diff options
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/ac97_codec.h | 2 | ||||
| -rw-r--r-- | include/sound/core.h | 37 | ||||
| -rw-r--r-- | include/sound/jack.h | 2 | ||||
| -rw-r--r-- | include/sound/l3.h | 18 | ||||
| -rw-r--r-- | include/sound/memalloc.h | 2 | ||||
| -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 | 3 | ||||
| -rw-r--r-- | include/sound/soc.h | 209 | ||||
| -rw-r--r-- | include/sound/tea575x-tuner.h | 1 | ||||
| -rw-r--r-- | include/sound/uda134x.h | 26 | ||||
| -rw-r--r-- | include/sound/version.h | 2 |
12 files changed, 364 insertions, 183 deletions
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 9c309da..251fc1c 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/core.h b/include/sound/core.h index e5eec5f..f632484 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -43,9 +43,6 @@ #ifdef CONFIG_PCI struct pci_dev; #endif -#ifdef CONFIG_SBUS -struct sbus_dev; -#endif /* device allocation stuff */ @@ -356,7 +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...) \ @@ -383,14 +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...) /* nothing */ -#define snd_BUG() /* nothing */ -#define snd_BUG_ON(cond) ({/*(void)(cond);*/ 0;}) /* always false */ +#define snd_printd(fmt, args...) do { } while (0) +#define snd_BUG() do { } while (0) +static inline int __snd_bug_on(int cond) +{ + return 0; +} +#define snd_BUG_ON(cond) __snd_bug_on(0 && (cond)) /* always false */ #endif /* CONFIG_SND_DEBUG */ diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b..2e0315c 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 0000000..423a08f --- /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/memalloc.h b/include/sound/memalloc.h index d787a6b..7ccce94 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -37,7 +37,6 @@ struct snd_dma_device { #ifndef snd_dma_pci_data #define snd_dma_pci_data(pci) (&(pci)->dev) #define snd_dma_isa_data() NULL -#define snd_dma_sbus_data(sbus) ((struct device *)(sbus)) #define snd_dma_continuous_data(x) ((struct device *)(unsigned long)(x)) #endif @@ -49,7 +48,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ -#define SNDRV_DMA_TYPE_SBUS 4 /* SBUS continuous */ /* * info for buffer allocation diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h new file mode 100644 index 0000000..33df4cb --- /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 0000000..24247f7 --- /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 c1b26fc..7ee2f70 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, @@ -240,6 +238,7 @@ int snd_soc_dapm_sys_add(struct device *dev); /* dapm audio pin control and status */ int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin); int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin); +int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin); int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin); int snd_soc_dapm_sync(struct snd_soc_codec *codec); diff --git a/include/sound/soc.h b/include/sound/soc.h index a1e0357..f86e455 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -21,14 +21,13 @@ #include <sound/control.h> #include <sound/ac97_codec.h> -#define SND_SOC_VERSION "0.13.2" - /* * Convenience kcontrol builders */ #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ - {.reg = xreg, .shift = xshift, .max = xmax, .invert = xinvert}) + {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \ + .invert = xinvert}) #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .max = xmax, .invert = xinvert}) @@ -144,105 +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, @@ -262,27 +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 */ @@ -340,66 +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 *, @@ -425,6 +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; @@ -434,6 +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 */ @@ -447,13 +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 *, @@ -483,9 +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); @@ -498,23 +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; }; @@ -541,4 +406,6 @@ struct soc_enum { void *dapm; }; +#include <sound/soc-dai.h> + #endif diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h index b62ce3e..b6870cb 100644 --- a/include/sound/tea575x-tuner.h +++ b/include/sound/tea575x-tuner.h @@ -43,6 +43,7 @@ struct snd_tea575x { unsigned int freq_fixup; /* crystal onboard */ unsigned int val; /* hw value */ unsigned long freq; /* frequency */ + unsigned long in_use; /* set if the device is in use */ struct snd_tea575x_ops *ops; void *private_data; }; diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h new file mode 100644 index 0000000..475ef8b --- /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 4aafeda..2b48237 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 "" |
