/* * 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 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 */
/* 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 16 #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_GATED (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_CONT (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT)
/* * 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 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 (1 << 12) /* codec clk provider & frame provider */ #define SND_SOC_DAIFMT_CBC_CFP (2 << 12) /* codec clk consumer & frame provider */ #define SND_SOC_DAIFMT_CBP_CFC (3 << 12) /* codec clk provider & frame consumer */ #define SND_SOC_DAIFMT_CBC_CFC (4 << 12) /* codec clk consumer & frame consumer */
/* when passed to set_fmt directly indicate if the device is provider or consumer */ #define SND_SOC_DAIFMT_BP_FP SND_SOC_DAIFMT_CBP_CFP #define SND_SOC_DAIFMT_BC_FP SND_SOC_DAIFMT_CBC_CFP #define SND_SOC_DAIFMT_BP_FC SND_SOC_DAIFMT_CBP_CFC #define SND_SOC_DAIFMT_BC_FC SND_SOC_DAIFMT_CBC_CFC
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
int snd_soc_dai_prepare(struct snd_soc_dai *dai, struct snd_pcm_substream *substream);
/* Digital Audio Interface mute */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); int snd_soc_dai_mute_is_ctrled_at_trigger(struct snd_soc_dai *dai);
int snd_soc_dai_is_dummy(conststruct 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); bool snd_soc_dai_stream_valid(conststruct snd_soc_dai *dai, int stream); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); staticinlinevoid snd_soc_dai_activate(struct snd_soc_dai *dai, int stream)
{
snd_soc_dai_action(dai, stream, 1);
} staticinlinevoid snd_soc_dai_deactivate(struct snd_soc_dai *dai, int stream)
{
snd_soc_dai_action(dai, stream, -1);
} int snd_soc_dai_active(conststruct 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); 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);
struct snd_soc_dai_ops { /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); int (*remove)(struct snd_soc_dai *dai); /* compress dai */ int (*compress_new)(struct snd_soc_pcm_runtime *rtd); /* Optional Callback used at pcm creation*/ int (*pcm_new)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
/* * 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, unsignedint freq, int dir); int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsignedint freq_in, unsignedint freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsignedint ratio);
/* * DAI format configuration * Called by soc_card drivers, normally in their hw_params.
*/ int (*set_fmt)(struct snd_soc_dai *dai, unsignedint fmt); int (*xlate_tdm_slot_mask)(unsignedint slots, unsignedint *tx_mask, unsignedint *rx_mask); int (*set_tdm_slot)(struct snd_soc_dai *dai, unsignedint tx_mask, unsignedint rx_mask, int slots, int slot_width); int (*set_channel_map)(struct snd_soc_dai *dai, unsignedint tx_num, constunsignedint *tx_slot, unsignedint rx_num, constunsignedint *rx_slot); int (*get_channel_map)(conststruct snd_soc_dai *dai, unsignedint *tx_num, unsignedint *tx_slot, unsignedint *rx_num, unsignedint *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
int (*set_stream)(struct snd_soc_dai *dai, void *stream, int direction); void *(*get_stream)(struct snd_soc_dai *dai, int direction);
/* * DAI digital mute - optional. * Called by soc-core to minimise any pops.
*/ 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 *);
/* * For hardware based FIFO caused delay reporting. * Optional.
*/
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *, struct snd_soc_dai *);
/* * Format list for auto selection. * Format will be increased if priority format was * not selected. * see * snd_soc_dai_get_fmt()
*/ const u64 *auto_selectable_formats; int num_auto_selectable_formats;
/* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order;
/* bit field */ unsignedint no_capture_mute:1; unsignedint mute_unmute_on_trigger:1;
};
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 */ constchar *name; unsignedint id; unsignedint base; struct snd_soc_dobj dobj; conststruct of_phandle_args *dai_args;
/* DAI runtime info */ struct snd_soc_dai_stream stream[SNDRV_PCM_STREAM_LAST + 1];
/* Symmetry data - only valid if symmetry is being enforced */ unsignedint symmetric_rate; unsignedint symmetric_channels; unsignedint symmetric_sample_bits;
/** * 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.
*/ staticinlineint snd_soc_dai_set_stream(struct snd_soc_dai *dai, void *stream, int direction)
{ if (dai->driver->ops->set_stream) return dai->driver->ops->set_stream(dai, stream, direction); else return -ENOTSUPP;
}
/** * 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;
*/ staticinlinevoid *snd_soc_dai_get_stream(struct snd_soc_dai *dai, int direction)
{ if (dai->driver->ops->get_stream) return dai->driver->ops->get_stream(dai, direction); else return ERR_PTR(-ENOTSUPP);
}
#endif
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