/* * 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 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 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 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S32_LE) 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; unsigned int symmetric_rates:1; /* 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