| /* |
| * 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 */ |
| #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_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 */ |
| 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; |
| }; |
| |
| /* |
| * 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; |
| |
| /* 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 |