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Diffstat (limited to 'include/sound/soc-dai.h')
| -rw-r--r-- | include/sound/soc-dai.h | 256 |
1 files changed, 256 insertions, 0 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h new file mode 100644 index 00000000..8ae6bfee --- /dev/null +++ b/include/sound/soc-dai.h @@ -0,0 +1,256 @@ +/* + * 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> + +#include <sound/soc.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 */ +#define SND_SOC_DAIFMT_PDM 6 /* 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 (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_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_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, 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); + +/* + * Digital Audio Interface. + * + * 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_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 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; + void *ac97_pdata; /* platform_data for the ac97 codec */ + + /* 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_soc_codec *codec; + unsigned int active; + unsigned char pop_wait:1; + + unsigned int use_idma; + + /* DAI private data */ + void *private_data; + + /* parent platform */ + struct snd_soc_platform *platform; + + struct list_head 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; +} + +#endif |