include/sound/soc-dai.h

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