFG 0x70 /* codec configuration */ #define AC97_AD_JACK_SPDIF 0x72 /* Jack Sense & S/PDIF */ #define AC97_AD_SERIAL_CFG 0x74 /* Serial Configuration */ #define AC97_AD_MISC 0x76 /* Misc Control Bits */ +#define AC97_AD_VREFD_SHIFT 2 /* V_REFOUT Disable (AD1888) */ /* specific - Cirrus Logic */ #define AC97_CSR_ACMODE 0x5e /* AC Mode Register */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 2c4dc908a54a..1c02ed1d7c4a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -575,6 +575,7 @@ enum { #define SNDRV_TIMER_GLOBAL_SYSTEM 0 #define SNDRV_TIMER_GLOBAL_RTC 1 #define SNDRV_TIMER_GLOBAL_HPET 2 +#define SNDRV_TIMER_GLOBAL_HRTIMER 3 /* info flags */ #define SNDRV_TIMER_FLG_SLAVE (1<<0) /* cannot be controlled */ diff --git a/include/sound/core.h b/include/sound/core.h index 1508c4ec1ba9..f632484bc743 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -353,7 +353,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) * snd_printk - printk wrapper * @fmt: format string * - * Works like print() but prints the file and the line of the caller + * Works like printk() but prints the file and the line of the caller * when configured with CONFIG_SND_VERBOSE_PRINTK. */ #define snd_printk(fmt, args...) \ @@ -380,18 +380,40 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...) printk(fmt ,##args) #endif +/** + * snd_BUG - give a BUG warning message and stack trace + * + * Calls WARN() if CONFIG_SND_DEBUG is set. + * Ignored when CONFIG_SND_DEBUG is not set. + */ #define snd_BUG() WARN(1, "BUG?\n") + +/** + * snd_BUG_ON - debugging check macro + * @cond: condition to evaluate + * + * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition, + * and call WARN() and returns the value if it's non-zero. + * + * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given + * condition is ignored. + * + * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n. + * Thus, don't put any statement that influences on the code behavior, + * such as pre/post increment, to the argument of this macro. + * If you want to evaluate and give a warning, use standard WARN_ON(). + */ #define snd_BUG_ON(cond) WARN((cond), "BUG? (%s)\n", __stringify(cond)) #else /* !CONFIG_SND_DEBUG */ #define snd_printd(fmt, args...) do { } while (0) #define snd_BUG() do { } while (0) -static inline int __snd_bug_on(void) +static inline int __snd_bug_on(int cond) { return 0; } -#define snd_BUG_ON(cond) __snd_bug_on() /* always false */ +#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */ #endif /* CONFIG_SND_DEBUG */ diff --git a/include/sound/info.h b/include/sound/info.h index 8ae72e74f898..7c2ee1a21b00 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -40,30 +40,34 @@ struct snd_info_buffer { struct snd_info_entry; struct snd_info_entry_text { - void (*read) (struct snd_info_entry *entry, struct snd_info_buffer *buffer); - void (*write) (struct snd_info_entry *entry, struct snd_info_buffer *buffer); + void (*read)(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); + void (*write)(struct snd_info_entry *entry, + struct snd_info_buffer *buffer); }; struct snd_info_entry_ops { - int (*open) (struct snd_info_entry *entry, - unsigned short mode, void **file_private_data); - int (*release) (struct snd_info_entry * entry, - unsigned short mode, void *file_private_data); - long (*read) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, char __user *buf, + int (*open)(struct snd_info_entry *entry, + unsigned short mode, void **file_private_data); + int (*release)(struct snd_info_entry *entry, + unsigned short mode, void *file_private_data); + long (*read)(struct snd_info_entry *entry, void *file_private_data, + struct file *file, char __user *buf, + unsigned long count, unsigned long pos); + long (*write)(struct snd_info_entry *entry, void *file_private_data, + struct file *file, const char __user *buf, unsigned long count, unsigned long pos); - long (*write) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, const char __user *buf, - unsigned long count, unsigned long pos); - long long (*llseek) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, long long offset, int orig); - unsigned int (*poll) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, poll_table * wait); - int (*ioctl) (struct snd_info_entry *entry, void *file_private_data, - struct file * file, unsigned int cmd, unsigned long arg); - int (*mmap) (struct snd_info_entry *entry, void *file_private_data, - struct inode * inode, struct file * file, - struct vm_area_struct * vma); + long long (*llseek)(struct snd_info_entry *entry, + void *file_private_data, struct file *file, + long long offset, int orig); + unsigned int(*poll)(struct snd_info_entry *entry, + void *file_private_data, struct file *file, + poll_table *wait); + int (*ioctl)(struct snd_info_entry *entry, void *file_private_data, + struct file *file, unsigned int cmd, unsigned long arg); + int (*mmap)(struct snd_info_entry *entry, void *file_private_data, + struct inode *inode, struct file *file, + struct vm_area_struct *vma); }; struct snd_info_entry { @@ -106,34 +110,37 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer); static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {} #endif -int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3))); +int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \ + __attribute__ ((format (printf, 2, 3))); int snd_info_init(void); int snd_info_done(void); -int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len); +int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len); char *snd_info_get_str(char *dest, char *src, int len); -struct snd_info_entry *snd_info_create_module_entry(struct module * module, +struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, - struct snd_info_entry * parent); -struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, + struct snd_info_entry *parent); +struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, - struct snd_info_entry * parent); -void snd_info_free_entry(struct snd_info_entry * entry); -int snd_info_store_text(struct snd_info_entry * entry); -int snd_info_restore_text(struct snd_info_entry * entry); - -int snd_info_card_create(struct snd_card * card); -int snd_info_card_register(struct snd_card * card); -int snd_info_card_free(struct snd_card * card); -void snd_info_card_disconnect(struct snd_card * card); -int snd_info_register(struct snd_info_entry * entry); + struct snd_info_entry *parent); +void snd_info_free_entry(struct snd_info_entry *entry); +int snd_info_store_text(struct snd_info_entry *entry); +int snd_info_restore_text(struct snd_info_entry *entry); + +int snd_info_card_create(struct snd_card *card); +int snd_info_card_register(struct snd_card *card); +int snd_info_card_free(struct snd_card *card); +void snd_info_card_disconnect(struct snd_card *card); +void snd_info_card_id_change(struct snd_card *card); +int snd_info_register(struct snd_info_entry *entry); /* for card drivers */ -int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp); +int snd_card_proc_new(struct snd_card *card, const char *name, + struct snd_info_entry **entryp); static inline void snd_info_set_text_ops(struct snd_info_entry *entry, - void *private_data, - void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) + void *private_data, + void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) { entry->private_data = private_data; entry->c.text.read = read; @@ -146,21 +153,22 @@ int snd_info_check_reserved_words(const char *str); #define snd_seq_root NULL #define snd_oss_root NULL -static inline int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) { return 0; } +static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; } static inline int snd_info_init(void) { return 0; } static inline int snd_info_done(void) { return 0; } -static inline int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len) { return 0; } +static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; } static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; } -static inline struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry * parent) { return NULL; } -static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, const char *name, struct snd_info_entry * parent) { return NULL; } -static inline void snd_info_free_entry(struct snd_info_entry * entry) { ; } - -static inline int snd_info_card_create(struct snd_card * card) { return 0; } -static inline int snd_info_card_register(struct snd_card * card) { return 0; } -static inline int snd_info_card_free(struct snd_card * card) { return 0; } -static inline void snd_info_card_disconnect(struct snd_card * card) { } -static inline int snd_info_register(struct snd_info_entry * entry) { return 0; } +static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; } +static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; } +static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; } + +static inline int snd_info_card_create(struct snd_card *card) { return 0; } +static inline int snd_info_card_register(struct snd_card *card) { return 0; } +static inline int snd_info_card_free(struct snd_card *card) { return 0; } +static inline void snd_info_card_disconnect(struct snd_card *card) { } +static inline void snd_info_card_id_change(struct snd_card *card) { } +static inline int snd_info_register(struct snd_info_entry *entry) { return 0; } static inline int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp) { return -EINVAL; } diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b59adb..2e0315cdd0d6 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -35,6 +35,8 @@ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, + SND_JACK_LINEOUT = 0x0004, + SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ }; struct snd_jack { diff --git a/include/sound/l3.h b/include/sound/l3.h new file mode 100644 index 000000000000..423a08f0f1b0 --- /dev/null +++ b/include/sound/l3.h @@ -0,0 +1,18 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +struct l3_pins { + void (*setdat)(int); + void (*setclk)(int); + void (*setmode)(int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; +}; + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h new file mode 100644 index 000000000000..33df4cb909d3 --- /dev/null +++ b/include/sound/s3c24xx_uda134x.h @@ -0,0 +1,14 @@ +#ifndef _S3C24XX_UDA134X_H_ +#define _S3C24XX_UDA134X_H_ 1 + +#include <sound/uda134x.h> + +struct s3c24xx_uda134x_platform_data { + int l3_clk; + int l3_mode; + int l3_data; + void (*power) (int); + int model; +}; + +#endif diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h new file mode 100644 index 000000000000..24247f763608 --- /dev/null +++ b/include/sound/soc-dai.h @@ -0,0 +1,231 @@ +/* + * linux/sound/soc-dai.h -- ALSA SoC Layer + * + * Copyright: 2005-2008 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Digital Audio Interface (DAI) API. + */ + +#ifndef __LINUX_SND_SOC_DAI_H +#define __LINUX_SND_SOC_DAI_H + + +#include <linux/list.h> + +struct snd_pcm_substream; + +/* + * DAI hardware audio formats. + * + * Describes the physical PCM data formating and clocking. Add new formats + * to the end. + */ +#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ +#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ +#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ +#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM LRC */ +#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ + +/* left and right justified also known as MSB and LSB respectively */ +#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J +#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J + +/* + * DAI Clock gating. + * + * DAI bit clocks can be be gated (disabled) when not the DAI is not + * sending or receiving PCM data in a frame. This can be used to save power. + */ +#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ +#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ + +/* + * DAI Left/Right Clocks. + * + * Specifies whether the DAI can support different samples for similtanious + * playback and capture. This usually requires a seperate physical frame + * clock for playback and capture. + */ +#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ +#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ + +/* + * TDM + * + * Time Division Multiplexing. Allows PCM data to be multplexed with other + * data on the DAI. + */ +#define SND_SOC_DAIFMT_TDM (1 << 6) + +/* + * DAI hardware signal inversions. + * + * Specifies whether the DAI can also support inverted clocks for the specified + * format. + */ +#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ +#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ +#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ + +/* + * DAI hardware clock masters. + * + * This is wrt the codec, the inverse is true for the interface + * i.e. if the codec is clk and frm master then the interface is + * clk and frame slave. + */ +#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ +#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ +#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ +#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ + +#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f +#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 +#define SND_SOC_DAIFMT_INV_MASK 0x0f00 +#define SND_SOC_DAIFMT_MASTER_MASK 0xf000 + +/* + * Master Clock Directions + */ +#define SND_SOC_CLOCK_IN 0 +#define SND_SOC_CLOCK_OUT 1 + +struct snd_soc_dai_ops; +struct snd_soc_dai; +struct snd_ac97_bus_ops; + +/* Digital Audio Interface registration */ +int snd_soc_register_dai(struct snd_soc_dai *dai); +void snd_soc_unregister_dai(struct snd_soc_dai *dai); +int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); +void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); + +/* Digital Audio Interface clocking API.*/ +int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); + +int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, + int div_id, int div); + +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + +/* Digital Audio interface formatting */ +int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); + +int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int mask, int slots); + +int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); + +/* Digital Audio Interface mute */ +int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); + +/* + * Digital Audio Interface. + * + * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97 + * operations an capabilities. Codec and platfom drivers will register a this + * structure for every DAI they have. + * + * This structure covers the clocking, formating and ALSA operations for each + * interface a + */ +struct snd_soc_dai_ops { + /* + * DAI clocking configuration, all optional. + * Called by soc_card drivers, normally in their hw_params. + */ + int (*set_sysclk)(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_dai *dai, + int pll_id, unsigned int freq_in, unsigned int freq_out); + int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + + /* + * DAI format configuration + * Called by soc_card drivers, normally in their hw_params. + */ + int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); + int (*set_tdm_slot)(struct snd_soc_dai *dai, + unsigned int mask, int slots); + int (*set_tristate)(struct snd_soc_dai *dai, int tristate); + + /* + * DAI digital mute - optional. + * Called by soc-core to minimise any pops. + */ + int (*digital_mute)(struct snd_soc_dai *dai, int mute); + + /* + * ALSA PCM audio operations - all optional. + * Called by soc-core during audio PCM operations. + */ + int (*startup)(struct snd_pcm_substream *, + struct snd_soc_dai *); + void (*shutdown)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*hw_params)(struct snd_pcm_substream *, + struct snd_pcm_hw_params *, struct snd_soc_dai *); + int (*hw_free)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*prepare)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*trigger)(struct snd_pcm_substream *, int, + struct snd_soc_dai *); +}; + +/* + * Digital Audio Interface runtime data. + * + * Holds runtime data for a DAI. + */ +struct snd_soc_dai { + /* DAI description */ + char *name; + unsigned int id; + int ac97_control; + + struct device *dev; + + /* DAI callbacks */ + int (*probe)(struct platform_device *pdev, + struct snd_soc_dai *dai); + void (*remove)(struct platform_device *pdev, + struct snd_soc_dai *dai); + int (*suspend)(struct snd_soc_dai *dai); + int (*resume)(struct snd_soc_dai *dai); + + /* ops */ + struct snd_soc_dai_ops ops; + + /* DAI capabilities */ + struct snd_soc_pcm_stream capture; + struct snd_soc_pcm_stream playback; + + /* DAI runtime info */ + struct snd_pcm_runtime *runtime; + struct snd_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; + void *dma_data; + + /* DAI private data */ + void *private_data; + + /* parent codec/platform */ + union { + struct snd_soc_codec *codec; + struct snd_soc_platform *platform; + }; + + struct list_head list; +}; + +#endif diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ca699a3017f3..7ee2f70ca42e 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, int num); /* dapm path setup */ -int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec, - const char *sink_name, const char *control_name, const char *src_name); int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); void snd_soc_dapm_free(struct snd_soc_device *socdev); int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, diff --git a/include/sound/soc.h b/include/sound/soc.h index 5e0189876afd..f86e455d3828 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -21,8 +21,6 @@ #include <sound/control.h> #include <sound/ac97_codec.h> -#define SND_SOC_VERSION "0.13.2" - /* * Convenience kcontrol builders */ @@ -145,105 +143,31 @@ enum snd_soc_bias_level { SND_SOC_BIAS_OFF, }; -/* - * Digital Audio Interface (DAI) types - */ -#define SND_SOC_DAI_AC97 0x1 -#define SND_SOC_DAI_I2S 0x2 -#define SND_SOC_DAI_PCM 0x4 -#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */ - -/* - * DAI hardware audio formats - */ -#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ -#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right justified mode */ -#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data msb after FRM or LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM or LRC */ -#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ - -#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J -#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J - -/* - * DAI Gating - */ -#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ -#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */ - -/* - * DAI Sync - * Synchronous LR (Left Right) clocks and Frame signals. - */ -#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */ -#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */ - -/* - * TDM - */ -#define SND_SOC_DAIFMT_TDM (1 << 6) - -/* - * DAI hardware signal inversions - */ -#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal bclk + inv frm */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert bclk + nor frm */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert bclk + frm */ - -/* - * DAI hardware clock masters - * This is wrt the codec, the inverse is true for the interface - * i.e. if the codec is clk and frm master then the interface is - * clk and frame slave. - */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & frm master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & frm master */ -#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & frm slave */ - -#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f -#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 -#define SND_SOC_DAIFMT_INV_MASK 0x0f00 -#define SND_SOC_DAIFMT_MASTER_MASK 0xf000 - - -/* - * Master Clock Directions - */ -#define SND_SOC_CLOCK_IN 0 -#define SND_SOC_CLOCK_OUT 1 - -/* - * AC97 codec ID's bitmask - */ -#define SND_SOC_DAI_AC97_ID0 (1 << 0) -#define SND_SOC_DAI_AC97_ID1 (1 << 1) -#define SND_SOC_DAI_AC97_ID2 (1 << 2) -#define SND_SOC_DAI_AC97_ID3 (1 << 3) - struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; +struct snd_soc_platform; struct snd_soc_codec; -struct snd_soc_machine_config; struct soc_enum; struct snd_soc_ac97_ops; -struct snd_soc_clock_info; typedef int (*hw_write_t)(void *,const char* ,int); typedef int (*hw_read_t)(void *,char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; +int snd_soc_register_platform(struct snd_soc_platform *platform); +void snd_soc_unregister_platform(struct snd_soc_platform *platform); +int snd_soc_register_codec(struct snd_soc_codec *codec); +void snd_soc_unregister_codec(struct snd_soc_codec *codec); + /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); -int snd_soc_register_card(struct snd_soc_device *socdev); +int snd_soc_init_card(struct snd_soc_device *socdev); /* set runtime hw params */ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, @@ -263,27 +187,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, struct snd_ac97_bus_ops *ops, int num); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); -/* Digital Audio Interface clocking API.*/ -int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, - unsigned int freq, int dir); - -int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, - int div_id, int div); - -int snd_soc_dai_set_pll(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - -/* Digital Audio interface formatting */ -int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); - -int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); - -int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); - -/* Digital Audio Interface mute */ -int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); - /* *Controls */ @@ -341,66 +244,14 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* ASoC DAI ops */ -struct snd_soc_dai_ops { - /* DAI clocking configuration */ - int (*set_sysclk)(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir); - int (*set_pll)(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out); - int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); - - /* DAI format configuration */ - int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); - int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); - int (*set_tristate)(struct snd_soc_dai *dai, int tristate); - - /* digital mute */ - int (*digital_mute)(struct snd_soc_dai *dai, int mute); -}; - -/* SoC DAI (Digital Audio Interface) */ -struct snd_soc_dai { - /* DAI description */ - char *name; - unsigned int id; - unsigned char type; - - /* DAI callbacks */ - int (*probe)(struct platform_device *pdev, - struct snd_soc_dai *dai); - void (*remove)(struct platform_device *pdev, - struct snd_soc_dai *dai); - int (*suspend)(struct platform_device *pdev, - struct snd_soc_dai *dai); - int (*resume)(struct platform_device *pdev, - struct snd_soc_dai *dai); - - /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; - - /* DAI capabilities */ - struct snd_soc_pcm_stream capture; - struct snd_soc_pcm_stream playback; - - /* DAI runtime info */ - struct snd_pcm_runtime *runtime; - struct snd_soc_codec *codec; - unsigned int active; - unsigned char pop_wait:1; - void *dma_data; - - /* DAI private data */ - void *private_data; -}; - /* SoC Audio Codec */ struct snd_soc_codec { char *name; struct module *owner; struct mutex mutex; + struct device *dev; + + struct list_head list; /* callbacks */ int (*set_bias_level)(struct snd_soc_codec *, @@ -426,6 +277,7 @@ struct snd_soc_codec { short reg_cache_step; /* dapm */ + u32 pop_time; struct list_head dapm_widgets; struct list_head dapm_paths; enum snd_soc_bias_level bias_level; @@ -435,6 +287,11 @@ struct snd_soc_codec { /* codec DAI's */ struct snd_soc_dai *dai; unsigned int num_dai; + +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_reg; + struct dentry *debugfs_pop_time; +#endif }; /* codec device */ @@ -448,13 +305,12 @@ struct snd_soc_codec_device { /* SoC platform interface */ struct snd_soc_platform { char *name; + struct list_head list; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, - struct snd_soc_dai *dai); - int (*resume)(struct platform_device *pdev, - struct snd_soc_dai *dai); + int (*suspend)(struct snd_soc_dai *dai); + int (*resume)(struct snd_soc_dai *dai); /* pcm creation and destruction */ int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, @@ -484,9 +340,14 @@ struct snd_soc_dai_link { struct snd_pcm *pcm; }; -/* SoC machine */ -struct snd_soc_machine { +/* SoC card */ +struct snd_soc_card { char *name; + struct device *dev; + + struct list_head list; + + int instantiated; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -499,23 +360,26 @@ struct snd_soc_machine { int (*resume_post)(struct platform_device *pdev); /* callbacks */ - int (*set_bias_level)(struct snd_soc_machine *, + int (*set_bias_level)(struct snd_soc_card *, enum snd_soc_bias_level level); /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + + struct snd_soc_device *socdev; + + struct snd_soc_platform *platform; + struct delayed_work delayed_work; + struct work_struct deferred_resume_work; }; /* SoC Device - the audio subsystem */ struct snd_soc_device { struct device *dev; - struct snd_soc_machine *machine; - struct snd_soc_platform *platform; + struct snd_soc_card *card; struct snd_soc_codec *codec; struct snd_soc_codec_device *codec_dev; - struct delayed_work delayed_work; - struct work_struct deferred_resume_work; void *codec_data; }; @@ -542,4 +406,6 @@ struct soc_enum { void *dapm; }; +#include <sound/soc-dai.h> + #endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h new file mode 100644 index 000000000000..475ef8bb7dcd --- /dev/null +++ b/include/sound/uda134x.h @@ -0,0 +1,26 @@ +/* + * uda134x.h -- UDA134x ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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 _UDA134X_H +#define _UDA134X_H + +#include <sound/l3.h> + +struct uda134x_platform_data { + struct l3_pins l3; + void (*power) (int); + int model; +#define UDA134X_UDA1340 1 +#define UDA134X_UDA1341 2 +#define UDA134X_UDA1344 3 +}; + +#endif /* _UDA134X_H */ diff --git a/include/sound/version.h b/include/sound/version.h index 4aafeda88634..2b48237e23bf 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.18rc3" +#define CONFIG_SND_VERSION "1.0.18a" #define CONFIG_SND_DATE "" |