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>
19 struct snd_pcm_substream
;
20 struct snd_soc_dapm_widget
;
21 struct snd_compr_stream
;
24 * DAI hardware audio formats.
26 * Describes the physical PCM data formating and clocking. Add new formats
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 */
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
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.
47 #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
48 #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */
51 * DAI hardware signal inversions.
53 * Specifies whether the DAI can also support inverted clocks for the specified
56 #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
57 #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */
58 #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */
59 #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */
62 * DAI hardware clock masters.
64 * This is wrt the codec, the inverse is true for the interface
65 * i.e. if the codec is clk and FRM master then the interface is
66 * clk and frame slave.
68 #define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
69 #define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
70 #define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
71 #define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
73 #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
74 #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
75 #define SND_SOC_DAIFMT_INV_MASK 0x0f00
76 #define SND_SOC_DAIFMT_MASTER_MASK 0xf000
79 * Master Clock Directions
81 #define SND_SOC_CLOCK_IN 0
82 #define SND_SOC_CLOCK_OUT 1
84 #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
85 SNDRV_PCM_FMTBIT_S16_LE |\
86 SNDRV_PCM_FMTBIT_S16_BE |\
87 SNDRV_PCM_FMTBIT_S20_3LE |\
88 SNDRV_PCM_FMTBIT_S20_3BE |\
89 SNDRV_PCM_FMTBIT_S24_3LE |\
90 SNDRV_PCM_FMTBIT_S24_3BE |\
91 SNDRV_PCM_FMTBIT_S32_LE |\
92 SNDRV_PCM_FMTBIT_S32_BE)
94 struct snd_soc_dai_driver
;
96 struct snd_ac97_bus_ops
;
98 /* Digital Audio Interface clocking API.*/
99 int snd_soc_dai_set_sysclk(struct snd_soc_dai
*dai
, int clk_id
,
100 unsigned int freq
, int dir
);
102 int snd_soc_dai_set_clkdiv(struct snd_soc_dai
*dai
,
103 int div_id
, int div
);
105 int snd_soc_dai_set_pll(struct snd_soc_dai
*dai
,
106 int pll_id
, int source
, unsigned int freq_in
, unsigned int freq_out
);
108 int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai
*dai
, unsigned int ratio
);
110 /* Digital Audio interface formatting */
111 int snd_soc_dai_set_fmt(struct snd_soc_dai
*dai
, unsigned int fmt
);
113 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai
*dai
,
114 unsigned int tx_mask
, unsigned int rx_mask
, int slots
, int slot_width
);
116 int snd_soc_dai_set_channel_map(struct snd_soc_dai
*dai
,
117 unsigned int tx_num
, unsigned int *tx_slot
,
118 unsigned int rx_num
, unsigned int *rx_slot
);
120 int snd_soc_dai_set_tristate(struct snd_soc_dai
*dai
, int tristate
);
122 /* Digital Audio Interface mute */
123 int snd_soc_dai_digital_mute(struct snd_soc_dai
*dai
, int mute
,
126 struct snd_soc_dai_ops
{
128 * DAI clocking configuration, all optional.
129 * Called by soc_card drivers, normally in their hw_params.
131 int (*set_sysclk
)(struct snd_soc_dai
*dai
,
132 int clk_id
, unsigned int freq
, int dir
);
133 int (*set_pll
)(struct snd_soc_dai
*dai
, int pll_id
, int source
,
134 unsigned int freq_in
, unsigned int freq_out
);
135 int (*set_clkdiv
)(struct snd_soc_dai
*dai
, int div_id
, int div
);
136 int (*set_bclk_ratio
)(struct snd_soc_dai
*dai
, unsigned int ratio
);
139 * DAI format configuration
140 * Called by soc_card drivers, normally in their hw_params.
142 int (*set_fmt
)(struct snd_soc_dai
*dai
, unsigned int fmt
);
143 int (*set_tdm_slot
)(struct snd_soc_dai
*dai
,
144 unsigned int tx_mask
, unsigned int rx_mask
,
145 int slots
, int slot_width
);
146 int (*set_channel_map
)(struct snd_soc_dai
*dai
,
147 unsigned int tx_num
, unsigned int *tx_slot
,
148 unsigned int rx_num
, unsigned int *rx_slot
);
149 int (*set_tristate
)(struct snd_soc_dai
*dai
, int tristate
);
152 * DAI digital mute - optional.
153 * Called by soc-core to minimise any pops.
155 int (*digital_mute
)(struct snd_soc_dai
*dai
, int mute
);
156 int (*mute_stream
)(struct snd_soc_dai
*dai
, int mute
, int stream
);
159 * ALSA PCM audio operations - all optional.
160 * Called by soc-core during audio PCM operations.
162 int (*startup
)(struct snd_pcm_substream
*,
163 struct snd_soc_dai
*);
164 void (*shutdown
)(struct snd_pcm_substream
*,
165 struct snd_soc_dai
*);
166 int (*hw_params
)(struct snd_pcm_substream
*,
167 struct snd_pcm_hw_params
*, struct snd_soc_dai
*);
168 int (*hw_free
)(struct snd_pcm_substream
*,
169 struct snd_soc_dai
*);
170 int (*prepare
)(struct snd_pcm_substream
*,
171 struct snd_soc_dai
*);
173 * NOTE: Commands passed to the trigger function are not necessarily
174 * compatible with the current state of the dai. For example this
175 * sequence of commands is possible: START STOP STOP.
176 * So do not unconditionally use refcounting functions in the trigger
177 * function, e.g. clk_enable/disable.
179 int (*trigger
)(struct snd_pcm_substream
*, int,
180 struct snd_soc_dai
*);
181 int (*bespoke_trigger
)(struct snd_pcm_substream
*, int,
182 struct snd_soc_dai
*);
184 * For hardware based FIFO caused delay reporting.
187 snd_pcm_sframes_t (*delay
)(struct snd_pcm_substream
*,
188 struct snd_soc_dai
*);
192 * Digital Audio Interface Driver.
194 * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
195 * operations and capabilities. Codec and platform drivers will register this
196 * structure for every DAI they have.
198 * This structure covers the clocking, formating and ALSA operations for each
201 struct snd_soc_dai_driver
{
202 /* DAI description */
208 /* DAI driver callbacks */
209 int (*probe
)(struct snd_soc_dai
*dai
);
210 int (*remove
)(struct snd_soc_dai
*dai
);
211 int (*suspend
)(struct snd_soc_dai
*dai
);
212 int (*resume
)(struct snd_soc_dai
*dai
);
217 const struct snd_soc_dai_ops
*ops
;
219 /* DAI capabilities */
220 struct snd_soc_pcm_stream capture
;
221 struct snd_soc_pcm_stream playback
;
222 unsigned int symmetric_rates
:1;
224 /* probe ordering - for components with runtime dependencies */
230 * Digital Audio Interface runtime data.
232 * Holds runtime data for a DAI.
238 void *ac97_pdata
; /* platform_data for the ac97 codec */
241 struct snd_soc_dai_driver
*driver
;
243 /* DAI runtime info */
244 unsigned int capture_active
:1; /* stream is in use */
245 unsigned int playback_active
:1; /* stream is in use */
246 unsigned int symmetric_rates
:1;
247 struct snd_pcm_runtime
*runtime
;
249 unsigned char probed
:1;
251 struct snd_soc_dapm_widget
*playback_widget
;
252 struct snd_soc_dapm_widget
*capture_widget
;
253 struct snd_soc_dapm_context dapm
;
256 void *playback_dma_data
;
257 void *capture_dma_data
;
259 /* Symmetry data - only valid if symmetry is being enforced */
262 /* parent platform/codec */
263 struct snd_soc_platform
*platform
;
264 struct snd_soc_codec
*codec
;
266 struct snd_soc_card
*card
;
268 struct list_head list
;
269 struct list_head card_list
;
272 static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai
*dai
,
273 const struct snd_pcm_substream
*ss
)
275 return (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) ?
276 dai
->playback_dma_data
: dai
->capture_dma_data
;
279 static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai
*dai
,
280 const struct snd_pcm_substream
*ss
,
283 if (ss
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
284 dai
->playback_dma_data
= data
;
286 dai
->capture_dma_data
= data
;
289 static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai
*dai
,
290 void *playback
, void *capture
)
292 dai
->playback_dma_data
= playback
;
293 dai
->capture_dma_data
= capture
;
296 static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai
*dai
,
299 dev_set_drvdata(dai
->dev
, data
);
302 static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai
*dai
)
304 return dev_get_drvdata(dai
->dev
);