|  | /* SPDX-License-Identifier: GPL-2.0 | 
|  | * | 
|  | * linux/sound/soc-dai.h -- ALSA SoC Layer | 
|  | * | 
|  | * Copyright:	2005-2008 Wolfson Microelectronics. PLC. | 
|  | * | 
|  | * Digital Audio Interface (DAI) API. | 
|  | */ | 
|  |  | 
|  | #ifndef __LINUX_SND_SOC_DAI_H | 
|  | #define __LINUX_SND_SOC_DAI_H | 
|  |  | 
|  |  | 
|  | #include <linux/list.h> | 
|  | #include <sound/asoc.h> | 
|  |  | 
|  | struct snd_pcm_substream; | 
|  | struct snd_soc_dapm_widget; | 
|  | struct snd_compr_stream; | 
|  |  | 
|  | /* | 
|  | * DAI hardware audio formats. | 
|  | * | 
|  | * Describes the physical PCM data formating and clocking. Add new formats | 
|  | * to the end. | 
|  | */ | 
|  | #define SND_SOC_DAIFMT_I2S		SND_SOC_DAI_FORMAT_I2S | 
|  | #define SND_SOC_DAIFMT_RIGHT_J		SND_SOC_DAI_FORMAT_RIGHT_J | 
|  | #define SND_SOC_DAIFMT_LEFT_J		SND_SOC_DAI_FORMAT_LEFT_J | 
|  | #define SND_SOC_DAIFMT_DSP_A		SND_SOC_DAI_FORMAT_DSP_A | 
|  | #define SND_SOC_DAIFMT_DSP_B		SND_SOC_DAI_FORMAT_DSP_B | 
|  | #define SND_SOC_DAIFMT_AC97		SND_SOC_DAI_FORMAT_AC97 | 
|  | #define SND_SOC_DAIFMT_PDM		SND_SOC_DAI_FORMAT_PDM | 
|  |  | 
|  | /* 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		(0 << 4) /* clock is gated */ | 
|  |  | 
|  | /* | 
|  | * DAI hardware signal polarity. | 
|  | * | 
|  | * Specifies whether the DAI can also support inverted clocks for the specified | 
|  | * format. | 
|  | * | 
|  | * BCLK: | 
|  | * - "normal" polarity means signal is available at rising edge of BCLK | 
|  | * - "inverted" polarity means signal is available at falling edge of BCLK | 
|  | * | 
|  | * FSYNC "normal" polarity depends on the frame format: | 
|  | * - I2S: frame consists of left then right channel data. Left channel starts | 
|  | *      with falling FSYNC edge, right channel starts with rising FSYNC edge. | 
|  | * - Left/Right Justified: frame consists of left then right channel data. | 
|  | *      Left channel starts with rising FSYNC edge, right channel starts with | 
|  | *      falling FSYNC edge. | 
|  | * - DSP A/B: Frame starts with rising FSYNC edge. | 
|  | * - AC97: Frame starts with rising FSYNC edge. | 
|  | * | 
|  | * "Negative" FSYNC polarity is the one opposite of "normal" polarity. | 
|  | */ | 
|  | #define SND_SOC_DAIFMT_NB_NF		(0 << 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_S20_LE |\ | 
|  | SNDRV_PCM_FMTBIT_S20_BE |\ | 
|  | 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 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); | 
|  |  | 
|  | int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); | 
|  |  | 
|  | /* 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, | 
|  | int direction); | 
|  |  | 
|  |  | 
|  | int snd_soc_dai_get_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_is_dummy(struct snd_soc_dai *dai); | 
|  |  | 
|  | 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); | 
|  | int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); | 
|  |  | 
|  | /* | 
|  | * 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 (*xlate_tdm_slot_mask)(unsigned int slots, | 
|  | unsigned int *tx_mask, unsigned int *rx_mask); | 
|  | 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 (*get_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); | 
|  |  | 
|  | int (*set_sdw_stream)(struct snd_soc_dai *dai, | 
|  | void *stream, int direction); | 
|  | /* | 
|  | * DAI digital mute - optional. | 
|  | * Called by soc-core to minimise any pops. | 
|  | */ | 
|  | int (*digital_mute)(struct snd_soc_dai *dai, int mute); | 
|  | int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream); | 
|  |  | 
|  | /* | 
|  | * 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 *); | 
|  | /* | 
|  | * NOTE: Commands passed to the trigger function are not necessarily | 
|  | * compatible with the current state of the dai. For example this | 
|  | * sequence of commands is possible: START STOP STOP. | 
|  | * So do not unconditionally use refcounting functions in the trigger | 
|  | * function, e.g. clk_enable/disable. | 
|  | */ | 
|  | int (*trigger)(struct snd_pcm_substream *, int, | 
|  | struct snd_soc_dai *); | 
|  | int (*bespoke_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 *); | 
|  | }; | 
|  |  | 
|  | struct snd_soc_cdai_ops { | 
|  | /* | 
|  | * for compress ops | 
|  | */ | 
|  | int (*startup)(struct snd_compr_stream *, | 
|  | struct snd_soc_dai *); | 
|  | int (*shutdown)(struct snd_compr_stream *, | 
|  | struct snd_soc_dai *); | 
|  | int (*set_params)(struct snd_compr_stream *, | 
|  | struct snd_compr_params *, struct snd_soc_dai *); | 
|  | int (*get_params)(struct snd_compr_stream *, | 
|  | struct snd_codec *, struct snd_soc_dai *); | 
|  | int (*set_metadata)(struct snd_compr_stream *, | 
|  | struct snd_compr_metadata *, struct snd_soc_dai *); | 
|  | int (*get_metadata)(struct snd_compr_stream *, | 
|  | struct snd_compr_metadata *, struct snd_soc_dai *); | 
|  | int (*trigger)(struct snd_compr_stream *, int, | 
|  | struct snd_soc_dai *); | 
|  | int (*pointer)(struct snd_compr_stream *, | 
|  | struct snd_compr_tstamp *, struct snd_soc_dai *); | 
|  | int (*ack)(struct snd_compr_stream *, size_t, | 
|  | 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; | 
|  | unsigned int base; | 
|  | struct snd_soc_dobj dobj; | 
|  |  | 
|  | /* 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); | 
|  | /* compress dai */ | 
|  | int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); | 
|  | /* Optional Callback used at pcm creation*/ | 
|  | int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, | 
|  | struct snd_soc_dai *dai); | 
|  | /* DAI is also used for the control bus */ | 
|  | bool bus_control; | 
|  |  | 
|  | /* ops */ | 
|  | const struct snd_soc_dai_ops *ops; | 
|  | const struct snd_soc_cdai_ops *cops; | 
|  |  | 
|  | /* DAI capabilities */ | 
|  | struct snd_soc_pcm_stream capture; | 
|  | struct snd_soc_pcm_stream playback; | 
|  | unsigned int symmetric_rates:1; | 
|  | unsigned int symmetric_channels:1; | 
|  | unsigned int symmetric_samplebits: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; | 
|  |  | 
|  | /* driver ops */ | 
|  | struct snd_soc_dai_driver *driver; | 
|  |  | 
|  | /* DAI runtime info */ | 
|  | unsigned int capture_active;		/* stream usage count */ | 
|  | unsigned int playback_active;		/* stream usage count */ | 
|  | unsigned int probed:1; | 
|  |  | 
|  | unsigned int active; | 
|  |  | 
|  | struct snd_soc_dapm_widget *playback_widget; | 
|  | struct snd_soc_dapm_widget *capture_widget; | 
|  |  | 
|  | /* DAI DMA data */ | 
|  | void *playback_dma_data; | 
|  | void *capture_dma_data; | 
|  |  | 
|  | /* Symmetry data - only valid if symmetry is being enforced */ | 
|  | unsigned int rate; | 
|  | unsigned int channels; | 
|  | unsigned int sample_bits; | 
|  |  | 
|  | /* parent platform/codec */ | 
|  | struct snd_soc_component *component; | 
|  |  | 
|  | /* CODEC TDM slot masks and params (for fixup) */ | 
|  | unsigned int tx_mask; | 
|  | unsigned int rx_mask; | 
|  |  | 
|  | 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; | 
|  | } | 
|  |  | 
|  | static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, | 
|  | void *playback, void *capture) | 
|  | { | 
|  | dai->playback_dma_data = playback; | 
|  | dai->capture_dma_data = capture; | 
|  | } | 
|  |  | 
|  | 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); | 
|  | } | 
|  |  | 
|  | /** | 
|  | * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation | 
|  | * @dai: DAI | 
|  | * @stream: STREAM | 
|  | * @direction: Stream direction(Playback/Capture) | 
|  | * SoundWire subsystem doesn't have a notion of direction and we reuse | 
|  | * the ASoC stream direction to configure sink/source ports. | 
|  | * Playback maps to source ports and Capture for sink ports. | 
|  | * | 
|  | * This should be invoked with NULL to clear the stream set previously. | 
|  | * Returns 0 on success, a negative error code otherwise. | 
|  | */ | 
|  | static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai, | 
|  | void *stream, int direction) | 
|  | { | 
|  | if (dai->driver->ops->set_sdw_stream) | 
|  | return dai->driver->ops->set_sdw_stream(dai, stream, direction); | 
|  | else | 
|  | return -ENOTSUPP; | 
|  | } | 
|  |  | 
|  | #endif |