LXR linux/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
  19struct snd_pcm_substream;
  20struct snd_soc_dapm_widget;
  21struct 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
 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
 141int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);
 142
 143struct 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 */
 220struct 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 */
 256struct 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
 296static 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
 303static 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
 313static 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
 320static 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
 326static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
 327{
 328        return dev_get_drvdata(dai->dev);
 329}
 330
 331#endif
 332