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author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 17:32:05 +0200 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 17:32:05 +0200 |
commit | 33081adf8b89d5a716d7e1c60171768d39795b39 (patch) | |
tree | 275de58bbbb5f7ddffcdc087844cfc7fbe4315be /include/sound | |
parent | Merge branch 'for-upstream' of git://git.kernel.org/pub/scm/linux/kernel/git/... (diff) | |
parent | Merge branch 'topic/hda' into for-linus (diff) | |
download | linux-33081adf8b89d5a716d7e1c60171768d39795b39.tar.xz linux-33081adf8b89d5a716d7e1c60171768d39795b39.zip |
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: (365 commits)
ALSA: hda - Disable sticky PCM stream assignment for AD codecs
ALSA: usb - Creative USB X-Fi volume knob support
ALSA: ca0106: Use card specific dac id for mute controls.
ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
ALSA: ca0106: Create a nice spot for mapping channels to dacs.
ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
ALSA: ca0106: Pull out dac powering routine into separate function.
ALSA: ca0106 - add Sound Blaster 5.1vx info.
ASoC: tlv320dac33: Use usleep_range for delays
ALSA: usb-audio: add Novation Launchpad support
ALSA: hda - Add workarounds for CT-IBG controllers
ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
ASoC: tpa6130a2: Error handling for broken chip
ASoC: max98088: Staticise m98088_eq_band
ASoC: soc-core: Fix codec->name memory leak
ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
ALSA: hda - Add some workarounds for Creative IBG
ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
ALSA: hda - Add alc_init_jacks() call to other codecs
...
Diffstat (limited to 'include/sound')
-rw-r--r-- | include/sound/core.h | 2 | ||||
-rw-r--r-- | include/sound/emu10k1.h | 2 | ||||
-rw-r--r-- | include/sound/jack.h | 5 | ||||
-rw-r--r-- | include/sound/max98088.h | 50 | ||||
-rw-r--r-- | include/sound/pcm.h | 1 | ||||
-rw-r--r-- | include/sound/sh_fsi.h | 3 | ||||
-rw-r--r-- | include/sound/soc-dai.h | 98 | ||||
-rw-r--r-- | include/sound/soc-dapm.h | 18 | ||||
-rw-r--r-- | include/sound/soc-of-simple.h | 25 | ||||
-rw-r--r-- | include/sound/soc.h | 245 | ||||
-rw-r--r-- | include/sound/tlv.h | 4 | ||||
-rw-r--r-- | include/sound/tlv320aic3x.h | 43 | ||||
-rw-r--r-- | include/sound/wm8962.h | 32 |
13 files changed, 360 insertions, 168 deletions
diff --git a/include/sound/core.h b/include/sound/core.h index df26ebbfa9c6..1fa2407c966f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -177,7 +177,7 @@ int snd_power_wait(struct snd_card *card, unsigned int power_state); #define snd_power_lock(card) do { (void)(card); } while (0) #define snd_power_unlock(card) do { (void)(card); } while (0) static inline int snd_power_wait(struct snd_card *card, unsigned int state) { return 0; } -#define snd_power_get_state(card) SNDRV_CTL_POWER_D0 +#define snd_power_get_state(card) ({ (void)(card); SNDRV_CTL_POWER_D0; }) #define snd_power_change_state(card, state) do { (void)(card); } while (0) #endif /* CONFIG_PM */ diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 7dc97d12253c..4f865df42f0f 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -438,6 +438,8 @@ #define CCCA_CURRADDR_MASK 0x00ffffff /* Current address of the selected channel */ #define CCCA_CURRADDR 0x18000008 +/* undefine CCR to avoid conflict with the definition for SH */ +#undef CCR #define CCR 0x09 /* Cache control register */ #define CCR_CACHEINVALIDSIZE 0x07190009 #define CCR_CACHEINVALIDSIZE_MASK 0xfe000000 /* Number of invalid samples cache for this channel */ diff --git a/include/sound/jack.h b/include/sound/jack.h index d90b9fa32707..c140fc7cbd3f 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -47,6 +47,9 @@ enum snd_jack_types { SND_JACK_BTN_0 = 0x4000, SND_JACK_BTN_1 = 0x2000, SND_JACK_BTN_2 = 0x1000, + SND_JACK_BTN_3 = 0x0800, + SND_JACK_BTN_4 = 0x0400, + SND_JACK_BTN_5 = 0x0200, }; struct snd_jack { @@ -55,7 +58,7 @@ struct snd_jack { int type; const char *id; char name[100]; - unsigned int key[3]; /* Keep in sync with definitions above */ + unsigned int key[6]; /* Keep in sync with definitions above */ void *private_data; void (*private_free)(struct snd_jack *); }; diff --git a/include/sound/max98088.h b/include/sound/max98088.h new file mode 100644 index 000000000000..c3ba8239182d --- /dev/null +++ b/include/sound/max98088.h @@ -0,0 +1,50 @@ +/* + * Platform data for MAX98088 + * + * Copyright 2010 Maxim Integrated Products + * + * 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 __SOUND_MAX98088_PDATA_H__ +#define __SOUND_MAX98088_PDATA_H__ + +/* Equalizer filter response configuration */ +struct max98088_eq_cfg { + const char *name; + unsigned int rate; + u16 band1[5]; + u16 band2[5]; + u16 band3[5]; + u16 band4[5]; + u16 band5[5]; +}; + +/* codec platform data */ +struct max98088_pdata { + + /* Equalizers for DAI1 and DAI2 */ + struct max98088_eq_cfg *eq_cfg; + unsigned int eq_cfgcnt; + + /* Receiver output can be configured as power amplifier or LINE out */ + /* Set receiver_mode to: + * 0 = amplifier output, or + * 1 = LINE level output + */ + unsigned int receiver_mode:1; + + /* Analog/digital microphone configuration: + * 0 = analog microphone input (normal setting) + * 1 = digital microphone input + */ + unsigned int digmic_left_mode:1; + unsigned int digmic_right_mode:1; + +}; + +#endif diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 85f1c6bf8566..dfd9b76b1853 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -278,6 +278,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ + unsigned long hw_ptr_buffer_jiffies; /* buffer time in jiffies */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ /* -- HW params -- */ diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 9d51d6f35893..fa60cbda90a4 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -114,7 +114,4 @@ struct sh_fsi_platform_info { int (*set_rate)(int is_porta, int rate); /* for master mode */ }; -extern struct snd_soc_dai fsi_soc_dai[2]; -extern struct snd_soc_platform fsi_soc_platform; - #endif /* __SOUND_FSI_H */ diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 377693a14385..e7b680248006 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -91,15 +91,17 @@ struct snd_pcm_substream; SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) -struct snd_soc_dai_ops; +struct snd_soc_dai_driver; 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); +int snd_soc_register_dai(struct device *dev, + struct snd_soc_dai_driver *dai_drv); +void snd_soc_unregister_dai(struct device *dev); +int snd_soc_register_dais(struct device *dev, + struct snd_soc_dai_driver *dai_drv, size_t count); +void snd_soc_unregister_dais(struct device *dev, size_t count); /* Digital Audio Interface clocking API.*/ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -126,16 +128,6 @@ 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 its ALSA, DAI and AC97 - * operations and capabilities. Codec and platform drivers will register this - * structure for every DAI they have. - * - * This structure covers the clocking, formating and ALSA operations for each - * interface. - */ struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. @@ -191,24 +183,24 @@ struct snd_soc_dai_ops { }; /* - * Digital Audio Interface runtime data. + * Digital Audio Interface Driver. * - * Holds runtime data for a DAI. + * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 + * operations and capabilities. Codec and platform drivers will register this + * structure for every DAI they have. + * + * This structure covers the clocking, formating and ALSA operations for each + * interface. */ -struct snd_soc_dai { +struct snd_soc_dai_driver { /* DAI description */ - char *name; + const char *name; unsigned int id; int ac97_control; - struct device *dev; - void *ac97_pdata; /* platform_data for the ac97 codec */ - - /* DAI callbacks */ - int (*probe)(struct platform_device *pdev, - struct snd_soc_dai *dai); - void (*remove)(struct platform_device *pdev, - struct snd_soc_dai *dai); + /* DAI driver callbacks */ + int (*probe)(struct snd_soc_dai *dai); + int (*remove)(struct snd_soc_dai *dai); int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); @@ -219,26 +211,51 @@ struct snd_soc_dai { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; +}; + +/* + * Digital Audio Interface runtime data. + * + * Holds runtime data for a DAI. + */ +struct snd_soc_dai { + const char *name; + int id; + struct device *dev; + void *ac97_pdata; /* platform_data for the ac97 codec */ + + /* driver ops */ + struct snd_soc_dai_driver *driver; /* DAI runtime info */ - struct snd_soc_codec *codec; + unsigned int capture_active:1; /* stream is in use */ + unsigned int playback_active:1; /* stream is in use */ + unsigned int symmetric_rates:1; + struct snd_pcm_runtime *runtime; unsigned int active; unsigned char pop_wait:1; + unsigned char probed:1; - /* DAI private data */ - void *private_data; + /* DAI DMA data */ + void *playback_dma_data; + void *capture_dma_data; - /* parent platform */ - struct snd_soc_platform *platform; + /* parent platform/codec */ + union { + struct snd_soc_platform *platform; + struct snd_soc_codec *codec; + }; + struct snd_soc_card *card; struct list_head list; + struct list_head card_list; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) { return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dai->playback.dma_data : dai->capture.dma_data; + dai->playback_dma_data : dai->capture_dma_data; } static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, @@ -246,9 +263,20 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, void *data) { if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback.dma_data = data; + dai->playback_dma_data = data; else - dai->capture.dma_data = data; + dai->capture_dma_data = data; +} + +static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, + void *data) +{ + dev_set_drvdata(dai->dev, data); +} + +static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) +{ + return dev_get_drvdata(dai->dev); } #endif diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c5d9987bc897..8fd3b41b763f 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -172,9 +172,19 @@ #define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert, \ + .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \ { .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ .reg = wreg, .shift = wshift, .invert = winvert } +#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \ + wevent, wflags) \ +{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \ + .reg = wreg, .shift = wshift, .invert = winvert, \ + .event = wevent, .event_flags = wflags } #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} @@ -322,14 +332,14 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); -void snd_soc_dapm_free(struct snd_soc_device *socdev); +void snd_soc_dapm_free(struct snd_soc_codec *codec); int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, - int event); -void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + const char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/include/sound/soc-of-simple.h b/include/sound/soc-of-simple.h deleted file mode 100644 index a064e1934a56..000000000000 --- a/include/sound/soc-of-simple.h +++ /dev/null @@ -1,25 +0,0 @@ -/* - * OF helpers for ALSA SoC - * - * Copyright (C) 2008, Secret Lab Technologies Ltd. - */ - -#ifndef _INCLUDE_SOC_OF_H_ -#define _INCLUDE_SOC_OF_H_ - -#if defined(CONFIG_SND_SOC_OF_SIMPLE) || defined(CONFIG_SND_SOC_OF_SIMPLE_MODULE) - -#include <linux/of.h> -#include <sound/soc.h> - -int of_snd_soc_register_codec(struct snd_soc_codec_device *codec_dev, - void *codec_data, struct snd_soc_dai *dai, - struct device_node *node); - -int of_snd_soc_register_platform(struct snd_soc_platform *platform, - struct device_node *node, - struct snd_soc_dai *cpu_dai); - -#endif - -#endif /* _INCLUDE_SOC_OF_H_ */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 65e9d03ed4f5..5c3bce83f28a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -214,10 +214,10 @@ * @OFF: Power Off. No restrictions on transition times. */ enum snd_soc_bias_level { - SND_SOC_BIAS_ON, - SND_SOC_BIAS_PREPARE, - SND_SOC_BIAS_STANDBY, SND_SOC_BIAS_OFF, + SND_SOC_BIAS_STANDBY, + SND_SOC_BIAS_PREPARE, + SND_SOC_BIAS_ON, }; struct snd_jack; @@ -228,13 +228,17 @@ struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; +struct snd_soc_dai_driver; struct snd_soc_platform; struct snd_soc_dai_link; +struct snd_soc_platform_driver; struct snd_soc_codec; +struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; + #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio; #endif @@ -249,19 +253,18 @@ enum snd_soc_control_type { SND_SOC_SPI, }; -int snd_soc_register_platform(struct snd_soc_platform *platform); -void snd_soc_unregister_platform(struct snd_soc_platform *platform); -int snd_soc_register_codec(struct snd_soc_codec *codec); -void snd_soc_unregister_codec(struct snd_soc_codec *codec); +int snd_soc_register_platform(struct device *dev, + struct snd_soc_platform_driver *platform_drv); +void snd_soc_unregister_platform(struct device *dev); +int snd_soc_register_codec(struct device *dev, + struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); +void snd_soc_unregister_codec(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control); -/* 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); - /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); @@ -273,7 +276,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, +int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, @@ -382,7 +385,7 @@ struct snd_soc_jack_gpio { int invert; int debounce_time; struct snd_soc_jack *jack; - struct work_struct work; + struct delayed_work work; int (*jack_status_check)(void); }; @@ -390,7 +393,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct snd_jack *jack; - struct snd_soc_card *card; + struct snd_soc_codec *codec; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -398,15 +401,13 @@ struct snd_soc_jack { /* SoC PCM stream information */ struct snd_soc_pcm_stream { - char *stream_name; + const char *stream_name; u64 formats; /* SNDRV_PCM_FMTBIT_* */ unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ - unsigned int active; /* stream is in use */ - void *dma_data; /* used by platform code */ }; /* SoC audio ops */ @@ -419,44 +420,36 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* SoC Audio Codec */ +/* SoC Audio Codec device */ struct snd_soc_codec { - char *name; - struct module *owner; - struct mutex mutex; + const char *name; + int id; struct device *dev; - struct snd_soc_device *socdev; + struct snd_soc_codec_driver *driver; + struct mutex mutex; + struct snd_soc_card *card; struct list_head list; - - /* callbacks */ - int (*set_bias_level)(struct snd_soc_codec *, - enum snd_soc_bias_level level); + struct list_head card_list; + int num_dai; /* runtime */ - struct snd_card *card; struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; - unsigned int pcm_devs; - void *drvdata; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + unsigned int cache_only:1; /* Suppress writes to hardware */ + unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ + unsigned int suspended:1; /* Codec is in suspend PM state */ + unsigned int probed:1; /* Codec has been probed */ + unsigned int ac97_registered:1; /* Codec has been AC97 registered */ + unsigned int ac97_created:1; /* Codec has been created by SoC */ + unsigned int sysfs_registered:1; /* codec has been sysfs registered */ /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*display_register)(struct snd_soc_codec *, char *, - size_t, unsigned int); - int (*volatile_register)(unsigned int); - int (*readable_register)(unsigned int); hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); void *reg_cache; - short reg_cache_size; - short reg_cache_step; - - unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - unsigned int cache_only:1; /* Suppress writes to hardware */ - unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ /* dapm */ u32 pop_time; @@ -466,10 +459,6 @@ struct snd_soc_codec { enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - /* codec DAI's */ - struct snd_soc_dai *dai; - unsigned int num_dai; - #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_codec_root; struct dentry *debugfs_reg; @@ -478,23 +467,40 @@ struct snd_soc_codec { #endif }; -/* codec device */ -struct snd_soc_codec_device { - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, pm_message_t state); - int (*resume)(struct platform_device *pdev); +/* codec driver */ +struct snd_soc_codec_driver { + + /* driver ops */ + int (*probe)(struct snd_soc_codec *); + int (*remove)(struct snd_soc_codec *); + int (*suspend)(struct snd_soc_codec *, + pm_message_t state); + int (*resume)(struct snd_soc_codec *); + + /* codec IO */ + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + int (*display_register)(struct snd_soc_codec *, char *, + size_t, unsigned int); + int (*volatile_register)(unsigned int); + int (*readable_register)(unsigned int); + short reg_cache_size; + short reg_cache_step; + short reg_word_size; + const void *reg_cache_default; + + /* codec bias level */ + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); }; /* SoC platform interface */ -struct snd_soc_platform { - char *name; - struct list_head list; +struct snd_soc_platform_driver { - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct snd_soc_dai_link *dai_link); - int (*resume)(struct snd_soc_dai_link *dai_link); + int (*probe)(struct snd_soc_platform *); + int (*remove)(struct snd_soc_platform *); + 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 *, @@ -509,23 +515,31 @@ struct snd_soc_platform { struct snd_soc_dai *); /* platform stream ops */ - struct snd_pcm_ops *pcm_ops; + struct snd_pcm_ops *ops; }; -/* SoC machine DAI configuration, glues a codec and cpu DAI together */ -struct snd_soc_dai_link { - char *name; /* Codec name */ - char *stream_name; /* Stream name */ +struct snd_soc_platform { + const char *name; + int id; + struct device *dev; + struct snd_soc_platform_driver *driver; - /* DAI */ - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + unsigned int suspended:1; /* platform is suspended */ + unsigned int probed:1; - /* machine stream operations */ - struct snd_soc_ops *ops; + struct snd_soc_card *card; + struct list_head list; + struct list_head card_list; +}; - /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_codec *codec); +struct snd_soc_dai_link { + /* config - must be set by machine driver */ + const char *name; /* Codec name */ + const char *stream_name; /* Stream name */ + const char *codec_name; /* for multi-codec */ + const char *platform_name; /* for multi-platform */ + const char *cpu_dai_name; + const char *codec_dai_name; /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; @@ -533,21 +547,24 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; + /* codec/machine specific init - e.g. add machine controls */ + int (*init)(struct snd_soc_pcm_runtime *rtd); - /* DAI pcm */ - struct snd_pcm *pcm; + /* machine stream operations */ + struct snd_soc_ops *ops; }; /* SoC card */ struct snd_soc_card { - char *name; + const char *name; struct device *dev; + struct snd_card *snd_card; + struct module *owner; struct list_head list; + struct mutex mutex; - int instantiated; + bool instantiated; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -568,28 +585,38 @@ struct snd_soc_card { /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + struct snd_soc_pcm_runtime *rtd; + int num_rtd; - struct snd_soc_device *socdev; - - struct snd_soc_codec *codec; - - struct snd_soc_platform *platform; - struct delayed_work delayed_work; struct work_struct deferred_resume_work; + + /* lists of probed devices belonging to this card */ + struct list_head codec_dev_list; + struct list_head platform_dev_list; + struct list_head dai_dev_list; }; -/* SoC Device - the audio subsystem */ -struct snd_soc_device { - struct device *dev; +/* SoC machine DAI configuration, glues a codec and cpu DAI together */ +struct snd_soc_pcm_runtime { + struct device dev; struct snd_soc_card *card; - struct snd_soc_codec_device *codec_dev; - void *codec_data; -}; + struct snd_soc_dai_link *dai_link; + + unsigned int complete:1; + unsigned int dev_registered:1; + + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + long pmdown_time; -/* runtime channel data */ -struct snd_soc_pcm_runtime { - struct snd_soc_dai_link *dai; - struct snd_soc_device *socdev; + /* runtime devices */ + struct snd_pcm *pcm; + struct snd_soc_codec *codec; + struct snd_soc_platform *platform; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; + + struct delayed_work delayed_work; }; /* mixer control */ @@ -615,24 +642,48 @@ struct soc_enum { static inline unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg) { - return codec->read(codec, reg); + return codec->driver->read(codec, reg); } static inline unsigned int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - return codec->write(codec, reg, val); + return codec->driver->write(codec, reg, val); } +/* device driver data */ + static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, - void *data) + void *data) { - codec->drvdata = data; + dev_set_drvdata(codec->dev, data); } static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec) { - return codec->drvdata; + return dev_get_drvdata(codec->dev); +} + +static inline void snd_soc_platform_set_drvdata(struct snd_soc_platform *platform, + void *data) +{ + dev_set_drvdata(platform->dev, data); +} + +static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platform) +{ + return dev_get_drvdata(platform->dev); +} + +static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd, + void *data) +{ + dev_set_drvdata(&rtd->dev, data); +} + +static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) +{ + return dev_get_drvdata(&rtd->dev); } #include <sound/soc-dai.h> diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 9fd5b19ccf5c..7067e2dfb0b9 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -38,9 +38,11 @@ #define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */ #define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */ +#define TLV_DB_SCALE_MASK 0xffff +#define TLV_DB_SCALE_MUTE 0x10000 #define TLV_DB_SCALE_ITEM(min, step, mute) \ SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \ - (min), ((step) & 0xffff) | ((mute) ? 0x10000 : 0) + (min), ((step) & TLV_DB_SCALE_MASK) | ((mute) ? TLV_DB_SCALE_MUTE : 0) #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \ unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) } diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index b1a5f34e5cfa..99e0308bf2c2 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -10,8 +10,49 @@ #ifndef __TLV320AIC3x_H__ #define __TLV320AIC3x_H__ +/* GPIO API */ +enum { + AIC3X_GPIO1_FUNC_DISABLED = 0, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK_ADC = 1, + AIC3X_GPIO1_FUNC_CLOCK_MUX = 2, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV2 = 3, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV4 = 4, + AIC3X_GPIO1_FUNC_CLOCK_MUX_DIV8 = 5, + AIC3X_GPIO1_FUNC_SHORT_CIRCUIT_IRQ = 6, + AIC3X_GPIO1_FUNC_AGC_NOISE_IRQ = 7, + AIC3X_GPIO1_FUNC_INPUT = 8, + AIC3X_GPIO1_FUNC_OUTPUT = 9, + AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK = 10, + AIC3X_GPIO1_FUNC_AUDIO_WORDCLK = 11, + AIC3X_GPIO1_FUNC_BUTTON_IRQ = 12, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_IRQ = 13, + AIC3X_GPIO1_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 14, + AIC3X_GPIO1_FUNC_ALL_IRQ = 16 +}; + +enum { + AIC3X_GPIO2_FUNC_DISABLED = 0, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_IRQ = 2, + AIC3X_GPIO2_FUNC_INPUT = 3, + AIC3X_GPIO2_FUNC_OUTPUT = 4, + AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT = 5, + AIC3X_GPIO2_FUNC_AUDIO_BITCLK = 8, + AIC3X_GPIO2_FUNC_HEADSET_DETECT_OR_BUTTON_IRQ = 9, + AIC3X_GPIO2_FUNC_ALL_IRQ = 10, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_OR_AGC_IRQ = 11, + AIC3X_GPIO2_FUNC_HEADSET_OR_BUTTON_PRESS_OR_SHORT_CIRCUIT_IRQ = 12, + AIC3X_GPIO2_FUNC_SHORT_CIRCUIT_IRQ = 13, + AIC3X_GPIO2_FUNC_AGC_NOISE_IRQ = 14, + AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 +}; + +struct aic3x_setup_data { + unsigned int gpio_func[2]; +}; + struct aic3x_pdata { int gpio_reset; /* < 0 if not used */ + struct aic3x_setup_data *setup; }; -#endif
\ No newline at end of file +#endif diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h new file mode 100644 index 000000000000..2b5306c503fb --- /dev/null +++ b/include/sound/wm8962.h @@ -0,0 +1,32 @@ +/* + * wm8962.h -- WM8962 Soc Audio driver platform data + * + * 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 _WM8962_PDATA_H +#define _WM8962_PDATA_H + +#define WM8962_MAX_GPIO 6 + +/* Use to set GPIO default values to zero */ +#define WM8962_GPIO_SET 0x10000 + +struct wm8962_pdata { + int gpio_base; + u32 gpio_init[WM8962_MAX_GPIO]; + + /* Setup for microphone detection, raw value to be written to + * R48(0x30) - only microphone related bits will be updated. + * Detection may be enabled here for use with signals brought + * out on the GPIOs. */ + u32 mic_cfg; + + bool irq_active_low; + + bool spk_mono; /* Speaker outputs tied together as mono */ +}; + +#endif |