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
  17 struct snd_pcm_substream;
  18 struct snd_soc_dapm_widget;
  19 struct 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
 109 struct snd_soc_dai_driver;
 110 struct snd_soc_dai;
 111 struct snd_ac97_bus_ops;
 112
 113 /* Digital Audio Interface clocking API.*/
 114 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 115         unsigned int freq, int dir);
 116
 117 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 118         int div_id, int div);
 119
 120 int 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
 123 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
 124
 125 /* Digital Audio interface formatting */
 126 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
 127
 128 int 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
 131 int 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
 135 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 136
 137 /* Digital Audio Interface mute */
 138 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
 139                              int direction);
 140
 141
 142 int 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
 146 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 147
 148 struct 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
 220 struct 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  */
 254 struct 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  */
 295 struct 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
 332 static 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
 339 static 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
 349 static 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
 356 static 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
 362 static 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  */
 379 static 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