| Author | Tokens | Token Proportion | Commits | Commit Proportion |
|---|---|---|---|---|
| Kuninori Morimoto | 675 | 29.09% | 51 | 45.54% |
| Mark Brown | 614 | 26.47% | 13 | 11.61% |
| Liam Girdwood | 181 | 7.80% | 10 | 8.93% |
| Richard Purdie | 132 | 5.69% | 1 | 0.89% |
| Shreyas NC | 111 | 4.78% | 1 | 0.89% |
| Vinod Koul | 97 | 4.18% | 2 | 1.79% |
| Barry Song | 78 | 3.36% | 1 | 0.89% |
| Pierre-Louis Bossart | 60 | 2.59% | 4 | 3.57% |
| Namarta Kohli | 51 | 2.20% | 1 | 0.89% |
| Peter Ujfalusi | 46 | 1.98% | 1 | 0.89% |
| Graeme Gregory | 46 | 1.98% | 1 | 0.89% |
| Frank Mandarino | 28 | 1.21% | 1 | 0.89% |
| Srinivas Kandagatla | 25 | 1.08% | 3 | 2.68% |
| Daniel Mack | 24 | 1.03% | 1 | 0.89% |
| Xiubo Li | 21 | 0.91% | 2 | 1.79% |
| Charles Keepax | 17 | 0.73% | 1 | 0.89% |
| Daniel Ribeiro | 16 | 0.69% | 1 | 0.89% |
| Nicolin Chen | 14 | 0.60% | 1 | 0.89% |
| Jie Yang | 13 | 0.56% | 1 | 0.89% |
| Arnaud Pouliquen | 11 | 0.47% | 1 | 0.89% |
| Daniel Glöckner | 11 | 0.47% | 1 | 0.89% |
| Mengdong Lin | 10 | 0.43% | 1 | 0.89% |
| Misael Lopez Cruz | 8 | 0.34% | 2 | 1.79% |
| Jeeja KP | 8 | 0.34% | 1 | 0.89% |
| Ricard Wanderlöf | 6 | 0.26% | 1 | 0.89% |
| Lars-Peter Clausen | 5 | 0.22% | 1 | 0.89% |
| Manuel Lauss | 4 | 0.17% | 1 | 0.89% |
| Peter Meerwald-Stadler | 3 | 0.13% | 1 | 0.89% |
| Markus Pargmann | 1 | 0.04% | 1 | 0.89% |
| Eric Miao | 1 | 0.04% | 1 | 0.89% |
| Randy Dunlap | 1 | 0.04% | 1 | 0.89% |
| Maciej S. Szmigiero | 1 | 0.04% | 1 | 0.89% |
| Anatol Pomozov | 1 | 0.04% | 1 | 0.89% |
| Total | 2320 | 112 |
/* 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 /* 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 0 #define SND_SOC_POSSIBLE_DAIFMT_FORMAT_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_I2S (1 << SND_SOC_DAI_FORMAT_I2S) #define SND_SOC_POSSIBLE_DAIFMT_RIGHT_J (1 << SND_SOC_DAI_FORMAT_RIGHT_J) #define SND_SOC_POSSIBLE_DAIFMT_LEFT_J (1 << SND_SOC_DAI_FORMAT_LEFT_J) #define SND_SOC_POSSIBLE_DAIFMT_DSP_A (1 << SND_SOC_DAI_FORMAT_DSP_A) #define SND_SOC_POSSIBLE_DAIFMT_DSP_B (1 << SND_SOC_DAI_FORMAT_DSP_B) #define SND_SOC_POSSIBLE_DAIFMT_AC97 (1 << SND_SOC_DAI_FORMAT_AC97) #define SND_SOC_POSSIBLE_DAIFMT_PDM (1 << SND_SOC_DAI_FORMAT_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 (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 */ /* Describes the possible PCM format */ #define SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT 32 #define SND_SOC_POSSIBLE_DAIFMT_INV_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_NB_NF (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_NB_IF (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_IB_NF (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_IB_IF (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) /* * 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 */ /* previous definitions kept for backwards-compatibility, do not use in new contributions */ #define SND_SOC_DAIFMT_CBM_CFM SND_SOC_DAIFMT_CBP_CFP #define SND_SOC_DAIFMT_CBS_CFM SND_SOC_DAIFMT_CBC_CFP #define SND_SOC_DAIFMT_CBM_CFS SND_SOC_DAIFMT_CBP_CFC #define SND_SOC_DAIFMT_CBS_CFS SND_SOC_DAIFMT_CBC_CFC /* 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 /* Describes the possible PCM format */ #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT 48 #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_CBP_CFP (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_CBC_CFP (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_CBP_CFC (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) #define SND_SOC_POSSIBLE_DAIFMT_CBC_CFC (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) #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_CLOCK_PROVIDER_MASK 0xf000 #define SND_SOC_DAIFMT_MASTER_MASK SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK /* * 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_get_fmt_max_priority(struct snd_soc_pcm_runtime *rtd); u64 snd_soc_dai_get_fmt(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, 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); 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(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) { snd_soc_dai_action(dai, stream, 1); } static inline void snd_soc_dai_deactivate(struct snd_soc_dai *dai, int stream) { snd_soc_dai_action(dai, stream, -1); } int snd_soc_dai_active(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(struct snd_soc_dai *dai); 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, int num); /* 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, 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_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 *); 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 *); /* * Format list for auto selection. * Format will be increased if priority format was * not selected. * see * snd_soc_dai_get_fmt() */ u64 *auto_selectable_formats; int num_auto_selectable_formats; /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; /* bit field */ unsigned int no_capture_mute:1; unsigned int 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 */ const char *name; unsigned int id; unsigned int base; struct snd_soc_dobj dobj; struct of_phandle_args *dai_args; /* 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_rate:1; unsigned int symmetric_channels:1; unsigned int symmetric_sample_bits:1; }; /* for Playback/Capture */ struct snd_soc_dai_stream { struct snd_soc_dapm_widget *widget; unsigned int active; /* usage count */ unsigned int tdm_mask; /* CODEC TDM slot masks and params (for fixup) */ void *dma_data; /* DAI DMA data */ }; /* * 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 */ struct snd_soc_dai_stream stream[SNDRV_PCM_STREAM_LAST + 1]; /* 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; struct list_head list; /* function mark */ struct snd_pcm_substream *mark_startup; struct snd_pcm_substream *mark_hw_params; struct snd_pcm_substream *mark_trigger; struct snd_compr_stream *mark_compr_startup; /* bit field */ unsigned int probed:1; }; static inline struct snd_soc_pcm_stream * snd_soc_dai_get_pcm_stream(const struct snd_soc_dai *dai, int stream) { return (stream == SNDRV_PCM_STREAM_PLAYBACK) ? &dai->driver->playback : &dai->driver->capture; } #define snd_soc_dai_get_widget_playback(dai) snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_PLAYBACK) #define snd_soc_dai_get_widget_capture(dai) snd_soc_dai_get_widget(dai, SNDRV_PCM_STREAM_CAPTURE) static inline struct snd_soc_dapm_widget *snd_soc_dai_get_widget(struct snd_soc_dai *dai, int stream) { return dai->stream[stream].widget; } #define snd_soc_dai_set_widget_playback(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_PLAYBACK, widget) #define snd_soc_dai_set_widget_capture(dai, widget) snd_soc_dai_set_widget(dai, SNDRV_PCM_STREAM_CAPTURE, widget) static inline void snd_soc_dai_set_widget(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_widget *widget) { dai->stream[stream].widget = widget; } #define snd_soc_dai_dma_data_get_playback(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_PLAYBACK) #define snd_soc_dai_dma_data_get_capture(dai) snd_soc_dai_dma_data_get(dai, SNDRV_PCM_STREAM_CAPTURE) #define snd_soc_dai_get_dma_data(dai, ss) snd_soc_dai_dma_data_get(dai, ss->stream) static inline void *snd_soc_dai_dma_data_get(const struct snd_soc_dai *dai, int stream) { return dai->stream[stream].dma_data; } #define snd_soc_dai_dma_data_set_playback(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_PLAYBACK, data) #define snd_soc_dai_dma_data_set_capture(dai, data) snd_soc_dai_dma_data_set(dai, SNDRV_PCM_STREAM_CAPTURE, data) #define snd_soc_dai_set_dma_data(dai, ss, data) snd_soc_dai_dma_data_set(dai, ss->stream, data) static inline void snd_soc_dai_dma_data_set(struct snd_soc_dai *dai, int stream, void *data) { dai->stream[stream].dma_data = data; } static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, void *playback, void *capture) { snd_soc_dai_dma_data_set_playback(dai, playback); snd_soc_dai_dma_data_set_capture(dai, capture); } static inline unsigned int snd_soc_dai_tdm_mask_get(struct snd_soc_dai *dai, int stream) { return dai->stream[stream].tdm_mask; } static inline void snd_soc_dai_tdm_mask_set(struct snd_soc_dai *dai, int stream, unsigned int tdm_mask) { dai->stream[stream].tdm_mask = tdm_mask; } static inline unsigned int snd_soc_dai_stream_active(struct snd_soc_dai *dai, int stream) { /* see snd_soc_dai_action() for setup */ return dai->stream[stream].active; } 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_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) { 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; */ static inline void *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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