/* 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 … #define SND_SOC_DAIFMT_RIGHT_J … #define SND_SOC_DAIFMT_LEFT_J … #define SND_SOC_DAIFMT_DSP_A … #define SND_SOC_DAIFMT_DSP_B … #define SND_SOC_DAIFMT_AC97 … #define SND_SOC_DAIFMT_PDM … /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB … #define SND_SOC_DAIFMT_LSB … /* Describes the possible PCM format */ /* * use SND_SOC_DAI_FORMAT_xx as eash shift. * see * snd_soc_runtime_get_dai_fmt() */ #define SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT … #define SND_SOC_POSSIBLE_DAIFMT_FORMAT_MASK … #define SND_SOC_POSSIBLE_DAIFMT_I2S … #define SND_SOC_POSSIBLE_DAIFMT_RIGHT_J … #define SND_SOC_POSSIBLE_DAIFMT_LEFT_J … #define SND_SOC_POSSIBLE_DAIFMT_DSP_A … #define SND_SOC_POSSIBLE_DAIFMT_DSP_B … #define SND_SOC_POSSIBLE_DAIFMT_AC97 … #define SND_SOC_POSSIBLE_DAIFMT_PDM … /* * DAI Clock gating. * * DAI bit clocks can 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 … #define SND_SOC_DAIFMT_GATED … /* Describes the possible PCM format */ /* * define GATED -> CONT. GATED will be selected if both are selected. * see * snd_soc_runtime_get_dai_fmt() */ #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT … #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_MASK … #define SND_SOC_POSSIBLE_DAIFMT_GATED … #define SND_SOC_POSSIBLE_DAIFMT_CONT … /* * 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 … #define SND_SOC_DAIFMT_NB_IF … #define SND_SOC_DAIFMT_IB_NF … #define SND_SOC_DAIFMT_IB_IF … /* Describes the possible PCM format */ #define SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT … #define SND_SOC_POSSIBLE_DAIFMT_INV_MASK … #define SND_SOC_POSSIBLE_DAIFMT_NB_NF … #define SND_SOC_POSSIBLE_DAIFMT_NB_IF … #define SND_SOC_POSSIBLE_DAIFMT_IB_NF … #define SND_SOC_POSSIBLE_DAIFMT_IB_IF … /* * DAI hardware clock providers/consumers * * This is wrt the codec, the inverse is true for the interface * i.e. if the codec is clk and FRM provider then the interface is * clk and frame consumer. */ #define SND_SOC_DAIFMT_CBP_CFP … #define SND_SOC_DAIFMT_CBC_CFP … #define SND_SOC_DAIFMT_CBP_CFC … #define SND_SOC_DAIFMT_CBC_CFC … /* previous definitions kept for backwards-compatibility, do not use in new contributions */ #define SND_SOC_DAIFMT_CBM_CFM … #define SND_SOC_DAIFMT_CBS_CFM … #define SND_SOC_DAIFMT_CBM_CFS … #define SND_SOC_DAIFMT_CBS_CFS … /* when passed to set_fmt directly indicate if the device is provider or consumer */ #define SND_SOC_DAIFMT_BP_FP … #define SND_SOC_DAIFMT_BC_FP … #define SND_SOC_DAIFMT_BP_FC … #define SND_SOC_DAIFMT_BC_FC … /* Describes the possible PCM format */ #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT … #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_MASK … #define SND_SOC_POSSIBLE_DAIFMT_CBP_CFP … #define SND_SOC_POSSIBLE_DAIFMT_CBC_CFP … #define SND_SOC_POSSIBLE_DAIFMT_CBP_CFC … #define SND_SOC_POSSIBLE_DAIFMT_CBC_CFC … #define SND_SOC_DAIFMT_FORMAT_MASK … #define SND_SOC_DAIFMT_CLOCK_MASK … #define SND_SOC_DAIFMT_INV_MASK … #define SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK … #define SND_SOC_DAIFMT_MASTER_MASK … /* * Master Clock Directions */ #define SND_SOC_CLOCK_IN … #define SND_SOC_CLOCK_OUT … #define SND_SOC_STD_AC97_FMTS … 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_get_fmt_max_priority(const struct snd_soc_pcm_runtime *rtd); u64 snd_soc_dai_get_fmt(const struct snd_soc_dai *dai, int priority); 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, const unsigned int *tx_slot, unsigned int rx_num, const 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(const 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(const struct snd_soc_dai *dai); int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params); void snd_soc_dai_hw_free(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int rollback); int snd_soc_dai_startup(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int rollback); void snd_soc_dai_suspend(struct snd_soc_dai *dai); void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream); void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); static inline void snd_soc_dai_activate(struct snd_soc_dai *dai, int stream) { … } static inline void snd_soc_dai_deactivate(struct snd_soc_dai *dai, int stream) { … } int snd_soc_dai_active(const struct snd_soc_dai *dai); int snd_soc_pcm_dai_probe(struct snd_soc_pcm_runtime *rtd, int order); int snd_soc_pcm_dai_remove(struct snd_soc_pcm_runtime *rtd, int order); int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); int snd_soc_dai_compr_startup(struct snd_soc_dai *dai, struct snd_compr_stream *cstream); void snd_soc_dai_compr_shutdown(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, int rollback); int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, int cmd); int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_params *params); int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_codec *params); int snd_soc_dai_compr_ack(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, size_t bytes); int snd_soc_dai_compr_pointer(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_tstamp *tstamp); int snd_soc_dai_compr_set_metadata(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata); int snd_soc_dai_compr_get_metadata(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_metadata *metadata); const char *snd_soc_dai_name_get(const struct snd_soc_dai *dai); struct snd_soc_dai_ops { … }; struct snd_soc_cdai_ops { … }; /* * 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 { … }; /* for Playback/Capture */ struct snd_soc_dai_stream { … }; /* * Digital Audio Interface runtime data. * * Holds runtime data for a DAI. */ struct snd_soc_dai { … }; static inline const struct snd_soc_pcm_stream * snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) { … } #define snd_soc_dai_get_widget_playback(dai) … #define snd_soc_dai_get_widget_capture(dai) … static inline struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream) { … } #define snd_soc_dai_set_widget_playback(dai, widget) … #define snd_soc_dai_set_widget_capture(dai, widget) … static inline void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget) { … } #define snd_soc_dai_dma_data_get_playback(dai) … #define snd_soc_dai_dma_data_get_capture(dai) … #define snd_soc_dai_get_dma_data(dai, ss) … static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int stream) { … } #define snd_soc_dai_dma_data_set_playback(dai, data) … #define snd_soc_dai_dma_data_set_capture(dai, data) … #define snd_soc_dai_set_dma_data(dai, ss, data) … static inline void snd_soc_dai_dma_data_set(struct snd_soc_dai *dai, int stream, void *data) { … } static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *playback, void *capture) { … } static inline unsigned int snd_soc_dai_tdm_mask_get(const struct snd_soc_dai *dai, int stream) { … } static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream, unsigned int tdm_mask) { … } static inline unsigned int snd_soc_dai_stream_active(const struct snd_soc_dai *dai, int stream) { … } static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) { … } static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) { … } /** * snd_soc_dai_set_stream() - Configures a DAI for stream operation * @dai: DAI * @stream: STREAM (opaque structure depending on DAI type) * @direction: Stream direction(Playback/Capture) * Some subsystems, such as SoundWire, don'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_stream(struct snd_soc_dai *dai, void *stream, int direction) { … } /** * snd_soc_dai_get_stream() - Retrieves stream from DAI * @dai: DAI * @direction: Stream direction(Playback/Capture) * * This routine only retrieves that was previously configured * with snd_soc_dai_get_stream() * * Returns pointer to stream or an ERR_PTR value, e.g. * ERR_PTR(-ENOTSUPP) if callback is not supported; */ static inline void *snd_soc_dai_get_stream(struct snd_soc_dai *dai, int direction) { … } #endif
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