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
  19 struct snd_pcm_substream;
  20 struct snd_soc_dapm_widget;
  21 struct snd_compr_stream;
  22
  23 /*
  24  * DAI hardware audio formats.
  25  *
  26  * Describes the physical PCM data formating and clocking. Add new formats
  27  * to the end.
  28  */
  29 #define SND_SOC_DAIFMT_I2S              1 /* I2S mode */
  30 #define SND_SOC_DAIFMT_RIGHT_J          2 /* Right Justified mode */
  31 #define SND_SOC_DAIFMT_LEFT_J           3 /* Left Justified mode */
  32 #define SND_SOC_DAIFMT_DSP_A            4 /* L data MSB after FRM LRC */
  33 #define SND_SOC_DAIFMT_DSP_B            5 /* L data MSB during FRM LRC */
  34 #define SND_SOC_DAIFMT_AC97             6 /* AC97 */
  35 #define SND_SOC_DAIFMT_PDM              7 /* Pulse density modulation */
  36
  37 /* left and right justified also known as MSB and LSB respectively */
  38 #define SND_SOC_DAIFMT_MSB              SND_SOC_DAIFMT_LEFT_J
  39 #define SND_SOC_DAIFMT_LSB              SND_SOC_DAIFMT_RIGHT_J
  40
  41 /*
  42  * DAI Clock gating.
  43  *
  44  * DAI bit clocks can be be gated (disabled) when the DAI is not
  45  * sending or receiving PCM data in a frame. This can be used to save power.
  46  */
  47 #define SND_SOC_DAIFMT_CONT             (1 << 4) /* continuous clock */
  48 #define SND_SOC_DAIFMT_GATED            (0 << 4) /* clock is gated */
  49
  50 /*
  51  * DAI hardware signal polarity.
  52  *
  53  * Specifies whether the DAI can also support inverted clocks for the specified
  54  * format.
  55  *
  56  * BCLK:
  57  * - "normal" polarity means signal is available at rising edge of BCLK
  58  * - "inverted" polarity means signal is available at falling edge of BCLK
  59  *
  60  * FSYNC "normal" polarity depends on the frame format:
  61  * - I2S: frame consists of left then right channel data. Left channel starts
  62  *      with falling FSYNC edge, right channel starts with rising FSYNC edge.
  63  * - Left/Right Justified: frame consists of left then right channel data.
  64  *      Left channel starts with rising FSYNC edge, right channel starts with
  65  *      falling FSYNC edge.
  66  * - DSP A/B: Frame starts with rising FSYNC edge.
  67  * - AC97: Frame starts with rising FSYNC edge.
  68  *
  69  * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
  70  */
  71 #define SND_SOC_DAIFMT_NB_NF            (0 << 8) /* normal bit clock + frame */
  72 #define SND_SOC_DAIFMT_NB_IF            (2 << 8) /* normal BCLK + inv FRM */
  73 #define SND_SOC_DAIFMT_IB_NF            (3 << 8) /* invert BCLK + nor FRM */
  74 #define SND_SOC_DAIFMT_IB_IF            (4 << 8) /* invert BCLK + FRM */
  75
  76 /*
  77  * DAI hardware clock masters.
  78  *
  79  * This is wrt the codec, the inverse is true for the interface
  80  * i.e. if the codec is clk and FRM master then the interface is
  81  * clk and frame slave.
  82  */
  83 #define SND_SOC_DAIFMT_CBM_CFM          (1 << 12) /* codec clk & FRM master */
  84 #define SND_SOC_DAIFMT_CBS_CFM          (2 << 12) /* codec clk slave & FRM master */
  85 #define SND_SOC_DAIFMT_CBM_CFS          (3 << 12) /* codec clk master & frame slave */
  86 #define SND_SOC_DAIFMT_CBS_CFS          (4 << 12) /* codec clk & FRM slave */
  87
  88 #define SND_SOC_DAIFMT_FORMAT_MASK      0x000f
  89 #define SND_SOC_DAIFMT_CLOCK_MASK       0x00f0
  90 #define SND_SOC_DAIFMT_INV_MASK         0x0f00
  91 #define SND_SOC_DAIFMT_MASTER_MASK      0xf000
  92
  93 /*
  94  * Master Clock Directions
  95  */
  96 #define SND_SOC_CLOCK_IN                0
  97 #define SND_SOC_CLOCK_OUT               1
  98
  99 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
 100                                SNDRV_PCM_FMTBIT_S16_LE |\
 101                                SNDRV_PCM_FMTBIT_S16_BE |\
 102                                SNDRV_PCM_FMTBIT_S20_3LE |\
 103                                SNDRV_PCM_FMTBIT_S20_3BE |\
 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 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 142
 143 struct snd_soc_dai_ops {
 144         /*
 145          * DAI clocking configuration, all optional.
 146          * Called by soc_card drivers, normally in their hw_params.
 147          */
 148         int (*set_sysclk)(struct snd_soc_dai *dai,
 149                 int clk_id, unsigned int freq, int dir);
 150         int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
 151                 unsigned int freq_in, unsigned int freq_out);
 152         int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
 153         int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);
 154
 155         /*
 156          * DAI format configuration
 157          * Called by soc_card drivers, normally in their hw_params.
 158          */
 159         int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
 160         int (*xlate_tdm_slot_mask)(unsigned int slots,
 161                 unsigned int *tx_mask, unsigned int *rx_mask);
 162         int (*set_tdm_slot)(struct snd_soc_dai *dai,
 163                 unsigned int tx_mask, unsigned int rx_mask,
 164                 int slots, int slot_width);
 165         int (*set_channel_map)(struct snd_soc_dai *dai,
 166                 unsigned int tx_num, unsigned int *tx_slot,
 167                 unsigned int rx_num, unsigned int *rx_slot);
 168         int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 169
 170         /*
 171          * DAI digital mute - optional.
 172          * Called by soc-core to minimise any pops.
 173          */
 174         int (*digital_mute)(struct snd_soc_dai *dai, int mute);
 175         int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 176
 177         /*
 178          * ALSA PCM audio operations - all optional.
 179          * Called by soc-core during audio PCM operations.
 180          */
 181         int (*startup)(struct snd_pcm_substream *,
 182                 struct snd_soc_dai *);
 183         void (*shutdown)(struct snd_pcm_substream *,
 184                 struct snd_soc_dai *);
 185         int (*hw_params)(struct snd_pcm_substream *,
 186                 struct snd_pcm_hw_params *, struct snd_soc_dai *);
 187         int (*hw_free)(struct snd_pcm_substream *,
 188                 struct snd_soc_dai *);
 189         int (*prepare)(struct snd_pcm_substream *,
 190                 struct snd_soc_dai *);
 191         /*
 192          * NOTE: Commands passed to the trigger function are not necessarily
 193          * compatible with the current state of the dai. For example this
 194          * sequence of commands is possible: START STOP STOP.
 195          * So do not unconditionally use refcounting functions in the trigger
 196          * function, e.g. clk_enable/disable.
 197          */
 198         int (*trigger)(struct snd_pcm_substream *, int,
 199                 struct snd_soc_dai *);
 200         int (*bespoke_trigger)(struct snd_pcm_substream *, int,
 201                 struct snd_soc_dai *);
 202         /*
 203          * For hardware based FIFO caused delay reporting.
 204          * Optional.
 205          */
 206         snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
 207                 struct snd_soc_dai *);
 208 };
 209
 210 /*
 211  * Digital Audio Interface Driver.
 212  *
 213  * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
 214  * operations and capabilities. Codec and platform drivers will register this
 215  * structure for every DAI they have.
 216  *
 217  * This structure covers the clocking, formating and ALSA operations for each
 218  * interface.
 219  */
 220 struct snd_soc_dai_driver {
 221         /* DAI description */
 222         const char *name;
 223         unsigned int id;
 224         unsigned int base;
 225
 226         /* DAI driver callbacks */
 227         int (*probe)(struct snd_soc_dai *dai);
 228         int (*remove)(struct snd_soc_dai *dai);
 229         int (*suspend)(struct snd_soc_dai *dai);
 230         int (*resume)(struct snd_soc_dai *dai);
 231         /* compress dai */
 232         int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
 233         /* DAI is also used for the control bus */
 234         bool bus_control;
 235
 236         /* ops */
 237         const struct snd_soc_dai_ops *ops;
 238
 239         /* DAI capabilities */
 240         struct snd_soc_pcm_stream capture;
 241         struct snd_soc_pcm_stream playback;
 242         unsigned int symmetric_rates:1;
 243         unsigned int symmetric_channels:1;
 244         unsigned int symmetric_samplebits:1;
 245
 246         /* probe ordering - for components with runtime dependencies */
 247         int probe_order;
 248         int remove_order;
 249 };
 250
 251 /*
 252  * Digital Audio Interface runtime data.
 253  *
 254  * Holds runtime data for a DAI.
 255  */
 256 struct snd_soc_dai {
 257         const char *name;
 258         int id;
 259         struct device *dev;
 260
 261         /* driver ops */
 262         struct snd_soc_dai_driver *driver;
 263
 264         /* DAI runtime info */
 265         unsigned int capture_active:1;          /* stream is in use */
 266         unsigned int playback_active:1;         /* stream is in use */
 267         unsigned int symmetric_rates:1;
 268         unsigned int symmetric_channels:1;
 269         unsigned int symmetric_samplebits:1;
 270         unsigned int active;
 271         unsigned char probed:1;
 272
 273         struct snd_soc_dapm_widget *playback_widget;
 274         struct snd_soc_dapm_widget *capture_widget;
 275
 276         /* DAI DMA data */
 277         void *playback_dma_data;
 278         void *capture_dma_data;
 279
 280         /* Symmetry data - only valid if symmetry is being enforced */
 281         unsigned int rate;
 282         unsigned int channels;
 283         unsigned int sample_bits;
 284
 285         /* parent platform/codec */
 286         struct snd_soc_codec *codec;
 287         struct snd_soc_component *component;
 288
 289         /* CODEC TDM slot masks and params (for fixup) */
 290         unsigned int tx_mask;
 291         unsigned int rx_mask;
 292
 293         struct list_head list;
 294 };
 295
 296 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
 297                                              const struct snd_pcm_substream *ss)
 298 {
 299         return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 300                 dai->playback_dma_data : dai->capture_dma_data;
 301 }
 302
 303 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
 304                                             const struct snd_pcm_substream *ss,
 305                                             void *data)
 306 {
 307         if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
 308                 dai->playback_dma_data = data;
 309         else
 310                 dai->capture_dma_data = data;
 311 }
 312
 313 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
 314                                              void *playback, void *capture)
 315 {
 316         dai->playback_dma_data = playback;
 317         dai->capture_dma_data = capture;
 318 }
 319
 320 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
 321                 void *data)
 322 {
 323         dev_set_drvdata(dai->dev, data);
 324 }
 325
 326 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
 327 {
 328         return dev_get_drvdata(dai->dev);
 329 }
 330
 331 #endif