2 * linux/sound/soc-dai.h -- ALSA SoC Layer
4 * Copyright: 2005-2008 Wolfson Microelectronics. PLC.
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.
10 * Digital Audio Interface (DAI) API.
13 #ifndef __LINUX_SND_SOC_DAI_H
14 #define __LINUX_SND_SOC_DAI_H
17 #include <linux/list.h>
18 #include <sound/asoc.h>
20 struct snd_pcm_substream
;
21 struct snd_soc_dapm_widget
;
22 struct snd_compr_stream
;
25 * DAI hardware audio formats.
27 * Describes the physical PCM data formating and clocking. Add new formats
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
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
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.
48 #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
49 #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
52 * DAI hardware signal polarity.
54 * Specifies whether the DAI can also support inverted clocks for the specified
58 * - "normal" polarity means signal is available at rising edge of BCLK
59 * - "inverted" polarity means signal is available at falling edge of BCLK
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
67 * - DSP A/B: Frame starts with rising FSYNC edge.
68 * - AC97: Frame starts with rising FSYNC edge.
70 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
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 */
78 * DAI hardware clock masters.
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.
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 */
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
95 * Master Clock Directions
97 #define SND_SOC_CLOCK_IN 0
98 #define SND_SOC_CLOCK_OUT 1
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_S24_3LE |\
106 SNDRV_PCM_FMTBIT_S24_3BE |\
107 SNDRV_PCM_FMTBIT_S32_LE |\
108 SNDRV_PCM_FMTBIT_S32_BE)
110 struct snd_soc_dai_driver
;
112 struct snd_ac97_bus_ops
;
114 /* Digital Audio Interface clocking API.*/
115 int snd_soc_dai_set_sysclk(struct snd_soc_dai
*dai
, int clk_id
,
116 unsigned int freq
, int dir
);
118 int snd_soc_dai_set_clkdiv(struct snd_soc_dai
*dai
,
119 int div_id
, int div
);
121 int snd_soc_dai_set_pll(struct snd_soc_dai
*dai
,
122 int pll_id
, int source
, unsigned int freq_in
, unsigned int freq_out
);
124 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai
*dai
, unsigned int ratio
);
126 /* Digital Audio interface formatting */
127 int snd_soc_dai_set_fmt(struct snd_soc_dai
*dai
, unsigned int fmt
);
129 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai
*dai
,
130 unsigned int tx_mask
, unsigned int rx_mask
, int slots
, int slot_width
);
132 int snd_soc_dai_set_channel_map(struct snd_soc_dai
*dai
,
133 unsigned int tx_num
, unsigned int *tx_slot
,
134 unsigned int rx_num
, unsigned int *rx_slot
);
136 int snd_soc_dai_set_tristate(struct snd_soc_dai
*dai
, int tristate
);
138 /* Digital Audio Interface mute */
139 int snd_soc_dai_digital_mute(struct snd_soc_dai
*dai
, int mute
,
142 int snd_soc_dai_is_dummy(struct snd_soc_dai
*dai
);
144 struct snd_soc_dai_ops
{
146 * DAI clocking configuration, all optional.
147 * Called by soc_card drivers, normally in their hw_params.
149 int (*set_sysclk
)(struct snd_soc_dai
*dai
,
150 int clk_id
, unsigned int freq
, int dir
);
151 int (*set_pll
)(struct snd_soc_dai
*dai
, int pll_id
, int source
,
152 unsigned int freq_in
, unsigned int freq_out
);
153 int (*set_clkdiv
)(struct snd_soc_dai
*dai
, int div_id
, int div
);
154 int (*set_bclk_ratio
)(struct snd_soc_dai
*dai
, unsigned int ratio
);
157 * DAI format configuration
158 * Called by soc_card drivers, normally in their hw_params.
160 int (*set_fmt
)(struct snd_soc_dai
*dai
, unsigned int fmt
);
161 int (*xlate_tdm_slot_mask
)(unsigned int slots
,
162 unsigned int *tx_mask
, unsigned int *rx_mask
);
163 int (*set_tdm_slot
)(struct snd_soc_dai
*dai
,
164 unsigned int tx_mask
, unsigned int rx_mask
,
165 int slots
, int slot_width
);
166 int (*set_channel_map
)(struct snd_soc_dai
*dai
,
167 unsigned int tx_num
, unsigned int *tx_slot
,
168 unsigned int rx_num
, unsigned int *rx_slot
);
169 int (*set_tristate
)(struct snd_soc_dai
*dai
, int tristate
);
172 * DAI digital mute - optional.
173 * Called by soc-core to minimise any pops.
175 int (*digital_mute
)(struct snd_soc_dai
*dai
, int mute
);
176 int (*mute_stream
)(struct snd_soc_dai
*dai
, int mute
, int stream
);
179 * ALSA PCM audio operations - all optional.
180 * Called by soc-core during audio PCM operations.
182 int (*startup
)(struct snd_pcm_substream
*,
183 struct snd_soc_dai
*);
184 void (*shutdown
)(struct snd_pcm_substream
*,
185 struct snd_soc_dai
*);
186 int (*hw_params
)(struct snd_pcm_substream
*,
187 struct snd_pcm_hw_params
*, struct snd_soc_dai
*);
188 int (*hw_free
)(struct snd_pcm_substream
*,
189 struct snd_soc_dai
*);
190 int (*prepare
)(struct snd_pcm_substream
*,
191 struct snd_soc_dai
*);
193 * NOTE: Commands passed to the trigger function are not necessarily
194 * compatible with the current state of the dai. For example this
195 * sequence of commands is possible: START STOP STOP.
196 * So do not unconditionally use refcounting functions in the trigger
197 * function, e.g. clk_enable/disable.
199 int (*trigger
)(struct snd_pcm_substream
*, int,
200 struct snd_soc_dai
*);
201 int (*bespoke_trigger
)(struct snd_pcm_substream
*, int,
202 struct snd_soc_dai
*);
204 * For hardware based FIFO caused delay reporting.
207 snd_pcm_sframes_t (*delay
)(struct snd_pcm_substream
*,
208 struct snd_soc_dai
*);
211 struct snd_soc_cdai_ops
{
215 int (*startup
)(struct snd_compr_stream
*,
216 struct snd_soc_dai
*);
217 int (*shutdown
)(struct snd_compr_stream
*,
218 struct snd_soc_dai
*);
219 int (*set_params
)(struct snd_compr_stream
*,
220 struct snd_compr_params
*, struct snd_soc_dai
*);
221 int (*get_params
)(struct snd_compr_stream
*,
222 struct snd_codec
*, struct snd_soc_dai
*);
223 int (*set_metadata
)(struct snd_compr_stream
*,
224 struct snd_compr_metadata
*, struct snd_soc_dai
*);
225 int (*get_metadata
)(struct snd_compr_stream
*,
226 struct snd_compr_metadata
*, struct snd_soc_dai
*);
227 int (*trigger
)(struct snd_compr_stream
*, int,
228 struct snd_soc_dai
*);
229 int (*pointer
)(struct snd_compr_stream
*,
230 struct snd_compr_tstamp
*, struct snd_soc_dai
*);
231 int (*ack
)(struct snd_compr_stream
*, size_t,
232 struct snd_soc_dai
*);
236 * Digital Audio Interface Driver.
238 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
239 * operations and capabilities. Codec and platform drivers will register this
240 * structure for every DAI they have.
242 * This structure covers the clocking, formating and ALSA operations for each
245 struct snd_soc_dai_driver
{
246 /* DAI description */
250 struct snd_soc_dobj dobj
;
252 /* DAI driver callbacks */
253 int (*probe
)(struct snd_soc_dai
*dai
);
254 int (*remove
)(struct snd_soc_dai
*dai
);
255 int (*suspend
)(struct snd_soc_dai
*dai
);
256 int (*resume
)(struct snd_soc_dai
*dai
);
258 int (*compress_new
)(struct snd_soc_pcm_runtime
*rtd
, int num
);
259 /* DAI is also used for the control bus */
263 const struct snd_soc_dai_ops
*ops
;
264 const struct snd_soc_cdai_ops
*cops
;
266 /* DAI capabilities */
267 struct snd_soc_pcm_stream capture
;
268 struct snd_soc_pcm_stream playback
;
269 unsigned int symmetric_rates
:1;
270 unsigned int symmetric_channels
:1;
271 unsigned int symmetric_samplebits
:1;
273 /* probe ordering - for components with runtime dependencies */
279 * Digital Audio Interface runtime data.
281 * Holds runtime data for a DAI.
289 struct snd_soc_dai_driver
*driver
;
291 /* DAI runtime info */
292 unsigned int capture_active
:1; /* stream is in use */
293 unsigned int playback_active
:1; /* stream is in use */
294 unsigned int symmetric_rates
:1;
295 unsigned int symmetric_channels
:1;
296 unsigned int symmetric_samplebits
:1;
297 unsigned int probed
:1;
301 struct snd_soc_dapm_widget
*playback_widget
;
302 struct snd_soc_dapm_widget
*capture_widget
;
305 void *playback_dma_data
;
306 void *capture_dma_data
;
308 /* Symmetry data - only valid if symmetry is being enforced */
310 unsigned int channels
;
311 unsigned int sample_bits
;
313 /* parent platform/codec */
314 struct snd_soc_codec
*codec
;
315 struct snd_soc_component
*component
;
317 /* CODEC TDM slot masks and params (for fixup) */
318 unsigned int tx_mask
;
319 unsigned int rx_mask
;
321 struct list_head list
;
324 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai
*dai
,
325 const struct snd_pcm_substream
*ss
)
327 return (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) ?
328 dai
->playback_dma_data
: dai
->capture_dma_data
;
331 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai
*dai
,
332 const struct snd_pcm_substream
*ss
,
335 if (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
336 dai
->playback_dma_data
= data
;
338 dai
->capture_dma_data
= data
;
341 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai
*dai
,
342 void *playback
, void *capture
)
344 dai
->playback_dma_data
= playback
;
345 dai
->capture_dma_data
= capture
;
348 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai
*dai
,
351 dev_set_drvdata(dai
->dev
, data
);
354 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai
*dai
)
356 return dev_get_drvdata(dai
->dev
);