2 * soc-core.c -- ALSA SoC Audio Layer
4 * Copyright 2005 Wolfson Microelectronics PLC.
5 * Copyright 2005 Openedhand Ltd.
7 * Author: Liam Girdwood
8 * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
9 * with code, comments and ideas from :-
10 * Richard Purdie <richard@openedhand.com>
12 * This program is free software; you can redistribute it and/or modify it
13 * under the terms of the GNU General Public License as published by the
14 * Free Software Foundation; either version 2 of the License, or (at your
15 * option) any later version.
18 * o Add hw rules to enforce rates, etc.
19 * o More testing with other codecs/machines.
20 * o Add more codecs and platforms to ensure good API coverage.
21 * o Support TDM on PCM and I2S
24 #include <linux/module.h>
25 #include <linux/moduleparam.h>
26 #include <linux/init.h>
27 #include <linux/delay.h>
29 #include <linux/bitops.h>
30 #include <linux/platform_device.h>
31 #include <sound/core.h>
32 #include <sound/pcm.h>
33 #include <sound/pcm_params.h>
34 #include <sound/soc.h>
35 #include <sound/soc-dapm.h>
36 #include <sound/initval.h>
41 #define dbg(format, arg...) printk(format, ## arg)
43 #define dbg(format, arg...)
46 static DEFINE_MUTEX(pcm_mutex
);
47 static DEFINE_MUTEX(io_mutex
);
48 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq
);
51 * This is a timeout to do a DAPM powerdown after a stream is closed().
52 * It can be used to eliminate pops between different playback streams, e.g.
53 * between two audio tracks.
55 static int pmdown_time
= 5000;
56 module_param(pmdown_time
, int, 0);
57 MODULE_PARM_DESC(pmdown_time
, "DAPM stream powerdown time (msecs)");
60 * This function forces any delayed work to be queued and run.
62 static int run_delayed_work(struct delayed_work
*dwork
)
66 /* cancel any work waiting to be queued. */
67 ret
= cancel_delayed_work(dwork
);
69 /* if there was any work waiting then we run it now and
70 * wait for it's completion */
72 schedule_delayed_work(dwork
, 0);
73 flush_scheduled_work();
78 #ifdef CONFIG_SND_SOC_AC97_BUS
79 /* unregister ac97 codec */
80 static int soc_ac97_dev_unregister(struct snd_soc_codec
*codec
)
82 if (codec
->ac97
->dev
.bus
)
83 device_unregister(&codec
->ac97
->dev
);
87 /* stop no dev release warning */
88 static void soc_ac97_device_release(struct device
*dev
){}
90 /* register ac97 codec to bus */
91 static int soc_ac97_dev_register(struct snd_soc_codec
*codec
)
95 codec
->ac97
->dev
.bus
= &ac97_bus_type
;
96 codec
->ac97
->dev
.parent
= NULL
;
97 codec
->ac97
->dev
.release
= soc_ac97_device_release
;
99 snprintf(codec
->ac97
->dev
.bus_id
, BUS_ID_SIZE
, "%d-%d:%s",
100 codec
->card
->number
, 0, codec
->name
);
101 err
= device_register(&codec
->ac97
->dev
);
103 snd_printk(KERN_ERR
"Can't register ac97 bus\n");
104 codec
->ac97
->dev
.bus
= NULL
;
111 static inline const char *get_dai_name(int type
)
114 case SND_SOC_DAI_AC97_BUS
:
115 case SND_SOC_DAI_AC97
:
117 case SND_SOC_DAI_I2S
:
119 case SND_SOC_DAI_PCM
:
126 * Called by ALSA when a PCM substream is opened, the runtime->hw record is
127 * then initialized and any private data can be allocated. This also calls
128 * startup for the cpu DAI, platform, machine and codec DAI.
130 static int soc_pcm_open(struct snd_pcm_substream
*substream
)
132 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
133 struct snd_soc_device
*socdev
= rtd
->socdev
;
134 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
135 struct snd_soc_dai_link
*machine
= rtd
->dai
;
136 struct snd_soc_platform
*platform
= socdev
->platform
;
137 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
138 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
141 mutex_lock(&pcm_mutex
);
143 /* startup the audio subsystem */
144 if (cpu_dai
->ops
.startup
) {
145 ret
= cpu_dai
->ops
.startup(substream
);
147 printk(KERN_ERR
"asoc: can't open interface %s\n",
153 if (platform
->pcm_ops
->open
) {
154 ret
= platform
->pcm_ops
->open(substream
);
156 printk(KERN_ERR
"asoc: can't open platform %s\n", platform
->name
);
161 if (codec_dai
->ops
.startup
) {
162 ret
= codec_dai
->ops
.startup(substream
);
164 printk(KERN_ERR
"asoc: can't open codec %s\n",
170 if (machine
->ops
&& machine
->ops
->startup
) {
171 ret
= machine
->ops
->startup(substream
);
173 printk(KERN_ERR
"asoc: %s startup failed\n", machine
->name
);
178 /* Check that the codec and cpu DAI's are compatible */
179 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) {
180 runtime
->hw
.rate_min
=
181 max(codec_dai
->playback
.rate_min
,
182 cpu_dai
->playback
.rate_min
);
183 runtime
->hw
.rate_max
=
184 min(codec_dai
->playback
.rate_max
,
185 cpu_dai
->playback
.rate_max
);
186 runtime
->hw
.channels_min
=
187 max(codec_dai
->playback
.channels_min
,
188 cpu_dai
->playback
.channels_min
);
189 runtime
->hw
.channels_max
=
190 min(codec_dai
->playback
.channels_max
,
191 cpu_dai
->playback
.channels_max
);
192 runtime
->hw
.formats
=
193 codec_dai
->playback
.formats
& cpu_dai
->playback
.formats
;
195 codec_dai
->playback
.rates
& cpu_dai
->playback
.rates
;
197 runtime
->hw
.rate_min
=
198 max(codec_dai
->capture
.rate_min
,
199 cpu_dai
->capture
.rate_min
);
200 runtime
->hw
.rate_max
=
201 min(codec_dai
->capture
.rate_max
,
202 cpu_dai
->capture
.rate_max
);
203 runtime
->hw
.channels_min
=
204 max(codec_dai
->capture
.channels_min
,
205 cpu_dai
->capture
.channels_min
);
206 runtime
->hw
.channels_max
=
207 min(codec_dai
->capture
.channels_max
,
208 cpu_dai
->capture
.channels_max
);
209 runtime
->hw
.formats
=
210 codec_dai
->capture
.formats
& cpu_dai
->capture
.formats
;
212 codec_dai
->capture
.rates
& cpu_dai
->capture
.rates
;
215 snd_pcm_limit_hw_rates(runtime
);
216 if (!runtime
->hw
.rates
) {
217 printk(KERN_ERR
"asoc: %s <-> %s No matching rates\n",
218 codec_dai
->name
, cpu_dai
->name
);
221 if (!runtime
->hw
.formats
) {
222 printk(KERN_ERR
"asoc: %s <-> %s No matching formats\n",
223 codec_dai
->name
, cpu_dai
->name
);
226 if (!runtime
->hw
.channels_min
|| !runtime
->hw
.channels_max
) {
227 printk(KERN_ERR
"asoc: %s <-> %s No matching channels\n",
228 codec_dai
->name
, cpu_dai
->name
);
232 dbg("asoc: %s <-> %s info:\n", codec_dai
->name
, cpu_dai
->name
);
233 dbg("asoc: rate mask 0x%x\n", runtime
->hw
.rates
);
234 dbg("asoc: min ch %d max ch %d\n", runtime
->hw
.channels_min
,
235 runtime
->hw
.channels_max
);
236 dbg("asoc: min rate %d max rate %d\n", runtime
->hw
.rate_min
,
237 runtime
->hw
.rate_max
);
239 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
240 cpu_dai
->playback
.active
= codec_dai
->playback
.active
= 1;
242 cpu_dai
->capture
.active
= codec_dai
->capture
.active
= 1;
243 cpu_dai
->active
= codec_dai
->active
= 1;
244 cpu_dai
->runtime
= runtime
;
245 socdev
->codec
->active
++;
246 mutex_unlock(&pcm_mutex
);
250 if (machine
->ops
&& machine
->ops
->shutdown
)
251 machine
->ops
->shutdown(substream
);
254 if (platform
->pcm_ops
->close
)
255 platform
->pcm_ops
->close(substream
);
258 if (cpu_dai
->ops
.shutdown
)
259 cpu_dai
->ops
.shutdown(substream
);
261 mutex_unlock(&pcm_mutex
);
266 * Power down the audio subsystem pmdown_time msecs after close is called.
267 * This is to ensure there are no pops or clicks in between any music tracks
268 * due to DAPM power cycling.
270 static void close_delayed_work(struct work_struct
*work
)
272 struct snd_soc_device
*socdev
=
273 container_of(work
, struct snd_soc_device
, delayed_work
.work
);
274 struct snd_soc_codec
*codec
= socdev
->codec
;
275 struct snd_soc_codec_dai
*codec_dai
;
278 mutex_lock(&pcm_mutex
);
279 for (i
= 0; i
< codec
->num_dai
; i
++) {
280 codec_dai
= &codec
->dai
[i
];
282 dbg("pop wq checking: %s status: %s waiting: %s\n",
283 codec_dai
->playback
.stream_name
,
284 codec_dai
->playback
.active
? "active" : "inactive",
285 codec_dai
->pop_wait
? "yes" : "no");
287 /* are we waiting on this codec DAI stream */
288 if (codec_dai
->pop_wait
== 1) {
290 /* Reduce power if no longer active */
291 if (codec
->active
== 0) {
292 dbg("pop wq D1 %s %s\n", codec
->name
,
293 codec_dai
->playback
.stream_name
);
294 snd_soc_dapm_set_bias_level(socdev
,
295 SND_SOC_BIAS_PREPARE
);
298 codec_dai
->pop_wait
= 0;
299 snd_soc_dapm_stream_event(codec
,
300 codec_dai
->playback
.stream_name
,
301 SND_SOC_DAPM_STREAM_STOP
);
303 /* Fall into standby if no longer active */
304 if (codec
->active
== 0) {
305 dbg("pop wq D3 %s %s\n", codec
->name
,
306 codec_dai
->playback
.stream_name
);
307 snd_soc_dapm_set_bias_level(socdev
,
308 SND_SOC_BIAS_STANDBY
);
312 mutex_unlock(&pcm_mutex
);
316 * Called by ALSA when a PCM substream is closed. Private data can be
317 * freed here. The cpu DAI, codec DAI, machine and platform are also
320 static int soc_codec_close(struct snd_pcm_substream
*substream
)
322 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
323 struct snd_soc_device
*socdev
= rtd
->socdev
;
324 struct snd_soc_dai_link
*machine
= rtd
->dai
;
325 struct snd_soc_platform
*platform
= socdev
->platform
;
326 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
327 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
328 struct snd_soc_codec
*codec
= socdev
->codec
;
330 mutex_lock(&pcm_mutex
);
332 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
333 cpu_dai
->playback
.active
= codec_dai
->playback
.active
= 0;
335 cpu_dai
->capture
.active
= codec_dai
->capture
.active
= 0;
337 if (codec_dai
->playback
.active
== 0 &&
338 codec_dai
->capture
.active
== 0) {
339 cpu_dai
->active
= codec_dai
->active
= 0;
343 if (cpu_dai
->ops
.shutdown
)
344 cpu_dai
->ops
.shutdown(substream
);
346 if (codec_dai
->ops
.shutdown
)
347 codec_dai
->ops
.shutdown(substream
);
349 if (machine
->ops
&& machine
->ops
->shutdown
)
350 machine
->ops
->shutdown(substream
);
352 if (platform
->pcm_ops
->close
)
353 platform
->pcm_ops
->close(substream
);
354 cpu_dai
->runtime
= NULL
;
356 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) {
357 /* start delayed pop wq here for playback streams */
358 codec_dai
->pop_wait
= 1;
359 schedule_delayed_work(&socdev
->delayed_work
,
360 msecs_to_jiffies(pmdown_time
));
362 /* capture streams can be powered down now */
363 snd_soc_dapm_stream_event(codec
,
364 codec_dai
->capture
.stream_name
,
365 SND_SOC_DAPM_STREAM_STOP
);
367 if (codec
->active
== 0 && codec_dai
->pop_wait
== 0)
368 snd_soc_dapm_set_bias_level(socdev
,
369 SND_SOC_BIAS_STANDBY
);
372 mutex_unlock(&pcm_mutex
);
377 * Called by ALSA when the PCM substream is prepared, can set format, sample
378 * rate, etc. This function is non atomic and can be called multiple times,
379 * it can refer to the runtime info.
381 static int soc_pcm_prepare(struct snd_pcm_substream
*substream
)
383 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
384 struct snd_soc_device
*socdev
= rtd
->socdev
;
385 struct snd_soc_dai_link
*machine
= rtd
->dai
;
386 struct snd_soc_platform
*platform
= socdev
->platform
;
387 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
388 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
389 struct snd_soc_codec
*codec
= socdev
->codec
;
392 mutex_lock(&pcm_mutex
);
394 if (machine
->ops
&& machine
->ops
->prepare
) {
395 ret
= machine
->ops
->prepare(substream
);
397 printk(KERN_ERR
"asoc: machine prepare error\n");
402 if (platform
->pcm_ops
->prepare
) {
403 ret
= platform
->pcm_ops
->prepare(substream
);
405 printk(KERN_ERR
"asoc: platform prepare error\n");
410 if (codec_dai
->ops
.prepare
) {
411 ret
= codec_dai
->ops
.prepare(substream
);
413 printk(KERN_ERR
"asoc: codec DAI prepare error\n");
418 if (cpu_dai
->ops
.prepare
) {
419 ret
= cpu_dai
->ops
.prepare(substream
);
421 printk(KERN_ERR
"asoc: cpu DAI prepare error\n");
426 /* we only want to start a DAPM playback stream if we are not waiting
427 * on an existing one stopping */
428 if (codec_dai
->pop_wait
) {
429 /* we are waiting for the delayed work to start */
430 if (substream
->stream
== SNDRV_PCM_STREAM_CAPTURE
)
431 snd_soc_dapm_stream_event(socdev
->codec
,
432 codec_dai
->capture
.stream_name
,
433 SND_SOC_DAPM_STREAM_START
);
435 codec_dai
->pop_wait
= 0;
436 cancel_delayed_work(&socdev
->delayed_work
);
437 if (codec_dai
->dai_ops
.digital_mute
)
438 codec_dai
->dai_ops
.digital_mute(codec_dai
, 0);
441 /* no delayed work - do we need to power up codec */
442 if (codec
->bias_level
!= SND_SOC_BIAS_ON
) {
444 snd_soc_dapm_set_bias_level(socdev
,
445 SND_SOC_BIAS_PREPARE
);
447 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
448 snd_soc_dapm_stream_event(codec
,
449 codec_dai
->playback
.stream_name
,
450 SND_SOC_DAPM_STREAM_START
);
452 snd_soc_dapm_stream_event(codec
,
453 codec_dai
->capture
.stream_name
,
454 SND_SOC_DAPM_STREAM_START
);
456 snd_soc_dapm_set_bias_level(socdev
, SND_SOC_BIAS_ON
);
457 if (codec_dai
->dai_ops
.digital_mute
)
458 codec_dai
->dai_ops
.digital_mute(codec_dai
, 0);
461 /* codec already powered - power on widgets */
462 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
463 snd_soc_dapm_stream_event(codec
,
464 codec_dai
->playback
.stream_name
,
465 SND_SOC_DAPM_STREAM_START
);
467 snd_soc_dapm_stream_event(codec
,
468 codec_dai
->capture
.stream_name
,
469 SND_SOC_DAPM_STREAM_START
);
470 if (codec_dai
->dai_ops
.digital_mute
)
471 codec_dai
->dai_ops
.digital_mute(codec_dai
, 0);
476 mutex_unlock(&pcm_mutex
);
481 * Called by ALSA when the hardware params are set by application. This
482 * function can also be called multiple times and can allocate buffers
483 * (using snd_pcm_lib_* ). It's non-atomic.
485 static int soc_pcm_hw_params(struct snd_pcm_substream
*substream
,
486 struct snd_pcm_hw_params
*params
)
488 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
489 struct snd_soc_device
*socdev
= rtd
->socdev
;
490 struct snd_soc_dai_link
*machine
= rtd
->dai
;
491 struct snd_soc_platform
*platform
= socdev
->platform
;
492 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
493 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
496 mutex_lock(&pcm_mutex
);
498 if (machine
->ops
&& machine
->ops
->hw_params
) {
499 ret
= machine
->ops
->hw_params(substream
, params
);
501 printk(KERN_ERR
"asoc: machine hw_params failed\n");
506 if (codec_dai
->ops
.hw_params
) {
507 ret
= codec_dai
->ops
.hw_params(substream
, params
);
509 printk(KERN_ERR
"asoc: can't set codec %s hw params\n",
515 if (cpu_dai
->ops
.hw_params
) {
516 ret
= cpu_dai
->ops
.hw_params(substream
, params
);
518 printk(KERN_ERR
"asoc: interface %s hw params failed\n",
524 if (platform
->pcm_ops
->hw_params
) {
525 ret
= platform
->pcm_ops
->hw_params(substream
, params
);
527 printk(KERN_ERR
"asoc: platform %s hw params failed\n",
534 mutex_unlock(&pcm_mutex
);
538 if (cpu_dai
->ops
.hw_free
)
539 cpu_dai
->ops
.hw_free(substream
);
542 if (codec_dai
->ops
.hw_free
)
543 codec_dai
->ops
.hw_free(substream
);
546 if (machine
->ops
&& machine
->ops
->hw_free
)
547 machine
->ops
->hw_free(substream
);
549 mutex_unlock(&pcm_mutex
);
554 * Free's resources allocated by hw_params, can be called multiple times
556 static int soc_pcm_hw_free(struct snd_pcm_substream
*substream
)
558 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
559 struct snd_soc_device
*socdev
= rtd
->socdev
;
560 struct snd_soc_dai_link
*machine
= rtd
->dai
;
561 struct snd_soc_platform
*platform
= socdev
->platform
;
562 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
563 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
564 struct snd_soc_codec
*codec
= socdev
->codec
;
566 mutex_lock(&pcm_mutex
);
568 /* apply codec digital mute */
569 if (!codec
->active
&& codec_dai
->dai_ops
.digital_mute
)
570 codec_dai
->dai_ops
.digital_mute(codec_dai
, 1);
572 /* free any machine hw params */
573 if (machine
->ops
&& machine
->ops
->hw_free
)
574 machine
->ops
->hw_free(substream
);
576 /* free any DMA resources */
577 if (platform
->pcm_ops
->hw_free
)
578 platform
->pcm_ops
->hw_free(substream
);
580 /* now free hw params for the DAI's */
581 if (codec_dai
->ops
.hw_free
)
582 codec_dai
->ops
.hw_free(substream
);
584 if (cpu_dai
->ops
.hw_free
)
585 cpu_dai
->ops
.hw_free(substream
);
587 mutex_unlock(&pcm_mutex
);
591 static int soc_pcm_trigger(struct snd_pcm_substream
*substream
, int cmd
)
593 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
594 struct snd_soc_device
*socdev
= rtd
->socdev
;
595 struct snd_soc_dai_link
*machine
= rtd
->dai
;
596 struct snd_soc_platform
*platform
= socdev
->platform
;
597 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
598 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
601 if (codec_dai
->ops
.trigger
) {
602 ret
= codec_dai
->ops
.trigger(substream
, cmd
);
607 if (platform
->pcm_ops
->trigger
) {
608 ret
= platform
->pcm_ops
->trigger(substream
, cmd
);
613 if (cpu_dai
->ops
.trigger
) {
614 ret
= cpu_dai
->ops
.trigger(substream
, cmd
);
621 /* ASoC PCM operations */
622 static struct snd_pcm_ops soc_pcm_ops
= {
623 .open
= soc_pcm_open
,
624 .close
= soc_codec_close
,
625 .hw_params
= soc_pcm_hw_params
,
626 .hw_free
= soc_pcm_hw_free
,
627 .prepare
= soc_pcm_prepare
,
628 .trigger
= soc_pcm_trigger
,
632 /* powers down audio subsystem for suspend */
633 static int soc_suspend(struct platform_device
*pdev
, pm_message_t state
)
635 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
636 struct snd_soc_machine
*machine
= socdev
->machine
;
637 struct snd_soc_platform
*platform
= socdev
->platform
;
638 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
639 struct snd_soc_codec
*codec
= socdev
->codec
;
642 /* mute any active DAC's */
643 for (i
= 0; i
< machine
->num_links
; i
++) {
644 struct snd_soc_codec_dai
*dai
= machine
->dai_link
[i
].codec_dai
;
645 if (dai
->dai_ops
.digital_mute
&& dai
->playback
.active
)
646 dai
->dai_ops
.digital_mute(dai
, 1);
649 /* suspend all pcms */
650 for (i
= 0; i
< machine
->num_links
; i
++)
651 snd_pcm_suspend_all(machine
->dai_link
[i
].pcm
);
653 if (machine
->suspend_pre
)
654 machine
->suspend_pre(pdev
, state
);
656 for (i
= 0; i
< machine
->num_links
; i
++) {
657 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
658 if (cpu_dai
->suspend
&& cpu_dai
->type
!= SND_SOC_DAI_AC97
)
659 cpu_dai
->suspend(pdev
, cpu_dai
);
660 if (platform
->suspend
)
661 platform
->suspend(pdev
, cpu_dai
);
664 /* close any waiting streams and save state */
665 run_delayed_work(&socdev
->delayed_work
);
666 codec
->suspend_bias_level
= codec
->bias_level
;
668 for (i
= 0; i
< codec
->num_dai
; i
++) {
669 char *stream
= codec
->dai
[i
].playback
.stream_name
;
671 snd_soc_dapm_stream_event(codec
, stream
,
672 SND_SOC_DAPM_STREAM_SUSPEND
);
673 stream
= codec
->dai
[i
].capture
.stream_name
;
675 snd_soc_dapm_stream_event(codec
, stream
,
676 SND_SOC_DAPM_STREAM_SUSPEND
);
679 if (codec_dev
->suspend
)
680 codec_dev
->suspend(pdev
, state
);
682 for (i
= 0; i
< machine
->num_links
; i
++) {
683 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
684 if (cpu_dai
->suspend
&& cpu_dai
->type
== SND_SOC_DAI_AC97
)
685 cpu_dai
->suspend(pdev
, cpu_dai
);
688 if (machine
->suspend_post
)
689 machine
->suspend_post(pdev
, state
);
694 /* powers up audio subsystem after a suspend */
695 static int soc_resume(struct platform_device
*pdev
)
697 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
698 struct snd_soc_machine
*machine
= socdev
->machine
;
699 struct snd_soc_platform
*platform
= socdev
->platform
;
700 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
701 struct snd_soc_codec
*codec
= socdev
->codec
;
704 if (machine
->resume_pre
)
705 machine
->resume_pre(pdev
);
707 for (i
= 0; i
< machine
->num_links
; i
++) {
708 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
709 if (cpu_dai
->resume
&& cpu_dai
->type
== SND_SOC_DAI_AC97
)
710 cpu_dai
->resume(pdev
, cpu_dai
);
713 if (codec_dev
->resume
)
714 codec_dev
->resume(pdev
);
716 for (i
= 0; i
< codec
->num_dai
; i
++) {
717 char *stream
= codec
->dai
[i
].playback
.stream_name
;
719 snd_soc_dapm_stream_event(codec
, stream
,
720 SND_SOC_DAPM_STREAM_RESUME
);
721 stream
= codec
->dai
[i
].capture
.stream_name
;
723 snd_soc_dapm_stream_event(codec
, stream
,
724 SND_SOC_DAPM_STREAM_RESUME
);
727 /* unmute any active DACs */
728 for (i
= 0; i
< machine
->num_links
; i
++) {
729 struct snd_soc_codec_dai
*dai
= machine
->dai_link
[i
].codec_dai
;
730 if (dai
->dai_ops
.digital_mute
&& dai
->playback
.active
)
731 dai
->dai_ops
.digital_mute(dai
, 0);
734 for (i
= 0; i
< machine
->num_links
; i
++) {
735 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
736 if (cpu_dai
->resume
&& cpu_dai
->type
!= SND_SOC_DAI_AC97
)
737 cpu_dai
->resume(pdev
, cpu_dai
);
738 if (platform
->resume
)
739 platform
->resume(pdev
, cpu_dai
);
742 if (machine
->resume_post
)
743 machine
->resume_post(pdev
);
749 #define soc_suspend NULL
750 #define soc_resume NULL
753 /* probes a new socdev */
754 static int soc_probe(struct platform_device
*pdev
)
757 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
758 struct snd_soc_machine
*machine
= socdev
->machine
;
759 struct snd_soc_platform
*platform
= socdev
->platform
;
760 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
762 if (machine
->probe
) {
763 ret
= machine
->probe(pdev
);
768 for (i
= 0; i
< machine
->num_links
; i
++) {
769 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
770 if (cpu_dai
->probe
) {
771 ret
= cpu_dai
->probe(pdev
);
777 if (codec_dev
->probe
) {
778 ret
= codec_dev
->probe(pdev
);
783 if (platform
->probe
) {
784 ret
= platform
->probe(pdev
);
789 /* DAPM stream work */
790 INIT_DELAYED_WORK(&socdev
->delayed_work
, close_delayed_work
);
794 if (codec_dev
->remove
)
795 codec_dev
->remove(pdev
);
798 for (i
--; i
>= 0; i
--) {
799 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
801 cpu_dai
->remove(pdev
);
805 machine
->remove(pdev
);
810 /* removes a socdev */
811 static int soc_remove(struct platform_device
*pdev
)
814 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
815 struct snd_soc_machine
*machine
= socdev
->machine
;
816 struct snd_soc_platform
*platform
= socdev
->platform
;
817 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
819 run_delayed_work(&socdev
->delayed_work
);
821 if (platform
->remove
)
822 platform
->remove(pdev
);
824 if (codec_dev
->remove
)
825 codec_dev
->remove(pdev
);
827 for (i
= 0; i
< machine
->num_links
; i
++) {
828 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
830 cpu_dai
->remove(pdev
);
834 machine
->remove(pdev
);
839 /* ASoC platform driver */
840 static struct platform_driver soc_driver
= {
843 .owner
= THIS_MODULE
,
846 .remove
= soc_remove
,
847 .suspend
= soc_suspend
,
848 .resume
= soc_resume
,
851 /* create a new pcm */
852 static int soc_new_pcm(struct snd_soc_device
*socdev
,
853 struct snd_soc_dai_link
*dai_link
, int num
)
855 struct snd_soc_codec
*codec
= socdev
->codec
;
856 struct snd_soc_codec_dai
*codec_dai
= dai_link
->codec_dai
;
857 struct snd_soc_cpu_dai
*cpu_dai
= dai_link
->cpu_dai
;
858 struct snd_soc_pcm_runtime
*rtd
;
861 int ret
= 0, playback
= 0, capture
= 0;
863 rtd
= kzalloc(sizeof(struct snd_soc_pcm_runtime
), GFP_KERNEL
);
868 rtd
->socdev
= socdev
;
869 codec_dai
->codec
= socdev
->codec
;
871 /* check client and interface hw capabilities */
872 sprintf(new_name
, "%s %s-%s-%d", dai_link
->stream_name
, codec_dai
->name
,
873 get_dai_name(cpu_dai
->type
), num
);
875 if (codec_dai
->playback
.channels_min
)
877 if (codec_dai
->capture
.channels_min
)
880 ret
= snd_pcm_new(codec
->card
, new_name
, codec
->pcm_devs
++, playback
,
883 printk(KERN_ERR
"asoc: can't create pcm for codec %s\n",
890 pcm
->private_data
= rtd
;
891 soc_pcm_ops
.mmap
= socdev
->platform
->pcm_ops
->mmap
;
892 soc_pcm_ops
.pointer
= socdev
->platform
->pcm_ops
->pointer
;
893 soc_pcm_ops
.ioctl
= socdev
->platform
->pcm_ops
->ioctl
;
894 soc_pcm_ops
.copy
= socdev
->platform
->pcm_ops
->copy
;
895 soc_pcm_ops
.silence
= socdev
->platform
->pcm_ops
->silence
;
896 soc_pcm_ops
.ack
= socdev
->platform
->pcm_ops
->ack
;
897 soc_pcm_ops
.page
= socdev
->platform
->pcm_ops
->page
;
900 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
, &soc_pcm_ops
);
903 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
, &soc_pcm_ops
);
905 ret
= socdev
->platform
->pcm_new(codec
->card
, codec_dai
, pcm
);
907 printk(KERN_ERR
"asoc: platform pcm constructor failed\n");
912 pcm
->private_free
= socdev
->platform
->pcm_free
;
913 printk(KERN_INFO
"asoc: %s <-> %s mapping ok\n", codec_dai
->name
,
918 /* codec register dump */
919 static ssize_t
codec_reg_show(struct device
*dev
,
920 struct device_attribute
*attr
, char *buf
)
922 struct snd_soc_device
*devdata
= dev_get_drvdata(dev
);
923 struct snd_soc_codec
*codec
= devdata
->codec
;
924 int i
, step
= 1, count
= 0;
926 if (!codec
->reg_cache_size
)
929 if (codec
->reg_cache_step
)
930 step
= codec
->reg_cache_step
;
932 count
+= sprintf(buf
, "%s registers\n", codec
->name
);
933 for (i
= 0; i
< codec
->reg_cache_size
; i
+= step
)
934 count
+= sprintf(buf
+ count
, "%2x: %4x\n", i
,
935 codec
->read(codec
, i
));
939 static DEVICE_ATTR(codec_reg
, 0444, codec_reg_show
, NULL
);
942 * snd_soc_new_ac97_codec - initailise AC97 device
943 * @codec: audio codec
944 * @ops: AC97 bus operations
945 * @num: AC97 codec number
947 * Initialises AC97 codec resources for use by ad-hoc devices only.
949 int snd_soc_new_ac97_codec(struct snd_soc_codec
*codec
,
950 struct snd_ac97_bus_ops
*ops
, int num
)
952 mutex_lock(&codec
->mutex
);
954 codec
->ac97
= kzalloc(sizeof(struct snd_ac97
), GFP_KERNEL
);
955 if (codec
->ac97
== NULL
) {
956 mutex_unlock(&codec
->mutex
);
960 codec
->ac97
->bus
= kzalloc(sizeof(struct snd_ac97_bus
), GFP_KERNEL
);
961 if (codec
->ac97
->bus
== NULL
) {
964 mutex_unlock(&codec
->mutex
);
968 codec
->ac97
->bus
->ops
= ops
;
969 codec
->ac97
->num
= num
;
970 mutex_unlock(&codec
->mutex
);
973 EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec
);
976 * snd_soc_free_ac97_codec - free AC97 codec device
977 * @codec: audio codec
979 * Frees AC97 codec device resources.
981 void snd_soc_free_ac97_codec(struct snd_soc_codec
*codec
)
983 mutex_lock(&codec
->mutex
);
984 kfree(codec
->ac97
->bus
);
987 mutex_unlock(&codec
->mutex
);
989 EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec
);
992 * snd_soc_update_bits - update codec register bits
993 * @codec: audio codec
994 * @reg: codec register
995 * @mask: register mask
998 * Writes new register value.
1000 * Returns 1 for change else 0.
1002 int snd_soc_update_bits(struct snd_soc_codec
*codec
, unsigned short reg
,
1003 unsigned short mask
, unsigned short value
)
1006 unsigned short old
, new;
1008 mutex_lock(&io_mutex
);
1009 old
= snd_soc_read(codec
, reg
);
1010 new = (old
& ~mask
) | value
;
1011 change
= old
!= new;
1013 snd_soc_write(codec
, reg
, new);
1015 mutex_unlock(&io_mutex
);
1018 EXPORT_SYMBOL_GPL(snd_soc_update_bits
);
1021 * snd_soc_test_bits - test register for change
1022 * @codec: audio codec
1023 * @reg: codec register
1024 * @mask: register mask
1027 * Tests a register with a new value and checks if the new value is
1028 * different from the old value.
1030 * Returns 1 for change else 0.
1032 int snd_soc_test_bits(struct snd_soc_codec
*codec
, unsigned short reg
,
1033 unsigned short mask
, unsigned short value
)
1036 unsigned short old
, new;
1038 mutex_lock(&io_mutex
);
1039 old
= snd_soc_read(codec
, reg
);
1040 new = (old
& ~mask
) | value
;
1041 change
= old
!= new;
1042 mutex_unlock(&io_mutex
);
1046 EXPORT_SYMBOL_GPL(snd_soc_test_bits
);
1049 * snd_soc_new_pcms - create new sound card and pcms
1050 * @socdev: the SoC audio device
1052 * Create a new sound card based upon the codec and interface pcms.
1054 * Returns 0 for success, else error.
1056 int snd_soc_new_pcms(struct snd_soc_device
*socdev
, int idx
, const char *xid
)
1058 struct snd_soc_codec
*codec
= socdev
->codec
;
1059 struct snd_soc_machine
*machine
= socdev
->machine
;
1062 mutex_lock(&codec
->mutex
);
1064 /* register a sound card */
1065 codec
->card
= snd_card_new(idx
, xid
, codec
->owner
, 0);
1067 printk(KERN_ERR
"asoc: can't create sound card for codec %s\n",
1069 mutex_unlock(&codec
->mutex
);
1073 codec
->card
->dev
= socdev
->dev
;
1074 codec
->card
->private_data
= codec
;
1075 strncpy(codec
->card
->driver
, codec
->name
, sizeof(codec
->card
->driver
));
1077 /* create the pcms */
1078 for (i
= 0; i
< machine
->num_links
; i
++) {
1079 ret
= soc_new_pcm(socdev
, &machine
->dai_link
[i
], i
);
1081 printk(KERN_ERR
"asoc: can't create pcm %s\n",
1082 machine
->dai_link
[i
].stream_name
);
1083 mutex_unlock(&codec
->mutex
);
1088 mutex_unlock(&codec
->mutex
);
1091 EXPORT_SYMBOL_GPL(snd_soc_new_pcms
);
1094 * snd_soc_register_card - register sound card
1095 * @socdev: the SoC audio device
1097 * Register a SoC sound card. Also registers an AC97 device if the
1098 * codec is AC97 for ad hoc devices.
1100 * Returns 0 for success, else error.
1102 int snd_soc_register_card(struct snd_soc_device
*socdev
)
1104 struct snd_soc_codec
*codec
= socdev
->codec
;
1105 struct snd_soc_machine
*machine
= socdev
->machine
;
1106 int ret
= 0, i
, ac97
= 0, err
= 0;
1108 for (i
= 0; i
< machine
->num_links
; i
++) {
1109 if (socdev
->machine
->dai_link
[i
].init
) {
1110 err
= socdev
->machine
->dai_link
[i
].init(codec
);
1112 printk(KERN_ERR
"asoc: failed to init %s\n",
1113 socdev
->machine
->dai_link
[i
].stream_name
);
1117 if (socdev
->machine
->dai_link
[i
].codec_dai
->type
==
1118 SND_SOC_DAI_AC97_BUS
)
1121 snprintf(codec
->card
->shortname
, sizeof(codec
->card
->shortname
),
1122 "%s", machine
->name
);
1123 snprintf(codec
->card
->longname
, sizeof(codec
->card
->longname
),
1124 "%s (%s)", machine
->name
, codec
->name
);
1126 ret
= snd_card_register(codec
->card
);
1128 printk(KERN_ERR
"asoc: failed to register soundcard for %s\n",
1133 mutex_lock(&codec
->mutex
);
1134 #ifdef CONFIG_SND_SOC_AC97_BUS
1136 ret
= soc_ac97_dev_register(codec
);
1138 printk(KERN_ERR
"asoc: AC97 device register failed\n");
1139 snd_card_free(codec
->card
);
1140 mutex_unlock(&codec
->mutex
);
1146 err
= snd_soc_dapm_sys_add(socdev
->dev
);
1148 printk(KERN_WARNING
"asoc: failed to add dapm sysfs entries\n");
1150 err
= device_create_file(socdev
->dev
, &dev_attr_codec_reg
);
1152 printk(KERN_WARNING
"asoc: failed to add codec sysfs files\n");
1154 mutex_unlock(&codec
->mutex
);
1159 EXPORT_SYMBOL_GPL(snd_soc_register_card
);
1162 * snd_soc_free_pcms - free sound card and pcms
1163 * @socdev: the SoC audio device
1165 * Frees sound card and pcms associated with the socdev.
1166 * Also unregister the codec if it is an AC97 device.
1168 void snd_soc_free_pcms(struct snd_soc_device
*socdev
)
1170 struct snd_soc_codec
*codec
= socdev
->codec
;
1171 #ifdef CONFIG_SND_SOC_AC97_BUS
1172 struct snd_soc_codec_dai
*codec_dai
;
1176 mutex_lock(&codec
->mutex
);
1177 #ifdef CONFIG_SND_SOC_AC97_BUS
1178 for (i
= 0; i
< codec
->num_dai
; i
++) {
1179 codec_dai
= &codec
->dai
[i
];
1180 if (codec_dai
->type
== SND_SOC_DAI_AC97_BUS
&& codec
->ac97
) {
1181 soc_ac97_dev_unregister(codec
);
1189 snd_card_free(codec
->card
);
1190 device_remove_file(socdev
->dev
, &dev_attr_codec_reg
);
1191 mutex_unlock(&codec
->mutex
);
1193 EXPORT_SYMBOL_GPL(snd_soc_free_pcms
);
1196 * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
1197 * @substream: the pcm substream
1198 * @hw: the hardware parameters
1200 * Sets the substream runtime hardware parameters.
1202 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream
*substream
,
1203 const struct snd_pcm_hardware
*hw
)
1205 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
1206 runtime
->hw
.info
= hw
->info
;
1207 runtime
->hw
.formats
= hw
->formats
;
1208 runtime
->hw
.period_bytes_min
= hw
->period_bytes_min
;
1209 runtime
->hw
.period_bytes_max
= hw
->period_bytes_max
;
1210 runtime
->hw
.periods_min
= hw
->periods_min
;
1211 runtime
->hw
.periods_max
= hw
->periods_max
;
1212 runtime
->hw
.buffer_bytes_max
= hw
->buffer_bytes_max
;
1213 runtime
->hw
.fifo_size
= hw
->fifo_size
;
1216 EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams
);
1219 * snd_soc_cnew - create new control
1220 * @_template: control template
1221 * @data: control private data
1222 * @lnng_name: control long name
1224 * Create a new mixer control from a template control.
1226 * Returns 0 for success, else error.
1228 struct snd_kcontrol
*snd_soc_cnew(const struct snd_kcontrol_new
*_template
,
1229 void *data
, char *long_name
)
1231 struct snd_kcontrol_new
template;
1233 memcpy(&template, _template
, sizeof(template));
1235 template.name
= long_name
;
1238 return snd_ctl_new1(&template, data
);
1240 EXPORT_SYMBOL_GPL(snd_soc_cnew
);
1243 * snd_soc_info_enum_double - enumerated double mixer info callback
1244 * @kcontrol: mixer control
1245 * @uinfo: control element information
1247 * Callback to provide information about a double enumerated
1250 * Returns 0 for success.
1252 int snd_soc_info_enum_double(struct snd_kcontrol
*kcontrol
,
1253 struct snd_ctl_elem_info
*uinfo
)
1255 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1257 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
1258 uinfo
->count
= e
->shift_l
== e
->shift_r
? 1 : 2;
1259 uinfo
->value
.enumerated
.items
= e
->mask
;
1261 if (uinfo
->value
.enumerated
.item
> e
->mask
- 1)
1262 uinfo
->value
.enumerated
.item
= e
->mask
- 1;
1263 strcpy(uinfo
->value
.enumerated
.name
,
1264 e
->texts
[uinfo
->value
.enumerated
.item
]);
1267 EXPORT_SYMBOL_GPL(snd_soc_info_enum_double
);
1270 * snd_soc_get_enum_double - enumerated double mixer get callback
1271 * @kcontrol: mixer control
1272 * @uinfo: control element information
1274 * Callback to get the value of a double enumerated mixer.
1276 * Returns 0 for success.
1278 int snd_soc_get_enum_double(struct snd_kcontrol
*kcontrol
,
1279 struct snd_ctl_elem_value
*ucontrol
)
1281 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1282 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1283 unsigned short val
, bitmask
;
1285 for (bitmask
= 1; bitmask
< e
->mask
; bitmask
<<= 1)
1287 val
= snd_soc_read(codec
, e
->reg
);
1288 ucontrol
->value
.enumerated
.item
[0]
1289 = (val
>> e
->shift_l
) & (bitmask
- 1);
1290 if (e
->shift_l
!= e
->shift_r
)
1291 ucontrol
->value
.enumerated
.item
[1] =
1292 (val
>> e
->shift_r
) & (bitmask
- 1);
1296 EXPORT_SYMBOL_GPL(snd_soc_get_enum_double
);
1299 * snd_soc_put_enum_double - enumerated double mixer put callback
1300 * @kcontrol: mixer control
1301 * @uinfo: control element information
1303 * Callback to set the value of a double enumerated mixer.
1305 * Returns 0 for success.
1307 int snd_soc_put_enum_double(struct snd_kcontrol
*kcontrol
,
1308 struct snd_ctl_elem_value
*ucontrol
)
1310 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1311 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1313 unsigned short mask
, bitmask
;
1315 for (bitmask
= 1; bitmask
< e
->mask
; bitmask
<<= 1)
1317 if (ucontrol
->value
.enumerated
.item
[0] > e
->mask
- 1)
1319 val
= ucontrol
->value
.enumerated
.item
[0] << e
->shift_l
;
1320 mask
= (bitmask
- 1) << e
->shift_l
;
1321 if (e
->shift_l
!= e
->shift_r
) {
1322 if (ucontrol
->value
.enumerated
.item
[1] > e
->mask
- 1)
1324 val
|= ucontrol
->value
.enumerated
.item
[1] << e
->shift_r
;
1325 mask
|= (bitmask
- 1) << e
->shift_r
;
1328 return snd_soc_update_bits(codec
, e
->reg
, mask
, val
);
1330 EXPORT_SYMBOL_GPL(snd_soc_put_enum_double
);
1333 * snd_soc_info_enum_ext - external enumerated single mixer info callback
1334 * @kcontrol: mixer control
1335 * @uinfo: control element information
1337 * Callback to provide information about an external enumerated
1340 * Returns 0 for success.
1342 int snd_soc_info_enum_ext(struct snd_kcontrol
*kcontrol
,
1343 struct snd_ctl_elem_info
*uinfo
)
1345 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1347 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
1349 uinfo
->value
.enumerated
.items
= e
->mask
;
1351 if (uinfo
->value
.enumerated
.item
> e
->mask
- 1)
1352 uinfo
->value
.enumerated
.item
= e
->mask
- 1;
1353 strcpy(uinfo
->value
.enumerated
.name
,
1354 e
->texts
[uinfo
->value
.enumerated
.item
]);
1357 EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext
);
1360 * snd_soc_info_volsw_ext - external single mixer info callback
1361 * @kcontrol: mixer control
1362 * @uinfo: control element information
1364 * Callback to provide information about a single external mixer control.
1366 * Returns 0 for success.
1368 int snd_soc_info_volsw_ext(struct snd_kcontrol
*kcontrol
,
1369 struct snd_ctl_elem_info
*uinfo
)
1371 int max
= kcontrol
->private_value
;
1374 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1376 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1379 uinfo
->value
.integer
.min
= 0;
1380 uinfo
->value
.integer
.max
= max
;
1383 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext
);
1386 * snd_soc_info_volsw - single mixer info callback
1387 * @kcontrol: mixer control
1388 * @uinfo: control element information
1390 * Callback to provide information about a single mixer control.
1392 * Returns 0 for success.
1394 int snd_soc_info_volsw(struct snd_kcontrol
*kcontrol
,
1395 struct snd_ctl_elem_info
*uinfo
)
1397 int max
= (kcontrol
->private_value
>> 16) & 0xff;
1398 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1399 int rshift
= (kcontrol
->private_value
>> 12) & 0x0f;
1402 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1404 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1406 uinfo
->count
= shift
== rshift
? 1 : 2;
1407 uinfo
->value
.integer
.min
= 0;
1408 uinfo
->value
.integer
.max
= max
;
1411 EXPORT_SYMBOL_GPL(snd_soc_info_volsw
);
1414 * snd_soc_get_volsw - single mixer get callback
1415 * @kcontrol: mixer control
1416 * @uinfo: control element information
1418 * Callback to get the value of a single mixer control.
1420 * Returns 0 for success.
1422 int snd_soc_get_volsw(struct snd_kcontrol
*kcontrol
,
1423 struct snd_ctl_elem_value
*ucontrol
)
1425 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1426 int reg
= kcontrol
->private_value
& 0xff;
1427 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1428 int rshift
= (kcontrol
->private_value
>> 12) & 0x0f;
1429 int max
= (kcontrol
->private_value
>> 16) & 0xff;
1430 int mask
= (1 << fls(max
)) - 1;
1431 int invert
= (kcontrol
->private_value
>> 24) & 0x01;
1433 ucontrol
->value
.integer
.value
[0] =
1434 (snd_soc_read(codec
, reg
) >> shift
) & mask
;
1435 if (shift
!= rshift
)
1436 ucontrol
->value
.integer
.value
[1] =
1437 (snd_soc_read(codec
, reg
) >> rshift
) & mask
;
1439 ucontrol
->value
.integer
.value
[0] =
1440 max
- ucontrol
->value
.integer
.value
[0];
1441 if (shift
!= rshift
)
1442 ucontrol
->value
.integer
.value
[1] =
1443 max
- ucontrol
->value
.integer
.value
[1];
1448 EXPORT_SYMBOL_GPL(snd_soc_get_volsw
);
1451 * snd_soc_put_volsw - single mixer put callback
1452 * @kcontrol: mixer control
1453 * @uinfo: control element information
1455 * Callback to set the value of a single mixer control.
1457 * Returns 0 for success.
1459 int snd_soc_put_volsw(struct snd_kcontrol
*kcontrol
,
1460 struct snd_ctl_elem_value
*ucontrol
)
1462 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1463 int reg
= kcontrol
->private_value
& 0xff;
1464 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1465 int rshift
= (kcontrol
->private_value
>> 12) & 0x0f;
1466 int max
= (kcontrol
->private_value
>> 16) & 0xff;
1467 int mask
= (1 << fls(max
)) - 1;
1468 int invert
= (kcontrol
->private_value
>> 24) & 0x01;
1469 unsigned short val
, val2
, val_mask
;
1471 val
= (ucontrol
->value
.integer
.value
[0] & mask
);
1474 val_mask
= mask
<< shift
;
1476 if (shift
!= rshift
) {
1477 val2
= (ucontrol
->value
.integer
.value
[1] & mask
);
1480 val_mask
|= mask
<< rshift
;
1481 val
|= val2
<< rshift
;
1483 return snd_soc_update_bits(codec
, reg
, val_mask
, val
);
1485 EXPORT_SYMBOL_GPL(snd_soc_put_volsw
);
1488 * snd_soc_info_volsw_2r - double mixer info callback
1489 * @kcontrol: mixer control
1490 * @uinfo: control element information
1492 * Callback to provide information about a double mixer control that
1493 * spans 2 codec registers.
1495 * Returns 0 for success.
1497 int snd_soc_info_volsw_2r(struct snd_kcontrol
*kcontrol
,
1498 struct snd_ctl_elem_info
*uinfo
)
1500 int max
= (kcontrol
->private_value
>> 12) & 0xff;
1503 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1505 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1508 uinfo
->value
.integer
.min
= 0;
1509 uinfo
->value
.integer
.max
= max
;
1512 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r
);
1515 * snd_soc_get_volsw_2r - double mixer get callback
1516 * @kcontrol: mixer control
1517 * @uinfo: control element information
1519 * Callback to get the value of a double mixer control that spans 2 registers.
1521 * Returns 0 for success.
1523 int snd_soc_get_volsw_2r(struct snd_kcontrol
*kcontrol
,
1524 struct snd_ctl_elem_value
*ucontrol
)
1526 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1527 int reg
= kcontrol
->private_value
& 0xff;
1528 int reg2
= (kcontrol
->private_value
>> 24) & 0xff;
1529 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1530 int max
= (kcontrol
->private_value
>> 12) & 0xff;
1531 int mask
= (1<<fls(max
))-1;
1532 int invert
= (kcontrol
->private_value
>> 20) & 0x01;
1534 ucontrol
->value
.integer
.value
[0] =
1535 (snd_soc_read(codec
, reg
) >> shift
) & mask
;
1536 ucontrol
->value
.integer
.value
[1] =
1537 (snd_soc_read(codec
, reg2
) >> shift
) & mask
;
1539 ucontrol
->value
.integer
.value
[0] =
1540 max
- ucontrol
->value
.integer
.value
[0];
1541 ucontrol
->value
.integer
.value
[1] =
1542 max
- ucontrol
->value
.integer
.value
[1];
1547 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r
);
1550 * snd_soc_put_volsw_2r - double mixer set callback
1551 * @kcontrol: mixer control
1552 * @uinfo: control element information
1554 * Callback to set the value of a double mixer control that spans 2 registers.
1556 * Returns 0 for success.
1558 int snd_soc_put_volsw_2r(struct snd_kcontrol
*kcontrol
,
1559 struct snd_ctl_elem_value
*ucontrol
)
1561 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1562 int reg
= kcontrol
->private_value
& 0xff;
1563 int reg2
= (kcontrol
->private_value
>> 24) & 0xff;
1564 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1565 int max
= (kcontrol
->private_value
>> 12) & 0xff;
1566 int mask
= (1 << fls(max
)) - 1;
1567 int invert
= (kcontrol
->private_value
>> 20) & 0x01;
1569 unsigned short val
, val2
, val_mask
;
1571 val_mask
= mask
<< shift
;
1572 val
= (ucontrol
->value
.integer
.value
[0] & mask
);
1573 val2
= (ucontrol
->value
.integer
.value
[1] & mask
);
1581 val2
= val2
<< shift
;
1583 err
= snd_soc_update_bits(codec
, reg
, val_mask
, val
);
1587 err
= snd_soc_update_bits(codec
, reg2
, val_mask
, val2
);
1590 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r
);
1592 static int __devinit
snd_soc_init(void)
1594 printk(KERN_INFO
"ASoC version %s\n", SND_SOC_VERSION
);
1595 return platform_driver_register(&soc_driver
);
1598 static void snd_soc_exit(void)
1600 platform_driver_unregister(&soc_driver
);
1603 module_init(snd_soc_init
);
1604 module_exit(snd_soc_exit
);
1606 /* Module information */
1607 MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
1608 MODULE_DESCRIPTION("ALSA SoC Core");
1609 MODULE_LICENSE("GPL");
1610 MODULE_ALIAS("platform:soc-audio");