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yaml --- r: 120235 b: refs/heads/master c: a47cbe7 h: refs/heads/master i: 120233: 316f841 120231: b18a5cd v: v3
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| /* | ||
| * 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. | ||
| */ | ||
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| #ifndef __LINUX_SND_SOC_DAI_H | ||
| #define __LINUX_SND_SOC_DAI_H | ||
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| #include <linux/list.h> | ||
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| struct snd_pcm_substream; | ||
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| /* | ||
| * 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 */ | ||
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| /* 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 | ||
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| /* | ||
| * 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 */ | ||
|
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| /* | ||
| * 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 */ | ||
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| /* | ||
| * TDM | ||
| * | ||
| * Time Division Multiplexing. Allows PCM data to be multplexed with other | ||
| * data on the DAI. | ||
| */ | ||
| #define SND_SOC_DAIFMT_TDM (1 << 6) | ||
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| /* | ||
| * 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 */ | ||
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| /* | ||
| * 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 */ | ||
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| #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 | ||
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| /* | ||
| * Master Clock Directions | ||
| */ | ||
| #define SND_SOC_CLOCK_IN 0 | ||
| #define SND_SOC_CLOCK_OUT 1 | ||
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| struct snd_soc_dai_ops; | ||
| struct snd_soc_dai; | ||
| struct snd_ac97_bus_ops; | ||
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| /* Digital Audio Interface clocking API.*/ | ||
| int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, | ||
| unsigned int freq, int dir); | ||
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| int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, | ||
| int div_id, int div); | ||
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| int snd_soc_dai_set_pll(struct snd_soc_dai *dai, | ||
| int pll_id, unsigned int freq_in, unsigned int freq_out); | ||
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| /* Digital Audio interface formatting */ | ||
| int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); | ||
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| int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, | ||
| unsigned int mask, int slots); | ||
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| int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); | ||
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| /* Digital Audio Interface mute */ | ||
| int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); | ||
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| /* | ||
| * 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); | ||
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| /* | ||
| * 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); | ||
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| /* | ||
| * DAI digital mute - optional. | ||
| * Called by soc-core to minimise any pops. | ||
| */ | ||
| int (*digital_mute)(struct snd_soc_dai *dai, int mute); | ||
| }; | ||
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| /* | ||
| * 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; | ||
|
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| /* 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); | ||
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| /* ops */ | ||
| struct snd_soc_ops ops; | ||
| struct snd_soc_dai_ops dai_ops; | ||
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| /* DAI capabilities */ | ||
| struct snd_soc_pcm_stream capture; | ||
| struct snd_soc_pcm_stream playback; | ||
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| /* DAI runtime info */ | ||
| struct snd_pcm_runtime *runtime; | ||
| struct snd_soc_codec *codec; | ||
| unsigned int active; | ||
| unsigned char pop_wait:1; | ||
| void *dma_data; | ||
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| /* DAI private data */ | ||
| void *private_data; | ||
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| /* parent codec/platform */ | ||
| union { | ||
| struct snd_soc_codec *codec; | ||
| struct snd_soc_platform *platform; | ||
| }; | ||
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| struct list_head list; | ||
| }; | ||
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| #endif |
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