/* * 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 1 /* I2S mode */ #define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ #define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ #define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ #define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ #define SND_SOC_DAIFMT_AC97 6 /* AC97 */ #define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */ /* 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 the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ #define SND_SOC_DAIFMT_GATED (2 << 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 (1 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ #define SND_SOC_DAIFMT_IB_IF (4 << 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 (1 << 12) /* codec clk & FRM master */ #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ #define SND_SOC_DAIFMT_CBS_CFS (4 << 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_S8 |\ SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) struct snd_soc_dai_driver; struct snd_soc_dai; struct snd_ac97_bus_ops; /* Digital Audio Interface registration */ 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, 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, int source, 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 tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); 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); 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, int source, 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 tx_mask, unsigned int rx_mask, int slots, int slot_width); int (*set_channel_map)(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot); 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 *); /* * For hardware based FIFO caused delay reporting. * Optional. */ snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, struct snd_soc_dai *); }; /* * Digital Audio Interface Driver. * * 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_driver { /* DAI description */ const char *name; unsigned int id; int ac97_control; /* 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); /* ops */ const 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; /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; }; /* * 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 */ 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 DMA data */ void *playback_dma_data; void *capture_dma_data; /* Symmetry data - only valid if symmetry is being enforced */ unsigned int rate; /* 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; } static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, const struct snd_pcm_substream *ss, void *data) { if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) dai->playback_dma_data = data; else 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