diff options
Diffstat (limited to 'include/sound')
-rw-r--r-- | include/sound/soc-dai.h | 209 | ||||
-rw-r--r-- | include/sound/soc.h | 148 |
2 files changed, 211 insertions, 146 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h new file mode 100644 index 00000000000..08b8f7025c6 --- /dev/null +++ b/include/sound/soc-dai.h @@ -0,0 +1,209 @@ +/* + * 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 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); +}; + +/* + * Digital Audio Interface runtime data. + * + * Holds runtime data for a DAI. + */ +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; + + /* 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.h b/include/sound/soc.h index 3be17b3c650..e4465f73aa4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -151,76 +151,6 @@ enum snd_soc_bias_level { #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; @@ -260,27 +190,6 @@ 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 */ @@ -338,61 +247,6 @@ 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; @@ -543,4 +397,6 @@ struct soc_enum { void *dapm; }; +#include <sound/soc-dai.h> + #endif |