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_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)
112 struct snd_soc_dai_driver
;
114 struct snd_ac97_bus_ops
;
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
);
120 int snd_soc_dai_set_clkdiv(struct snd_soc_dai
*dai
,
121 int div_id
, int div
);
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
);
126 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai
*dai
, unsigned int ratio
);
128 /* Digital Audio interface formatting */
129 int snd_soc_dai_set_fmt(struct snd_soc_dai
*dai
, unsigned int fmt
);
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
);
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
);
138 int snd_soc_dai_set_tristate(struct snd_soc_dai
*dai
, int tristate
);
140 /* Digital Audio Interface mute */
141 int snd_soc_dai_digital_mute(struct snd_soc_dai
*dai
, int mute
,
144 int snd_soc_dai_is_dummy(struct snd_soc_dai
*dai
);
146 struct snd_soc_dai_ops
{
148 * DAI clocking configuration, all optional.
149 * Called by soc_card drivers, normally in their hw_params.
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
);
159 * DAI format configuration
160 * Called by soc_card drivers, normally in their hw_params.
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
);
174 * DAI digital mute - optional.
175 * Called by soc-core to minimise any pops.
177 int (*digital_mute
)(struct snd_soc_dai
*dai
, int mute
);
178 int (*mute_stream
)(struct snd_soc_dai
*dai
, int mute
, int stream
);
181 * ALSA PCM audio operations - all optional.
182 * Called by soc-core during audio PCM operations.
184 int (*startup
)(struct snd_pcm_substream
*,
185 struct snd_soc_dai
*);
186 void (*shutdown
)(struct snd_pcm_substream
*,
187 struct snd_soc_dai
*);
188 int (*hw_params
)(struct snd_pcm_substream
*,
189 struct snd_pcm_hw_params
*, struct snd_soc_dai
*);
190 int (*hw_free
)(struct snd_pcm_substream
*,
191 struct snd_soc_dai
*);
192 int (*prepare
)(struct snd_pcm_substream
*,
193 struct snd_soc_dai
*);
195 * NOTE: Commands passed to the trigger function are not necessarily
196 * compatible with the current state of the dai. For example this
197 * sequence of commands is possible: START STOP STOP.
198 * So do not unconditionally use refcounting functions in the trigger
199 * function, e.g. clk_enable/disable.
201 int (*trigger
)(struct snd_pcm_substream
*, int,
202 struct snd_soc_dai
*);
203 int (*bespoke_trigger
)(struct snd_pcm_substream
*, int,
204 struct snd_soc_dai
*);
206 * For hardware based FIFO caused delay reporting.
209 snd_pcm_sframes_t (*delay
)(struct snd_pcm_substream
*,
210 struct snd_soc_dai
*);
213 struct snd_soc_cdai_ops
{
217 int (*startup
)(struct snd_compr_stream
*,
218 struct snd_soc_dai
*);
219 int (*shutdown
)(struct snd_compr_stream
*,
220 struct snd_soc_dai
*);
221 int (*set_params
)(struct snd_compr_stream
*,
222 struct snd_compr_params
*, struct snd_soc_dai
*);
223 int (*get_params
)(struct snd_compr_stream
*,
224 struct snd_codec
*, struct snd_soc_dai
*);
225 int (*set_metadata
)(struct snd_compr_stream
*,
226 struct snd_compr_metadata
*, struct snd_soc_dai
*);
227 int (*get_metadata
)(struct snd_compr_stream
*,
228 struct snd_compr_metadata
*, struct snd_soc_dai
*);
229 int (*trigger
)(struct snd_compr_stream
*, int,
230 struct snd_soc_dai
*);
231 int (*pointer
)(struct snd_compr_stream
*,
232 struct snd_compr_tstamp
*, struct snd_soc_dai
*);
233 int (*ack
)(struct snd_compr_stream
*, size_t,
234 struct snd_soc_dai
*);
238 * Digital Audio Interface Driver.
240 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
241 * operations and capabilities. Codec and platform drivers will register this
242 * structure for every DAI they have.
244 * This structure covers the clocking, formating and ALSA operations for each
247 struct snd_soc_dai_driver
{
248 /* DAI description */
252 struct snd_soc_dobj dobj
;
254 /* DAI driver callbacks */
255 int (*probe
)(struct snd_soc_dai
*dai
);
256 int (*remove
)(struct snd_soc_dai
*dai
);
257 int (*suspend
)(struct snd_soc_dai
*dai
);
258 int (*resume
)(struct snd_soc_dai
*dai
);
260 int (*compress_new
)(struct snd_soc_pcm_runtime
*rtd
, int num
);
261 /* Optional Callback used at pcm creation*/
262 int (*pcm_new
)(struct snd_soc_pcm_runtime
*rtd
,
263 struct snd_soc_dai
*dai
);
264 /* DAI is also used for the control bus */
268 const struct snd_soc_dai_ops
*ops
;
269 const struct snd_soc_cdai_ops
*cops
;
271 /* DAI capabilities */
272 struct snd_soc_pcm_stream capture
;
273 struct snd_soc_pcm_stream playback
;
274 unsigned int symmetric_rates
:1;
275 unsigned int symmetric_channels
:1;
276 unsigned int symmetric_samplebits
:1;
278 /* probe ordering - for components with runtime dependencies */
284 * Digital Audio Interface runtime data.
286 * Holds runtime data for a DAI.
294 struct snd_soc_dai_driver
*driver
;
296 /* DAI runtime info */
297 unsigned int capture_active
:1; /* stream is in use */
298 unsigned int playback_active
:1; /* stream is in use */
299 unsigned int probed
:1;
303 struct snd_soc_dapm_widget
*playback_widget
;
304 struct snd_soc_dapm_widget
*capture_widget
;
307 void *playback_dma_data
;
308 void *capture_dma_data
;
310 /* Symmetry data - only valid if symmetry is being enforced */
312 unsigned int channels
;
313 unsigned int sample_bits
;
315 /* parent platform/codec */
316 struct snd_soc_codec
*codec
;
317 struct snd_soc_component
*component
;
319 /* CODEC TDM slot masks and params (for fixup) */
320 unsigned int tx_mask
;
321 unsigned int rx_mask
;
323 struct list_head list
;
326 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai
*dai
,
327 const struct snd_pcm_substream
*ss
)
329 return (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) ?
330 dai
->playback_dma_data
: dai
->capture_dma_data
;
333 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai
*dai
,
334 const struct snd_pcm_substream
*ss
,
337 if (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
338 dai
->playback_dma_data
= data
;
340 dai
->capture_dma_data
= data
;
343 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai
*dai
,
344 void *playback
, void *capture
)
346 dai
->playback_dma_data
= playback
;
347 dai
->capture_dma_data
= capture
;
350 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai
*dai
,
353 dev_set_drvdata(dai
->dev
, data
);
356 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai
*dai
)
358 return dev_get_drvdata(dai
->dev
);