LXR linux/include/sound/soc-dai.h

   1/* SPDX-License-Identifier: GPL-2.0
   2 *
   3 * linux/sound/soc-dai.h -- ALSA SoC Layer
   4 *
   5 * Copyright:   2005-2008 Wolfson Microelectronics. PLC.
   6 *
   7 * Digital Audio Interface (DAI) API.
   8 */
   9
  10#ifndef __LINUX_SND_SOC_DAI_H
  11#define __LINUX_SND_SOC_DAI_H
  12
  13
  14#include <linux/list.h>
  15#include <sound/asoc.h>
  16
  17struct snd_pcm_substream;
  18struct snd_soc_dapm_widget;
  19struct snd_compr_stream;
  20
  21/*
  22 * DAI hardware audio formats.
  23 *
  24 * Describes the physical PCM data formating and clocking. Add new formats
  25 * to the end.
  26 */
  27#define SND_SOC_DAIFMT_I2S              SND_SOC_DAI_FORMAT_I2S
  28#define SND_SOC_DAIFMT_RIGHT_J          SND_SOC_DAI_FORMAT_RIGHT_J
  29#define SND_SOC_DAIFMT_LEFT_J           SND_SOC_DAI_FORMAT_LEFT_J
  30#define SND_SOC_DAIFMT_DSP_A            SND_SOC_DAI_FORMAT_DSP_A
  31#define SND_SOC_DAIFMT_DSP_B            SND_SOC_DAI_FORMAT_DSP_B
  32#define SND_SOC_DAIFMT_AC97             SND_SOC_DAI_FORMAT_AC97
  33#define SND_SOC_DAIFMT_PDM              SND_SOC_DAI_FORMAT_PDM
  34
  35/* left and right justified also known as MSB and LSB respectively */
  36#define SND_SOC_DAIFMT_MSB              SND_SOC_DAIFMT_LEFT_J
  37#define SND_SOC_DAIFMT_LSB              SND_SOC_DAIFMT_RIGHT_J
  38
  39/*
  40 * DAI Clock gating.
  41 *
  42 * DAI bit clocks can be be gated (disabled) when the DAI is not
  43 * sending or receiving PCM data in a frame. This can be used to save power.
  44 */
  45#define SND_SOC_DAIFMT_CONT             (1 << 4) /* continuous clock */
  46#define SND_SOC_DAIFMT_GATED            (0 << 4) /* clock is gated */
  47
  48/*
  49 * DAI hardware signal polarity.
  50 *
  51 * Specifies whether the DAI can also support inverted clocks for the specified
  52 * format.
  53 *
  54 * BCLK:
  55 * - "normal" polarity means signal is available at rising edge of BCLK
  56 * - "inverted" polarity means signal is available at falling edge of BCLK
  57 *
  58 * FSYNC "normal" polarity depends on the frame format:
  59 * - I2S: frame consists of left then right channel data. Left channel starts
  60 *      with falling FSYNC edge, right channel starts with rising FSYNC edge.
  61 * - Left/Right Justified: frame consists of left then right channel data.
  62 *      Left channel starts with rising FSYNC edge, right channel starts with
  63 *      falling FSYNC edge.
  64 * - DSP A/B: Frame starts with rising FSYNC edge.
  65 * - AC97: Frame starts with rising FSYNC edge.
  66 *
  67 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
  68 */
  69#define SND_SOC_DAIFMT_NB_NF            (0 << 8) /* normal bit clock + frame */
  70#define SND_SOC_DAIFMT_NB_IF            (2 << 8) /* normal BCLK + inv FRM */
  71#define SND_SOC_DAIFMT_IB_NF            (3 << 8) /* invert BCLK + nor FRM */
  72#define SND_SOC_DAIFMT_IB_IF            (4 << 8) /* invert BCLK + FRM */
  73
  74/*
  75 * DAI hardware clock masters.
  76 *
  77 * This is wrt the codec, the inverse is true for the interface
  78 * i.e. if the codec is clk and FRM master then the interface is
  79 * clk and frame slave.
  80 */
  81#define SND_SOC_DAIFMT_CBM_CFM          (1 << 12) /* codec clk & FRM master */
  82#define SND_SOC_DAIFMT_CBS_CFM          (2 << 12) /* codec clk slave & FRM master */
  83#define SND_SOC_DAIFMT_CBM_CFS          (3 << 12) /* codec clk master & frame slave */
  84#define SND_SOC_DAIFMT_CBS_CFS          (4 << 12) /* codec clk & FRM slave */
  85
  86#define SND_SOC_DAIFMT_FORMAT_MASK      0x000f
  87#define SND_SOC_DAIFMT_CLOCK_MASK       0x00f0
  88#define SND_SOC_DAIFMT_INV_MASK         0x0f00
  89#define SND_SOC_DAIFMT_MASTER_MASK      0xf000
  90
  91/*
  92 * Master Clock Directions
  93 */
  94#define SND_SOC_CLOCK_IN                0
  95#define SND_SOC_CLOCK_OUT               1
  96
  97#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
  98                               SNDRV_PCM_FMTBIT_S16_LE |\
  99                               SNDRV_PCM_FMTBIT_S16_BE |\
 100                               SNDRV_PCM_FMTBIT_S20_3LE |\
 101                               SNDRV_PCM_FMTBIT_S20_3BE |\
 102                               SNDRV_PCM_FMTBIT_S20_LE |\
 103                               SNDRV_PCM_FMTBIT_S20_BE |\
 104                               SNDRV_PCM_FMTBIT_S24_3LE |\
 105                               SNDRV_PCM_FMTBIT_S24_3BE |\
 106                               SNDRV_PCM_FMTBIT_S32_LE |\
 107                               SNDRV_PCM_FMTBIT_S32_BE)
 108
 109struct snd_soc_dai_driver;
 110struct snd_soc_dai;
 111struct snd_ac97_bus_ops;
 112
 113/* Digital Audio Interface clocking API.*/
 114int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 115        unsigned int freq, int dir);
 116
 117int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 118        int div_id, int div);
 119
 120int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 121        int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
 122
 123int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
 124
 125/* Digital Audio interface formatting */
 126int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
 127
 128int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 129        unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
 130
 131int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 132        unsigned int tx_num, unsigned int *tx_slot,
 133        unsigned int rx_num, unsigned int *rx_slot);
 134
 135int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 136
 137/* Digital Audio Interface mute */
 138int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 139                             int direction);
 140
 141
 142int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
 143                unsigned int *tx_num, unsigned int *tx_slot,
 144                unsigned int *rx_num, unsigned int *rx_slot);
 145
 146int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 147
 148struct snd_soc_dai_ops {
 149        /*
 150         * DAI clocking configuration, all optional.
 151         * Called by soc_card drivers, normally in their hw_params.
 152         */
 153        int (*set_sysclk)(struct snd_soc_dai *dai,
 154                int clk_id, unsigned int freq, int dir);
 155        int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
 156                unsigned int freq_in, unsigned int freq_out);
 157        int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
 158        int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
 159
 160        /*
 161         * DAI format configuration
 162         * Called by soc_card drivers, normally in their hw_params.
 163         */
 164        int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
 165        int (*xlate_tdm_slot_mask)(unsigned int slots,
 166                unsigned int *tx_mask, unsigned int *rx_mask);
 167        int (*set_tdm_slot)(struct snd_soc_dai *dai,
 168                unsigned int tx_mask, unsigned int rx_mask,
 169                int slots, int slot_width);
 170        int (*set_channel_map)(struct snd_soc_dai *dai,
 171                unsigned int tx_num, unsigned int *tx_slot,
 172                unsigned int rx_num, unsigned int *rx_slot);
 173        int (*get_channel_map)(struct snd_soc_dai *dai,
 174                        unsigned int *tx_num, unsigned int *tx_slot,
 175                        unsigned int *rx_num, unsigned int *rx_slot);
 176        int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 177
 178        int (*set_sdw_stream)(struct snd_soc_dai *dai,
 179                        void *stream, int direction);
 180        /*
 181         * DAI digital mute - optional.
 182         * Called by soc-core to minimise any pops.
 183         */
 184        int (*digital_mute)(struct snd_soc_dai *dai, int mute);
 185        int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 186
 187        /*
 188         * ALSA PCM audio operations - all optional.
 189         * Called by soc-core during audio PCM operations.
 190         */
 191        int (*startup)(struct snd_pcm_substream *,
 192                struct snd_soc_dai *);
 193        void (*shutdown)(struct snd_pcm_substream *,
 194                struct snd_soc_dai *);
 195        int (*hw_params)(struct snd_pcm_substream *,
 196                struct snd_pcm_hw_params *, struct snd_soc_dai *);
 197        int (*hw_free)(struct snd_pcm_substream *,
 198                struct snd_soc_dai *);
 199        int (*prepare)(struct snd_pcm_substream *,
 200                struct snd_soc_dai *);
 201        /*
 202         * NOTE: Commands passed to the trigger function are not necessarily
 203         * compatible with the current state of the dai. For example this
 204         * sequence of commands is possible: START STOP STOP.
 205         * So do not unconditionally use refcounting functions in the trigger
 206         * function, e.g. clk_enable/disable.
 207         */
 208        int (*trigger)(struct snd_pcm_substream *, int,
 209                struct snd_soc_dai *);
 210        int (*bespoke_trigger)(struct snd_pcm_substream *, int,
 211                struct snd_soc_dai *);
 212        /*
 213         * For hardware based FIFO caused delay reporting.
 214         * Optional.
 215         */
 216        snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
 217                struct snd_soc_dai *);
 218};
 219
 220struct snd_soc_cdai_ops {
 221        /*
 222         * for compress ops
 223         */
 224        int (*startup)(struct snd_compr_stream *,
 225                        struct snd_soc_dai *);
 226        int (*shutdown)(struct snd_compr_stream *,
 227                        struct snd_soc_dai *);
 228        int (*set_params)(struct snd_compr_stream *,
 229                        struct snd_compr_params *, struct snd_soc_dai *);
 230        int (*get_params)(struct snd_compr_stream *,
 231                        struct snd_codec *, struct snd_soc_dai *);
 232        int (*set_metadata)(struct snd_compr_stream *,
 233                        struct snd_compr_metadata *, struct snd_soc_dai *);
 234        int (*get_metadata)(struct snd_compr_stream *,
 235                        struct snd_compr_metadata *, struct snd_soc_dai *);
 236        int (*trigger)(struct snd_compr_stream *, int,
 237                        struct snd_soc_dai *);
 238        int (*pointer)(struct snd_compr_stream *,
 239                        struct snd_compr_tstamp *, struct snd_soc_dai *);
 240        int (*ack)(struct snd_compr_stream *, size_t,
 241                        struct snd_soc_dai *);
 242};
 243
 244/*
 245 * Digital Audio Interface Driver.
 246 *
 247 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
 248 * operations and capabilities. Codec and platform drivers will register this
 249 * structure for every DAI they have.
 250 *
 251 * This structure covers the clocking, formating and ALSA operations for each
 252 * interface.
 253 */
 254struct snd_soc_dai_driver {
 255        /* DAI description */
 256        const char *name;
 257        unsigned int id;
 258        unsigned int base;
 259        struct snd_soc_dobj dobj;
 260
 261        /* DAI driver callbacks */
 262        int (*probe)(struct snd_soc_dai *dai);
 263        int (*remove)(struct snd_soc_dai *dai);
 264        int (*suspend)(struct snd_soc_dai *dai);
 265        int (*resume)(struct snd_soc_dai *dai);
 266        /* compress dai */
 267        int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
 268        /* Optional Callback used at pcm creation*/
 269        int (*pcm_new)(struct snd_soc_pcm_runtime *rtd,
 270                       struct snd_soc_dai *dai);
 271        /* DAI is also used for the control bus */
 272        bool bus_control;
 273
 274        /* ops */
 275        const struct snd_soc_dai_ops *ops;
 276        const struct snd_soc_cdai_ops *cops;
 277
 278        /* DAI capabilities */
 279        struct snd_soc_pcm_stream capture;
 280        struct snd_soc_pcm_stream playback;
 281        unsigned int symmetric_rates:1;
 282        unsigned int symmetric_channels:1;
 283        unsigned int symmetric_samplebits:1;
 284
 285        /* probe ordering - for components with runtime dependencies */
 286        int probe_order;
 287        int remove_order;
 288};
 289
 290/*
 291 * Digital Audio Interface runtime data.
 292 *
 293 * Holds runtime data for a DAI.
 294 */
 295struct snd_soc_dai {
 296        const char *name;
 297        int id;
 298        struct device *dev;
 299
 300        /* driver ops */
 301        struct snd_soc_dai_driver *driver;
 302
 303        /* DAI runtime info */
 304        unsigned int capture_active;            /* stream usage count */
 305        unsigned int playback_active;           /* stream usage count */
 306        unsigned int probed:1;
 307
 308        unsigned int active;
 309
 310        struct snd_soc_dapm_widget *playback_widget;
 311        struct snd_soc_dapm_widget *capture_widget;
 312
 313        /* DAI DMA data */
 314        void *playback_dma_data;
 315        void *capture_dma_data;
 316
 317        /* Symmetry data - only valid if symmetry is being enforced */
 318        unsigned int rate;
 319        unsigned int channels;
 320        unsigned int sample_bits;
 321
 322        /* parent platform/codec */
 323        struct snd_soc_component *component;
 324
 325        /* CODEC TDM slot masks and params (for fixup) */
 326        unsigned int tx_mask;
 327        unsigned int rx_mask;
 328
 329        struct list_head list;
 330};
 331
 332static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
 333                                             const struct snd_pcm_substream *ss)
 334{
 335        return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 336                dai->playback_dma_data : dai->capture_dma_data;
 337}
 338
 339static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
 340                                            const struct snd_pcm_substream *ss,
 341                                            void *data)
 342{
 343        if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
 344                dai->playback_dma_data = data;
 345        else
 346                dai->capture_dma_data = data;
 347}
 348
 349static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
 350                                             void *playback, void *capture)
 351{
 352        dai->playback_dma_data = playback;
 353        dai->capture_dma_data = capture;
 354}
 355
 356static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
 357                void *data)
 358{
 359        dev_set_drvdata(dai->dev, data);
 360}
 361
 362static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
 363{
 364        return dev_get_drvdata(dai->dev);
 365}
 366
 367/**
 368 * snd_soc_dai_set_sdw_stream() - Configures a DAI for SDW stream operation
 369 * @dai: DAI
 370 * @stream: STREAM
 371 * @direction: Stream direction(Playback/Capture)
 372 * SoundWire subsystem doesn't have a notion of direction and we reuse
 373 * the ASoC stream direction to configure sink/source ports.
 374 * Playback maps to source ports and Capture for sink ports.
 375 *
 376 * This should be invoked with NULL to clear the stream set previously.
 377 * Returns 0 on success, a negative error code otherwise.
 378 */
 379static inline int snd_soc_dai_set_sdw_stream(struct snd_soc_dai *dai,
 380                                void *stream, int direction)
 381{
 382        if (dai->driver->ops->set_sdw_stream)
 383                return dai->driver->ops->set_sdw_stream(dai, stream, direction);
 384        else
 385                return -ENOTSUPP;
 386}
 387
 388#endif
 389