2 * soc-core.c -- ALSA SoC Audio Layer
4 * Copyright 2005 Wolfson Microelectronics PLC.
5 * Copyright 2005 Openedhand Ltd.
7 * Author: Liam Girdwood <lrg@slimlogic.co.uk>
8 * with code, comments and ideas from :-
9 * Richard Purdie <richard@openedhand.com>
11 * This program is free software; you can redistribute it and/or modify it
12 * under the terms of the GNU General Public License as published by the
13 * Free Software Foundation; either version 2 of the License, or (at your
14 * option) any later version.
17 * o Add hw rules to enforce rates, etc.
18 * o More testing with other codecs/machines.
19 * o Add more codecs and platforms to ensure good API coverage.
20 * o Support TDM on PCM and I2S
23 #include <linux/module.h>
24 #include <linux/moduleparam.h>
25 #include <linux/init.h>
26 #include <linux/delay.h>
28 #include <linux/bitops.h>
29 #include <linux/debugfs.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>
38 static DEFINE_MUTEX(pcm_mutex
);
39 static DEFINE_MUTEX(io_mutex
);
40 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq
);
43 * This is a timeout to do a DAPM powerdown after a stream is closed().
44 * It can be used to eliminate pops between different playback streams, e.g.
45 * between two audio tracks.
47 static int pmdown_time
= 5000;
48 module_param(pmdown_time
, int, 0);
49 MODULE_PARM_DESC(pmdown_time
, "DAPM stream powerdown time (msecs)");
52 * This function forces any delayed work to be queued and run.
54 static int run_delayed_work(struct delayed_work
*dwork
)
58 /* cancel any work waiting to be queued. */
59 ret
= cancel_delayed_work(dwork
);
61 /* if there was any work waiting then we run it now and
62 * wait for it's completion */
64 schedule_delayed_work(dwork
, 0);
65 flush_scheduled_work();
70 #ifdef CONFIG_SND_SOC_AC97_BUS
71 /* unregister ac97 codec */
72 static int soc_ac97_dev_unregister(struct snd_soc_codec
*codec
)
74 if (codec
->ac97
->dev
.bus
)
75 device_unregister(&codec
->ac97
->dev
);
79 /* stop no dev release warning */
80 static void soc_ac97_device_release(struct device
*dev
){}
82 /* register ac97 codec to bus */
83 static int soc_ac97_dev_register(struct snd_soc_codec
*codec
)
87 codec
->ac97
->dev
.bus
= &ac97_bus_type
;
88 codec
->ac97
->dev
.parent
= NULL
;
89 codec
->ac97
->dev
.release
= soc_ac97_device_release
;
91 dev_set_name(&codec
->ac97
->dev
, "%d-%d:%s",
92 codec
->card
->number
, 0, codec
->name
);
93 err
= device_register(&codec
->ac97
->dev
);
95 snd_printk(KERN_ERR
"Can't register ac97 bus\n");
96 codec
->ac97
->dev
.bus
= NULL
;
104 * Called by ALSA when a PCM substream is opened, the runtime->hw record is
105 * then initialized and any private data can be allocated. This also calls
106 * startup for the cpu DAI, platform, machine and codec DAI.
108 static int soc_pcm_open(struct snd_pcm_substream
*substream
)
110 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
111 struct snd_soc_device
*socdev
= rtd
->socdev
;
112 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
113 struct snd_soc_dai_link
*machine
= rtd
->dai
;
114 struct snd_soc_platform
*platform
= socdev
->platform
;
115 struct snd_soc_dai
*cpu_dai
= machine
->cpu_dai
;
116 struct snd_soc_dai
*codec_dai
= machine
->codec_dai
;
119 mutex_lock(&pcm_mutex
);
121 /* startup the audio subsystem */
122 if (cpu_dai
->ops
.startup
) {
123 ret
= cpu_dai
->ops
.startup(substream
, cpu_dai
);
125 printk(KERN_ERR
"asoc: can't open interface %s\n",
131 if (platform
->pcm_ops
->open
) {
132 ret
= platform
->pcm_ops
->open(substream
);
134 printk(KERN_ERR
"asoc: can't open platform %s\n", platform
->name
);
139 if (codec_dai
->ops
.startup
) {
140 ret
= codec_dai
->ops
.startup(substream
, codec_dai
);
142 printk(KERN_ERR
"asoc: can't open codec %s\n",
148 if (machine
->ops
&& machine
->ops
->startup
) {
149 ret
= machine
->ops
->startup(substream
);
151 printk(KERN_ERR
"asoc: %s startup failed\n", machine
->name
);
156 /* Check that the codec and cpu DAI's are compatible */
157 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) {
158 runtime
->hw
.rate_min
=
159 max(codec_dai
->playback
.rate_min
,
160 cpu_dai
->playback
.rate_min
);
161 runtime
->hw
.rate_max
=
162 min(codec_dai
->playback
.rate_max
,
163 cpu_dai
->playback
.rate_max
);
164 runtime
->hw
.channels_min
=
165 max(codec_dai
->playback
.channels_min
,
166 cpu_dai
->playback
.channels_min
);
167 runtime
->hw
.channels_max
=
168 min(codec_dai
->playback
.channels_max
,
169 cpu_dai
->playback
.channels_max
);
170 runtime
->hw
.formats
=
171 codec_dai
->playback
.formats
& cpu_dai
->playback
.formats
;
173 codec_dai
->playback
.rates
& cpu_dai
->playback
.rates
;
175 runtime
->hw
.rate_min
=
176 max(codec_dai
->capture
.rate_min
,
177 cpu_dai
->capture
.rate_min
);
178 runtime
->hw
.rate_max
=
179 min(codec_dai
->capture
.rate_max
,
180 cpu_dai
->capture
.rate_max
);
181 runtime
->hw
.channels_min
=
182 max(codec_dai
->capture
.channels_min
,
183 cpu_dai
->capture
.channels_min
);
184 runtime
->hw
.channels_max
=
185 min(codec_dai
->capture
.channels_max
,
186 cpu_dai
->capture
.channels_max
);
187 runtime
->hw
.formats
=
188 codec_dai
->capture
.formats
& cpu_dai
->capture
.formats
;
190 codec_dai
->capture
.rates
& cpu_dai
->capture
.rates
;
193 snd_pcm_limit_hw_rates(runtime
);
194 if (!runtime
->hw
.rates
) {
195 printk(KERN_ERR
"asoc: %s <-> %s No matching rates\n",
196 codec_dai
->name
, cpu_dai
->name
);
199 if (!runtime
->hw
.formats
) {
200 printk(KERN_ERR
"asoc: %s <-> %s No matching formats\n",
201 codec_dai
->name
, cpu_dai
->name
);
204 if (!runtime
->hw
.channels_min
|| !runtime
->hw
.channels_max
) {
205 printk(KERN_ERR
"asoc: %s <-> %s No matching channels\n",
206 codec_dai
->name
, cpu_dai
->name
);
210 pr_debug("asoc: %s <-> %s info:\n", codec_dai
->name
, cpu_dai
->name
);
211 pr_debug("asoc: rate mask 0x%x\n", runtime
->hw
.rates
);
212 pr_debug("asoc: min ch %d max ch %d\n", runtime
->hw
.channels_min
,
213 runtime
->hw
.channels_max
);
214 pr_debug("asoc: min rate %d max rate %d\n", runtime
->hw
.rate_min
,
215 runtime
->hw
.rate_max
);
217 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
218 cpu_dai
->playback
.active
= codec_dai
->playback
.active
= 1;
220 cpu_dai
->capture
.active
= codec_dai
->capture
.active
= 1;
221 cpu_dai
->active
= codec_dai
->active
= 1;
222 cpu_dai
->runtime
= runtime
;
223 socdev
->codec
->active
++;
224 mutex_unlock(&pcm_mutex
);
228 if (machine
->ops
&& machine
->ops
->shutdown
)
229 machine
->ops
->shutdown(substream
);
232 if (platform
->pcm_ops
->close
)
233 platform
->pcm_ops
->close(substream
);
236 if (cpu_dai
->ops
.shutdown
)
237 cpu_dai
->ops
.shutdown(substream
, cpu_dai
);
239 mutex_unlock(&pcm_mutex
);
244 * Power down the audio subsystem pmdown_time msecs after close is called.
245 * This is to ensure there are no pops or clicks in between any music tracks
246 * due to DAPM power cycling.
248 static void close_delayed_work(struct work_struct
*work
)
250 struct snd_soc_device
*socdev
=
251 container_of(work
, struct snd_soc_device
, delayed_work
.work
);
252 struct snd_soc_codec
*codec
= socdev
->codec
;
253 struct snd_soc_dai
*codec_dai
;
256 mutex_lock(&pcm_mutex
);
257 for (i
= 0; i
< codec
->num_dai
; i
++) {
258 codec_dai
= &codec
->dai
[i
];
260 pr_debug("pop wq checking: %s status: %s waiting: %s\n",
261 codec_dai
->playback
.stream_name
,
262 codec_dai
->playback
.active
? "active" : "inactive",
263 codec_dai
->pop_wait
? "yes" : "no");
265 /* are we waiting on this codec DAI stream */
266 if (codec_dai
->pop_wait
== 1) {
268 /* Reduce power if no longer active */
269 if (codec
->active
== 0) {
270 pr_debug("pop wq D1 %s %s\n", codec
->name
,
271 codec_dai
->playback
.stream_name
);
272 snd_soc_dapm_set_bias_level(socdev
,
273 SND_SOC_BIAS_PREPARE
);
276 codec_dai
->pop_wait
= 0;
277 snd_soc_dapm_stream_event(codec
,
278 codec_dai
->playback
.stream_name
,
279 SND_SOC_DAPM_STREAM_STOP
);
281 /* Fall into standby if no longer active */
282 if (codec
->active
== 0) {
283 pr_debug("pop wq D3 %s %s\n", codec
->name
,
284 codec_dai
->playback
.stream_name
);
285 snd_soc_dapm_set_bias_level(socdev
,
286 SND_SOC_BIAS_STANDBY
);
290 mutex_unlock(&pcm_mutex
);
294 * Called by ALSA when a PCM substream is closed. Private data can be
295 * freed here. The cpu DAI, codec DAI, machine and platform are also
298 static int soc_codec_close(struct snd_pcm_substream
*substream
)
300 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
301 struct snd_soc_device
*socdev
= rtd
->socdev
;
302 struct snd_soc_dai_link
*machine
= rtd
->dai
;
303 struct snd_soc_platform
*platform
= socdev
->platform
;
304 struct snd_soc_dai
*cpu_dai
= machine
->cpu_dai
;
305 struct snd_soc_dai
*codec_dai
= machine
->codec_dai
;
306 struct snd_soc_codec
*codec
= socdev
->codec
;
308 mutex_lock(&pcm_mutex
);
310 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
311 cpu_dai
->playback
.active
= codec_dai
->playback
.active
= 0;
313 cpu_dai
->capture
.active
= codec_dai
->capture
.active
= 0;
315 if (codec_dai
->playback
.active
== 0 &&
316 codec_dai
->capture
.active
== 0) {
317 cpu_dai
->active
= codec_dai
->active
= 0;
321 /* Muting the DAC suppresses artifacts caused during digital
322 * shutdown, for example from stopping clocks.
324 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
325 snd_soc_dai_digital_mute(codec_dai
, 1);
327 if (cpu_dai
->ops
.shutdown
)
328 cpu_dai
->ops
.shutdown(substream
, cpu_dai
);
330 if (codec_dai
->ops
.shutdown
)
331 codec_dai
->ops
.shutdown(substream
, codec_dai
);
333 if (machine
->ops
&& machine
->ops
->shutdown
)
334 machine
->ops
->shutdown(substream
);
336 if (platform
->pcm_ops
->close
)
337 platform
->pcm_ops
->close(substream
);
338 cpu_dai
->runtime
= NULL
;
340 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) {
341 /* start delayed pop wq here for playback streams */
342 codec_dai
->pop_wait
= 1;
343 schedule_delayed_work(&socdev
->delayed_work
,
344 msecs_to_jiffies(pmdown_time
));
346 /* capture streams can be powered down now */
347 snd_soc_dapm_stream_event(codec
,
348 codec_dai
->capture
.stream_name
,
349 SND_SOC_DAPM_STREAM_STOP
);
351 if (codec
->active
== 0 && codec_dai
->pop_wait
== 0)
352 snd_soc_dapm_set_bias_level(socdev
,
353 SND_SOC_BIAS_STANDBY
);
356 mutex_unlock(&pcm_mutex
);
361 * Called by ALSA when the PCM substream is prepared, can set format, sample
362 * rate, etc. This function is non atomic and can be called multiple times,
363 * it can refer to the runtime info.
365 static int soc_pcm_prepare(struct snd_pcm_substream
*substream
)
367 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
368 struct snd_soc_device
*socdev
= rtd
->socdev
;
369 struct snd_soc_dai_link
*machine
= rtd
->dai
;
370 struct snd_soc_platform
*platform
= socdev
->platform
;
371 struct snd_soc_dai
*cpu_dai
= machine
->cpu_dai
;
372 struct snd_soc_dai
*codec_dai
= machine
->codec_dai
;
373 struct snd_soc_codec
*codec
= socdev
->codec
;
376 mutex_lock(&pcm_mutex
);
378 if (machine
->ops
&& machine
->ops
->prepare
) {
379 ret
= machine
->ops
->prepare(substream
);
381 printk(KERN_ERR
"asoc: machine prepare error\n");
386 if (platform
->pcm_ops
->prepare
) {
387 ret
= platform
->pcm_ops
->prepare(substream
);
389 printk(KERN_ERR
"asoc: platform prepare error\n");
394 if (codec_dai
->ops
.prepare
) {
395 ret
= codec_dai
->ops
.prepare(substream
, codec_dai
);
397 printk(KERN_ERR
"asoc: codec DAI prepare error\n");
402 if (cpu_dai
->ops
.prepare
) {
403 ret
= cpu_dai
->ops
.prepare(substream
, cpu_dai
);
405 printk(KERN_ERR
"asoc: cpu DAI prepare error\n");
410 /* cancel any delayed stream shutdown that is pending */
411 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
&&
412 codec_dai
->pop_wait
) {
413 codec_dai
->pop_wait
= 0;
414 cancel_delayed_work(&socdev
->delayed_work
);
417 /* do we need to power up codec */
418 if (codec
->bias_level
!= SND_SOC_BIAS_ON
) {
419 snd_soc_dapm_set_bias_level(socdev
,
420 SND_SOC_BIAS_PREPARE
);
422 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
423 snd_soc_dapm_stream_event(codec
,
424 codec_dai
->playback
.stream_name
,
425 SND_SOC_DAPM_STREAM_START
);
427 snd_soc_dapm_stream_event(codec
,
428 codec_dai
->capture
.stream_name
,
429 SND_SOC_DAPM_STREAM_START
);
431 snd_soc_dapm_set_bias_level(socdev
, SND_SOC_BIAS_ON
);
432 snd_soc_dai_digital_mute(codec_dai
, 0);
435 /* codec already powered - power on widgets */
436 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
437 snd_soc_dapm_stream_event(codec
,
438 codec_dai
->playback
.stream_name
,
439 SND_SOC_DAPM_STREAM_START
);
441 snd_soc_dapm_stream_event(codec
,
442 codec_dai
->capture
.stream_name
,
443 SND_SOC_DAPM_STREAM_START
);
445 snd_soc_dai_digital_mute(codec_dai
, 0);
449 mutex_unlock(&pcm_mutex
);
454 * Called by ALSA when the hardware params are set by application. This
455 * function can also be called multiple times and can allocate buffers
456 * (using snd_pcm_lib_* ). It's non-atomic.
458 static int soc_pcm_hw_params(struct snd_pcm_substream
*substream
,
459 struct snd_pcm_hw_params
*params
)
461 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
462 struct snd_soc_device
*socdev
= rtd
->socdev
;
463 struct snd_soc_dai_link
*machine
= rtd
->dai
;
464 struct snd_soc_platform
*platform
= socdev
->platform
;
465 struct snd_soc_dai
*cpu_dai
= machine
->cpu_dai
;
466 struct snd_soc_dai
*codec_dai
= machine
->codec_dai
;
469 mutex_lock(&pcm_mutex
);
471 if (machine
->ops
&& machine
->ops
->hw_params
) {
472 ret
= machine
->ops
->hw_params(substream
, params
);
474 printk(KERN_ERR
"asoc: machine hw_params failed\n");
479 if (codec_dai
->ops
.hw_params
) {
480 ret
= codec_dai
->ops
.hw_params(substream
, params
, codec_dai
);
482 printk(KERN_ERR
"asoc: can't set codec %s hw params\n",
488 if (cpu_dai
->ops
.hw_params
) {
489 ret
= cpu_dai
->ops
.hw_params(substream
, params
, cpu_dai
);
491 printk(KERN_ERR
"asoc: interface %s hw params failed\n",
497 if (platform
->pcm_ops
->hw_params
) {
498 ret
= platform
->pcm_ops
->hw_params(substream
, params
);
500 printk(KERN_ERR
"asoc: platform %s hw params failed\n",
507 mutex_unlock(&pcm_mutex
);
511 if (cpu_dai
->ops
.hw_free
)
512 cpu_dai
->ops
.hw_free(substream
, cpu_dai
);
515 if (codec_dai
->ops
.hw_free
)
516 codec_dai
->ops
.hw_free(substream
, codec_dai
);
519 if (machine
->ops
&& machine
->ops
->hw_free
)
520 machine
->ops
->hw_free(substream
);
522 mutex_unlock(&pcm_mutex
);
527 * Free's resources allocated by hw_params, can be called multiple times
529 static int soc_pcm_hw_free(struct snd_pcm_substream
*substream
)
531 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
532 struct snd_soc_device
*socdev
= rtd
->socdev
;
533 struct snd_soc_dai_link
*machine
= rtd
->dai
;
534 struct snd_soc_platform
*platform
= socdev
->platform
;
535 struct snd_soc_dai
*cpu_dai
= machine
->cpu_dai
;
536 struct snd_soc_dai
*codec_dai
= machine
->codec_dai
;
537 struct snd_soc_codec
*codec
= socdev
->codec
;
539 mutex_lock(&pcm_mutex
);
541 /* apply codec digital mute */
543 snd_soc_dai_digital_mute(codec_dai
, 1);
545 /* free any machine hw params */
546 if (machine
->ops
&& machine
->ops
->hw_free
)
547 machine
->ops
->hw_free(substream
);
549 /* free any DMA resources */
550 if (platform
->pcm_ops
->hw_free
)
551 platform
->pcm_ops
->hw_free(substream
);
553 /* now free hw params for the DAI's */
554 if (codec_dai
->ops
.hw_free
)
555 codec_dai
->ops
.hw_free(substream
, codec_dai
);
557 if (cpu_dai
->ops
.hw_free
)
558 cpu_dai
->ops
.hw_free(substream
, cpu_dai
);
560 mutex_unlock(&pcm_mutex
);
564 static int soc_pcm_trigger(struct snd_pcm_substream
*substream
, int cmd
)
566 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
567 struct snd_soc_device
*socdev
= rtd
->socdev
;
568 struct snd_soc_dai_link
*machine
= rtd
->dai
;
569 struct snd_soc_platform
*platform
= socdev
->platform
;
570 struct snd_soc_dai
*cpu_dai
= machine
->cpu_dai
;
571 struct snd_soc_dai
*codec_dai
= machine
->codec_dai
;
574 if (codec_dai
->ops
.trigger
) {
575 ret
= codec_dai
->ops
.trigger(substream
, cmd
, codec_dai
);
580 if (platform
->pcm_ops
->trigger
) {
581 ret
= platform
->pcm_ops
->trigger(substream
, cmd
);
586 if (cpu_dai
->ops
.trigger
) {
587 ret
= cpu_dai
->ops
.trigger(substream
, cmd
, cpu_dai
);
594 /* ASoC PCM operations */
595 static struct snd_pcm_ops soc_pcm_ops
= {
596 .open
= soc_pcm_open
,
597 .close
= soc_codec_close
,
598 .hw_params
= soc_pcm_hw_params
,
599 .hw_free
= soc_pcm_hw_free
,
600 .prepare
= soc_pcm_prepare
,
601 .trigger
= soc_pcm_trigger
,
605 /* powers down audio subsystem for suspend */
606 static int soc_suspend(struct platform_device
*pdev
, pm_message_t state
)
608 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
609 struct snd_soc_card
*card
= socdev
->card
;
610 struct snd_soc_platform
*platform
= socdev
->platform
;
611 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
612 struct snd_soc_codec
*codec
= socdev
->codec
;
615 /* Due to the resume being scheduled into a workqueue we could
616 * suspend before that's finished - wait for it to complete.
618 snd_power_lock(codec
->card
);
619 snd_power_wait(codec
->card
, SNDRV_CTL_POWER_D0
);
620 snd_power_unlock(codec
->card
);
622 /* we're going to block userspace touching us until resume completes */
623 snd_power_change_state(codec
->card
, SNDRV_CTL_POWER_D3hot
);
625 /* mute any active DAC's */
626 for (i
= 0; i
< card
->num_links
; i
++) {
627 struct snd_soc_dai
*dai
= card
->dai_link
[i
].codec_dai
;
628 if (dai
->ops
.digital_mute
&& dai
->playback
.active
)
629 dai
->ops
.digital_mute(dai
, 1);
632 /* suspend all pcms */
633 for (i
= 0; i
< card
->num_links
; i
++)
634 snd_pcm_suspend_all(card
->dai_link
[i
].pcm
);
636 if (card
->suspend_pre
)
637 card
->suspend_pre(pdev
, state
);
639 for (i
= 0; i
< card
->num_links
; i
++) {
640 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
641 if (cpu_dai
->suspend
&& !cpu_dai
->ac97_control
)
642 cpu_dai
->suspend(pdev
, cpu_dai
);
643 if (platform
->suspend
)
644 platform
->suspend(pdev
, cpu_dai
);
647 /* close any waiting streams and save state */
648 run_delayed_work(&socdev
->delayed_work
);
649 codec
->suspend_bias_level
= codec
->bias_level
;
651 for (i
= 0; i
< codec
->num_dai
; i
++) {
652 char *stream
= codec
->dai
[i
].playback
.stream_name
;
654 snd_soc_dapm_stream_event(codec
, stream
,
655 SND_SOC_DAPM_STREAM_SUSPEND
);
656 stream
= codec
->dai
[i
].capture
.stream_name
;
658 snd_soc_dapm_stream_event(codec
, stream
,
659 SND_SOC_DAPM_STREAM_SUSPEND
);
662 if (codec_dev
->suspend
)
663 codec_dev
->suspend(pdev
, state
);
665 for (i
= 0; i
< card
->num_links
; i
++) {
666 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
667 if (cpu_dai
->suspend
&& cpu_dai
->ac97_control
)
668 cpu_dai
->suspend(pdev
, cpu_dai
);
671 if (card
->suspend_post
)
672 card
->suspend_post(pdev
, state
);
677 /* deferred resume work, so resume can complete before we finished
678 * setting our codec back up, which can be very slow on I2C
680 static void soc_resume_deferred(struct work_struct
*work
)
682 struct snd_soc_device
*socdev
= container_of(work
,
683 struct snd_soc_device
,
684 deferred_resume_work
);
685 struct snd_soc_card
*card
= socdev
->card
;
686 struct snd_soc_platform
*platform
= socdev
->platform
;
687 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
688 struct snd_soc_codec
*codec
= socdev
->codec
;
689 struct platform_device
*pdev
= to_platform_device(socdev
->dev
);
692 /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
693 * so userspace apps are blocked from touching us
696 dev_dbg(socdev
->dev
, "starting resume work\n");
698 if (card
->resume_pre
)
699 card
->resume_pre(pdev
);
701 for (i
= 0; i
< card
->num_links
; i
++) {
702 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
703 if (cpu_dai
->resume
&& cpu_dai
->ac97_control
)
704 cpu_dai
->resume(pdev
, cpu_dai
);
707 if (codec_dev
->resume
)
708 codec_dev
->resume(pdev
);
710 for (i
= 0; i
< codec
->num_dai
; i
++) {
711 char *stream
= codec
->dai
[i
].playback
.stream_name
;
713 snd_soc_dapm_stream_event(codec
, stream
,
714 SND_SOC_DAPM_STREAM_RESUME
);
715 stream
= codec
->dai
[i
].capture
.stream_name
;
717 snd_soc_dapm_stream_event(codec
, stream
,
718 SND_SOC_DAPM_STREAM_RESUME
);
721 /* unmute any active DACs */
722 for (i
= 0; i
< card
->num_links
; i
++) {
723 struct snd_soc_dai
*dai
= card
->dai_link
[i
].codec_dai
;
724 if (dai
->ops
.digital_mute
&& dai
->playback
.active
)
725 dai
->ops
.digital_mute(dai
, 0);
728 for (i
= 0; i
< card
->num_links
; i
++) {
729 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
730 if (cpu_dai
->resume
&& !cpu_dai
->ac97_control
)
731 cpu_dai
->resume(pdev
, cpu_dai
);
732 if (platform
->resume
)
733 platform
->resume(pdev
, cpu_dai
);
736 if (card
->resume_post
)
737 card
->resume_post(pdev
);
739 dev_dbg(socdev
->dev
, "resume work completed\n");
741 /* userspace can access us now we are back as we were before */
742 snd_power_change_state(codec
->card
, SNDRV_CTL_POWER_D0
);
745 /* powers up audio subsystem after a suspend */
746 static int soc_resume(struct platform_device
*pdev
)
748 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
750 dev_dbg(socdev
->dev
, "scheduling resume work\n");
752 if (!schedule_work(&socdev
->deferred_resume_work
))
753 dev_err(socdev
->dev
, "resume work item may be lost\n");
759 #define soc_suspend NULL
760 #define soc_resume NULL
763 /* probes a new socdev */
764 static int soc_probe(struct platform_device
*pdev
)
767 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
768 struct snd_soc_card
*card
= socdev
->card
;
769 struct snd_soc_platform
*platform
= socdev
->platform
;
770 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
773 ret
= card
->probe(pdev
);
778 for (i
= 0; i
< card
->num_links
; i
++) {
779 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
780 if (cpu_dai
->probe
) {
781 ret
= cpu_dai
->probe(pdev
, cpu_dai
);
787 if (codec_dev
->probe
) {
788 ret
= codec_dev
->probe(pdev
);
793 if (platform
->probe
) {
794 ret
= platform
->probe(pdev
);
799 /* DAPM stream work */
800 INIT_DELAYED_WORK(&socdev
->delayed_work
, close_delayed_work
);
802 /* deferred resume work */
803 INIT_WORK(&socdev
->deferred_resume_work
, soc_resume_deferred
);
809 if (codec_dev
->remove
)
810 codec_dev
->remove(pdev
);
813 for (i
--; i
>= 0; i
--) {
814 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
816 cpu_dai
->remove(pdev
, cpu_dai
);
825 /* removes a socdev */
826 static int soc_remove(struct platform_device
*pdev
)
829 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
830 struct snd_soc_card
*card
= socdev
->card
;
831 struct snd_soc_platform
*platform
= socdev
->platform
;
832 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
834 run_delayed_work(&socdev
->delayed_work
);
836 if (platform
->remove
)
837 platform
->remove(pdev
);
839 if (codec_dev
->remove
)
840 codec_dev
->remove(pdev
);
842 for (i
= 0; i
< card
->num_links
; i
++) {
843 struct snd_soc_dai
*cpu_dai
= card
->dai_link
[i
].cpu_dai
;
845 cpu_dai
->remove(pdev
, cpu_dai
);
854 /* ASoC platform driver */
855 static struct platform_driver soc_driver
= {
858 .owner
= THIS_MODULE
,
861 .remove
= soc_remove
,
862 .suspend
= soc_suspend
,
863 .resume
= soc_resume
,
866 /* create a new pcm */
867 static int soc_new_pcm(struct snd_soc_device
*socdev
,
868 struct snd_soc_dai_link
*dai_link
, int num
)
870 struct snd_soc_codec
*codec
= socdev
->codec
;
871 struct snd_soc_dai
*codec_dai
= dai_link
->codec_dai
;
872 struct snd_soc_dai
*cpu_dai
= dai_link
->cpu_dai
;
873 struct snd_soc_pcm_runtime
*rtd
;
876 int ret
= 0, playback
= 0, capture
= 0;
878 rtd
= kzalloc(sizeof(struct snd_soc_pcm_runtime
), GFP_KERNEL
);
883 rtd
->socdev
= socdev
;
884 codec_dai
->codec
= socdev
->codec
;
886 /* check client and interface hw capabilities */
887 sprintf(new_name
, "%s %s-%d", dai_link
->stream_name
, codec_dai
->name
,
890 if (codec_dai
->playback
.channels_min
)
892 if (codec_dai
->capture
.channels_min
)
895 ret
= snd_pcm_new(codec
->card
, new_name
, codec
->pcm_devs
++, playback
,
898 printk(KERN_ERR
"asoc: can't create pcm for codec %s\n",
905 pcm
->private_data
= rtd
;
906 soc_pcm_ops
.mmap
= socdev
->platform
->pcm_ops
->mmap
;
907 soc_pcm_ops
.pointer
= socdev
->platform
->pcm_ops
->pointer
;
908 soc_pcm_ops
.ioctl
= socdev
->platform
->pcm_ops
->ioctl
;
909 soc_pcm_ops
.copy
= socdev
->platform
->pcm_ops
->copy
;
910 soc_pcm_ops
.silence
= socdev
->platform
->pcm_ops
->silence
;
911 soc_pcm_ops
.ack
= socdev
->platform
->pcm_ops
->ack
;
912 soc_pcm_ops
.page
= socdev
->platform
->pcm_ops
->page
;
915 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
, &soc_pcm_ops
);
918 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
, &soc_pcm_ops
);
920 ret
= socdev
->platform
->pcm_new(codec
->card
, codec_dai
, pcm
);
922 printk(KERN_ERR
"asoc: platform pcm constructor failed\n");
927 pcm
->private_free
= socdev
->platform
->pcm_free
;
928 printk(KERN_INFO
"asoc: %s <-> %s mapping ok\n", codec_dai
->name
,
933 /* codec register dump */
934 static ssize_t
soc_codec_reg_show(struct snd_soc_device
*devdata
, char *buf
)
936 struct snd_soc_codec
*codec
= devdata
->codec
;
937 int i
, step
= 1, count
= 0;
939 if (!codec
->reg_cache_size
)
942 if (codec
->reg_cache_step
)
943 step
= codec
->reg_cache_step
;
945 count
+= sprintf(buf
, "%s registers\n", codec
->name
);
946 for (i
= 0; i
< codec
->reg_cache_size
; i
+= step
) {
947 count
+= sprintf(buf
+ count
, "%2x: ", i
);
948 if (count
>= PAGE_SIZE
- 1)
951 if (codec
->display_register
)
952 count
+= codec
->display_register(codec
, buf
+ count
,
953 PAGE_SIZE
- count
, i
);
955 count
+= snprintf(buf
+ count
, PAGE_SIZE
- count
,
956 "%4x", codec
->read(codec
, i
));
958 if (count
>= PAGE_SIZE
- 1)
961 count
+= snprintf(buf
+ count
, PAGE_SIZE
- count
, "\n");
962 if (count
>= PAGE_SIZE
- 1)
966 /* Truncate count; min() would cause a warning */
967 if (count
>= PAGE_SIZE
)
968 count
= PAGE_SIZE
- 1;
972 static ssize_t
codec_reg_show(struct device
*dev
,
973 struct device_attribute
*attr
, char *buf
)
975 struct snd_soc_device
*devdata
= dev_get_drvdata(dev
);
976 return soc_codec_reg_show(devdata
, buf
);
979 static DEVICE_ATTR(codec_reg
, 0444, codec_reg_show
, NULL
);
981 #ifdef CONFIG_DEBUG_FS
982 static int codec_reg_open_file(struct inode
*inode
, struct file
*file
)
984 file
->private_data
= inode
->i_private
;
988 static ssize_t
codec_reg_read_file(struct file
*file
, char __user
*user_buf
,
989 size_t count
, loff_t
*ppos
)
992 struct snd_soc_device
*devdata
= file
->private_data
;
993 char *buf
= kmalloc(PAGE_SIZE
, GFP_KERNEL
);
996 ret
= soc_codec_reg_show(devdata
, buf
);
998 ret
= simple_read_from_buffer(user_buf
, count
, ppos
, buf
, ret
);
1003 static ssize_t
codec_reg_write_file(struct file
*file
,
1004 const char __user
*user_buf
, size_t count
, loff_t
*ppos
)
1009 unsigned long reg
, value
;
1011 struct snd_soc_device
*devdata
= file
->private_data
;
1012 struct snd_soc_codec
*codec
= devdata
->codec
;
1014 buf_size
= min(count
, (sizeof(buf
)-1));
1015 if (copy_from_user(buf
, user_buf
, buf_size
))
1019 if (codec
->reg_cache_step
)
1020 step
= codec
->reg_cache_step
;
1022 while (*start
== ' ')
1024 reg
= simple_strtoul(start
, &start
, 16);
1025 if ((reg
>= codec
->reg_cache_size
) || (reg
% step
))
1027 while (*start
== ' ')
1029 if (strict_strtoul(start
, 16, &value
))
1031 codec
->write(codec
, reg
, value
);
1035 static const struct file_operations codec_reg_fops
= {
1036 .open
= codec_reg_open_file
,
1037 .read
= codec_reg_read_file
,
1038 .write
= codec_reg_write_file
,
1041 static void soc_init_debugfs(struct snd_soc_device
*socdev
)
1043 struct dentry
*root
, *file
;
1044 struct snd_soc_codec
*codec
= socdev
->codec
;
1045 root
= debugfs_create_dir(dev_name(socdev
->dev
), NULL
);
1046 if (IS_ERR(root
) || !root
)
1049 file
= debugfs_create_file("codec_reg", 0644,
1050 root
, socdev
, &codec_reg_fops
);
1054 file
= debugfs_create_u32("dapm_pop_time", 0744,
1055 root
, &codec
->pop_time
);
1058 socdev
->debugfs_root
= root
;
1061 debugfs_remove_recursive(root
);
1063 dev_err(socdev
->dev
, "debugfs is not available\n");
1066 static void soc_cleanup_debugfs(struct snd_soc_device
*socdev
)
1068 debugfs_remove_recursive(socdev
->debugfs_root
);
1069 socdev
->debugfs_root
= NULL
;
1074 static inline void soc_init_debugfs(struct snd_soc_device
*socdev
)
1078 static inline void soc_cleanup_debugfs(struct snd_soc_device
*socdev
)
1084 * snd_soc_new_ac97_codec - initailise AC97 device
1085 * @codec: audio codec
1086 * @ops: AC97 bus operations
1087 * @num: AC97 codec number
1089 * Initialises AC97 codec resources for use by ad-hoc devices only.
1091 int snd_soc_new_ac97_codec(struct snd_soc_codec
*codec
,
1092 struct snd_ac97_bus_ops
*ops
, int num
)
1094 mutex_lock(&codec
->mutex
);
1096 codec
->ac97
= kzalloc(sizeof(struct snd_ac97
), GFP_KERNEL
);
1097 if (codec
->ac97
== NULL
) {
1098 mutex_unlock(&codec
->mutex
);
1102 codec
->ac97
->bus
= kzalloc(sizeof(struct snd_ac97_bus
), GFP_KERNEL
);
1103 if (codec
->ac97
->bus
== NULL
) {
1106 mutex_unlock(&codec
->mutex
);
1110 codec
->ac97
->bus
->ops
= ops
;
1111 codec
->ac97
->num
= num
;
1112 mutex_unlock(&codec
->mutex
);
1115 EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec
);
1118 * snd_soc_free_ac97_codec - free AC97 codec device
1119 * @codec: audio codec
1121 * Frees AC97 codec device resources.
1123 void snd_soc_free_ac97_codec(struct snd_soc_codec
*codec
)
1125 mutex_lock(&codec
->mutex
);
1126 kfree(codec
->ac97
->bus
);
1129 mutex_unlock(&codec
->mutex
);
1131 EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec
);
1134 * snd_soc_update_bits - update codec register bits
1135 * @codec: audio codec
1136 * @reg: codec register
1137 * @mask: register mask
1140 * Writes new register value.
1142 * Returns 1 for change else 0.
1144 int snd_soc_update_bits(struct snd_soc_codec
*codec
, unsigned short reg
,
1145 unsigned short mask
, unsigned short value
)
1148 unsigned short old
, new;
1150 mutex_lock(&io_mutex
);
1151 old
= snd_soc_read(codec
, reg
);
1152 new = (old
& ~mask
) | value
;
1153 change
= old
!= new;
1155 snd_soc_write(codec
, reg
, new);
1157 mutex_unlock(&io_mutex
);
1160 EXPORT_SYMBOL_GPL(snd_soc_update_bits
);
1163 * snd_soc_test_bits - test register for change
1164 * @codec: audio codec
1165 * @reg: codec register
1166 * @mask: register mask
1169 * Tests a register with a new value and checks if the new value is
1170 * different from the old value.
1172 * Returns 1 for change else 0.
1174 int snd_soc_test_bits(struct snd_soc_codec
*codec
, unsigned short reg
,
1175 unsigned short mask
, unsigned short value
)
1178 unsigned short old
, new;
1180 mutex_lock(&io_mutex
);
1181 old
= snd_soc_read(codec
, reg
);
1182 new = (old
& ~mask
) | value
;
1183 change
= old
!= new;
1184 mutex_unlock(&io_mutex
);
1188 EXPORT_SYMBOL_GPL(snd_soc_test_bits
);
1191 * snd_soc_new_pcms - create new sound card and pcms
1192 * @socdev: the SoC audio device
1194 * Create a new sound card based upon the codec and interface pcms.
1196 * Returns 0 for success, else error.
1198 int snd_soc_new_pcms(struct snd_soc_device
*socdev
, int idx
, const char *xid
)
1200 struct snd_soc_codec
*codec
= socdev
->codec
;
1201 struct snd_soc_card
*card
= socdev
->card
;
1204 mutex_lock(&codec
->mutex
);
1206 /* register a sound card */
1207 codec
->card
= snd_card_new(idx
, xid
, codec
->owner
, 0);
1209 printk(KERN_ERR
"asoc: can't create sound card for codec %s\n",
1211 mutex_unlock(&codec
->mutex
);
1215 codec
->card
->dev
= socdev
->dev
;
1216 codec
->card
->private_data
= codec
;
1217 strncpy(codec
->card
->driver
, codec
->name
, sizeof(codec
->card
->driver
));
1219 /* create the pcms */
1220 for (i
= 0; i
< card
->num_links
; i
++) {
1221 ret
= soc_new_pcm(socdev
, &card
->dai_link
[i
], i
);
1223 printk(KERN_ERR
"asoc: can't create pcm %s\n",
1224 card
->dai_link
[i
].stream_name
);
1225 mutex_unlock(&codec
->mutex
);
1230 mutex_unlock(&codec
->mutex
);
1233 EXPORT_SYMBOL_GPL(snd_soc_new_pcms
);
1236 * snd_soc_init_card - register sound card
1237 * @socdev: the SoC audio device
1239 * Register a SoC sound card. Also registers an AC97 device if the
1240 * codec is AC97 for ad hoc devices.
1242 * Returns 0 for success, else error.
1244 int snd_soc_init_card(struct snd_soc_device
*socdev
)
1246 struct snd_soc_codec
*codec
= socdev
->codec
;
1247 struct snd_soc_card
*card
= socdev
->card
;
1248 int ret
= 0, i
, ac97
= 0, err
= 0;
1250 for (i
= 0; i
< card
->num_links
; i
++) {
1251 if (card
->dai_link
[i
].init
) {
1252 err
= card
->dai_link
[i
].init(codec
);
1254 printk(KERN_ERR
"asoc: failed to init %s\n",
1255 card
->dai_link
[i
].stream_name
);
1259 if (card
->dai_link
[i
].codec_dai
->ac97_control
)
1262 snprintf(codec
->card
->shortname
, sizeof(codec
->card
->shortname
),
1264 snprintf(codec
->card
->longname
, sizeof(codec
->card
->longname
),
1265 "%s (%s)", card
->name
, codec
->name
);
1267 ret
= snd_card_register(codec
->card
);
1269 printk(KERN_ERR
"asoc: failed to register soundcard for %s\n",
1274 mutex_lock(&codec
->mutex
);
1275 #ifdef CONFIG_SND_SOC_AC97_BUS
1277 ret
= soc_ac97_dev_register(codec
);
1279 printk(KERN_ERR
"asoc: AC97 device register failed\n");
1280 snd_card_free(codec
->card
);
1281 mutex_unlock(&codec
->mutex
);
1287 err
= snd_soc_dapm_sys_add(socdev
->dev
);
1289 printk(KERN_WARNING
"asoc: failed to add dapm sysfs entries\n");
1291 err
= device_create_file(socdev
->dev
, &dev_attr_codec_reg
);
1293 printk(KERN_WARNING
"asoc: failed to add codec sysfs files\n");
1295 soc_init_debugfs(socdev
);
1296 mutex_unlock(&codec
->mutex
);
1301 EXPORT_SYMBOL_GPL(snd_soc_init_card
);
1304 * snd_soc_free_pcms - free sound card and pcms
1305 * @socdev: the SoC audio device
1307 * Frees sound card and pcms associated with the socdev.
1308 * Also unregister the codec if it is an AC97 device.
1310 void snd_soc_free_pcms(struct snd_soc_device
*socdev
)
1312 struct snd_soc_codec
*codec
= socdev
->codec
;
1313 #ifdef CONFIG_SND_SOC_AC97_BUS
1314 struct snd_soc_dai
*codec_dai
;
1318 mutex_lock(&codec
->mutex
);
1319 soc_cleanup_debugfs(socdev
);
1320 #ifdef CONFIG_SND_SOC_AC97_BUS
1321 for (i
= 0; i
< codec
->num_dai
; i
++) {
1322 codec_dai
= &codec
->dai
[i
];
1323 if (codec_dai
->ac97_control
&& codec
->ac97
) {
1324 soc_ac97_dev_unregister(codec
);
1332 snd_card_free(codec
->card
);
1333 device_remove_file(socdev
->dev
, &dev_attr_codec_reg
);
1334 mutex_unlock(&codec
->mutex
);
1336 EXPORT_SYMBOL_GPL(snd_soc_free_pcms
);
1339 * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
1340 * @substream: the pcm substream
1341 * @hw: the hardware parameters
1343 * Sets the substream runtime hardware parameters.
1345 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream
*substream
,
1346 const struct snd_pcm_hardware
*hw
)
1348 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
1349 runtime
->hw
.info
= hw
->info
;
1350 runtime
->hw
.formats
= hw
->formats
;
1351 runtime
->hw
.period_bytes_min
= hw
->period_bytes_min
;
1352 runtime
->hw
.period_bytes_max
= hw
->period_bytes_max
;
1353 runtime
->hw
.periods_min
= hw
->periods_min
;
1354 runtime
->hw
.periods_max
= hw
->periods_max
;
1355 runtime
->hw
.buffer_bytes_max
= hw
->buffer_bytes_max
;
1356 runtime
->hw
.fifo_size
= hw
->fifo_size
;
1359 EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams
);
1362 * snd_soc_cnew - create new control
1363 * @_template: control template
1364 * @data: control private data
1365 * @lnng_name: control long name
1367 * Create a new mixer control from a template control.
1369 * Returns 0 for success, else error.
1371 struct snd_kcontrol
*snd_soc_cnew(const struct snd_kcontrol_new
*_template
,
1372 void *data
, char *long_name
)
1374 struct snd_kcontrol_new
template;
1376 memcpy(&template, _template
, sizeof(template));
1378 template.name
= long_name
;
1381 return snd_ctl_new1(&template, data
);
1383 EXPORT_SYMBOL_GPL(snd_soc_cnew
);
1386 * snd_soc_info_enum_double - enumerated double mixer info callback
1387 * @kcontrol: mixer control
1388 * @uinfo: control element information
1390 * Callback to provide information about a double enumerated
1393 * Returns 0 for success.
1395 int snd_soc_info_enum_double(struct snd_kcontrol
*kcontrol
,
1396 struct snd_ctl_elem_info
*uinfo
)
1398 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1400 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
1401 uinfo
->count
= e
->shift_l
== e
->shift_r
? 1 : 2;
1402 uinfo
->value
.enumerated
.items
= e
->max
;
1404 if (uinfo
->value
.enumerated
.item
> e
->max
- 1)
1405 uinfo
->value
.enumerated
.item
= e
->max
- 1;
1406 strcpy(uinfo
->value
.enumerated
.name
,
1407 e
->texts
[uinfo
->value
.enumerated
.item
]);
1410 EXPORT_SYMBOL_GPL(snd_soc_info_enum_double
);
1413 * snd_soc_get_enum_double - enumerated double mixer get callback
1414 * @kcontrol: mixer control
1415 * @uinfo: control element information
1417 * Callback to get the value of a double enumerated mixer.
1419 * Returns 0 for success.
1421 int snd_soc_get_enum_double(struct snd_kcontrol
*kcontrol
,
1422 struct snd_ctl_elem_value
*ucontrol
)
1424 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1425 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1426 unsigned short val
, bitmask
;
1428 for (bitmask
= 1; bitmask
< e
->max
; bitmask
<<= 1)
1430 val
= snd_soc_read(codec
, e
->reg
);
1431 ucontrol
->value
.enumerated
.item
[0]
1432 = (val
>> e
->shift_l
) & (bitmask
- 1);
1433 if (e
->shift_l
!= e
->shift_r
)
1434 ucontrol
->value
.enumerated
.item
[1] =
1435 (val
>> e
->shift_r
) & (bitmask
- 1);
1439 EXPORT_SYMBOL_GPL(snd_soc_get_enum_double
);
1442 * snd_soc_put_enum_double - enumerated double mixer put callback
1443 * @kcontrol: mixer control
1444 * @uinfo: control element information
1446 * Callback to set the value of a double enumerated mixer.
1448 * Returns 0 for success.
1450 int snd_soc_put_enum_double(struct snd_kcontrol
*kcontrol
,
1451 struct snd_ctl_elem_value
*ucontrol
)
1453 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1454 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1456 unsigned short mask
, bitmask
;
1458 for (bitmask
= 1; bitmask
< e
->max
; bitmask
<<= 1)
1460 if (ucontrol
->value
.enumerated
.item
[0] > e
->max
- 1)
1462 val
= ucontrol
->value
.enumerated
.item
[0] << e
->shift_l
;
1463 mask
= (bitmask
- 1) << e
->shift_l
;
1464 if (e
->shift_l
!= e
->shift_r
) {
1465 if (ucontrol
->value
.enumerated
.item
[1] > e
->max
- 1)
1467 val
|= ucontrol
->value
.enumerated
.item
[1] << e
->shift_r
;
1468 mask
|= (bitmask
- 1) << e
->shift_r
;
1471 return snd_soc_update_bits(codec
, e
->reg
, mask
, val
);
1473 EXPORT_SYMBOL_GPL(snd_soc_put_enum_double
);
1476 * snd_soc_info_enum_ext - external enumerated single mixer info callback
1477 * @kcontrol: mixer control
1478 * @uinfo: control element information
1480 * Callback to provide information about an external enumerated
1483 * Returns 0 for success.
1485 int snd_soc_info_enum_ext(struct snd_kcontrol
*kcontrol
,
1486 struct snd_ctl_elem_info
*uinfo
)
1488 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1490 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
1492 uinfo
->value
.enumerated
.items
= e
->max
;
1494 if (uinfo
->value
.enumerated
.item
> e
->max
- 1)
1495 uinfo
->value
.enumerated
.item
= e
->max
- 1;
1496 strcpy(uinfo
->value
.enumerated
.name
,
1497 e
->texts
[uinfo
->value
.enumerated
.item
]);
1500 EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext
);
1503 * snd_soc_info_volsw_ext - external single mixer info callback
1504 * @kcontrol: mixer control
1505 * @uinfo: control element information
1507 * Callback to provide information about a single external mixer control.
1509 * Returns 0 for success.
1511 int snd_soc_info_volsw_ext(struct snd_kcontrol
*kcontrol
,
1512 struct snd_ctl_elem_info
*uinfo
)
1514 int max
= kcontrol
->private_value
;
1517 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1519 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1522 uinfo
->value
.integer
.min
= 0;
1523 uinfo
->value
.integer
.max
= max
;
1526 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext
);
1529 * snd_soc_info_volsw - single mixer info callback
1530 * @kcontrol: mixer control
1531 * @uinfo: control element information
1533 * Callback to provide information about a single mixer control.
1535 * Returns 0 for success.
1537 int snd_soc_info_volsw(struct snd_kcontrol
*kcontrol
,
1538 struct snd_ctl_elem_info
*uinfo
)
1540 struct soc_mixer_control
*mc
=
1541 (struct soc_mixer_control
*)kcontrol
->private_value
;
1543 unsigned int shift
= mc
->shift
;
1544 unsigned int rshift
= mc
->rshift
;
1547 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1549 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1551 uinfo
->count
= shift
== rshift
? 1 : 2;
1552 uinfo
->value
.integer
.min
= 0;
1553 uinfo
->value
.integer
.max
= max
;
1556 EXPORT_SYMBOL_GPL(snd_soc_info_volsw
);
1559 * snd_soc_get_volsw - single mixer get callback
1560 * @kcontrol: mixer control
1561 * @uinfo: control element information
1563 * Callback to get the value of a single mixer control.
1565 * Returns 0 for success.
1567 int snd_soc_get_volsw(struct snd_kcontrol
*kcontrol
,
1568 struct snd_ctl_elem_value
*ucontrol
)
1570 struct soc_mixer_control
*mc
=
1571 (struct soc_mixer_control
*)kcontrol
->private_value
;
1572 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1573 unsigned int reg
= mc
->reg
;
1574 unsigned int shift
= mc
->shift
;
1575 unsigned int rshift
= mc
->rshift
;
1577 unsigned int mask
= (1 << fls(max
)) - 1;
1578 unsigned int invert
= mc
->invert
;
1580 ucontrol
->value
.integer
.value
[0] =
1581 (snd_soc_read(codec
, reg
) >> shift
) & mask
;
1582 if (shift
!= rshift
)
1583 ucontrol
->value
.integer
.value
[1] =
1584 (snd_soc_read(codec
, reg
) >> rshift
) & mask
;
1586 ucontrol
->value
.integer
.value
[0] =
1587 max
- ucontrol
->value
.integer
.value
[0];
1588 if (shift
!= rshift
)
1589 ucontrol
->value
.integer
.value
[1] =
1590 max
- ucontrol
->value
.integer
.value
[1];
1595 EXPORT_SYMBOL_GPL(snd_soc_get_volsw
);
1598 * snd_soc_put_volsw - single mixer put callback
1599 * @kcontrol: mixer control
1600 * @uinfo: control element information
1602 * Callback to set the value of a single mixer control.
1604 * Returns 0 for success.
1606 int snd_soc_put_volsw(struct snd_kcontrol
*kcontrol
,
1607 struct snd_ctl_elem_value
*ucontrol
)
1609 struct soc_mixer_control
*mc
=
1610 (struct soc_mixer_control
*)kcontrol
->private_value
;
1611 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1612 unsigned int reg
= mc
->reg
;
1613 unsigned int shift
= mc
->shift
;
1614 unsigned int rshift
= mc
->rshift
;
1616 unsigned int mask
= (1 << fls(max
)) - 1;
1617 unsigned int invert
= mc
->invert
;
1618 unsigned short val
, val2
, val_mask
;
1620 val
= (ucontrol
->value
.integer
.value
[0] & mask
);
1623 val_mask
= mask
<< shift
;
1625 if (shift
!= rshift
) {
1626 val2
= (ucontrol
->value
.integer
.value
[1] & mask
);
1629 val_mask
|= mask
<< rshift
;
1630 val
|= val2
<< rshift
;
1632 return snd_soc_update_bits(codec
, reg
, val_mask
, val
);
1634 EXPORT_SYMBOL_GPL(snd_soc_put_volsw
);
1637 * snd_soc_info_volsw_2r - double mixer info callback
1638 * @kcontrol: mixer control
1639 * @uinfo: control element information
1641 * Callback to provide information about a double mixer control that
1642 * spans 2 codec registers.
1644 * Returns 0 for success.
1646 int snd_soc_info_volsw_2r(struct snd_kcontrol
*kcontrol
,
1647 struct snd_ctl_elem_info
*uinfo
)
1649 struct soc_mixer_control
*mc
=
1650 (struct soc_mixer_control
*)kcontrol
->private_value
;
1654 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1656 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1659 uinfo
->value
.integer
.min
= 0;
1660 uinfo
->value
.integer
.max
= max
;
1663 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r
);
1666 * snd_soc_get_volsw_2r - double mixer get callback
1667 * @kcontrol: mixer control
1668 * @uinfo: control element information
1670 * Callback to get the value of a double mixer control that spans 2 registers.
1672 * Returns 0 for success.
1674 int snd_soc_get_volsw_2r(struct snd_kcontrol
*kcontrol
,
1675 struct snd_ctl_elem_value
*ucontrol
)
1677 struct soc_mixer_control
*mc
=
1678 (struct soc_mixer_control
*)kcontrol
->private_value
;
1679 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1680 unsigned int reg
= mc
->reg
;
1681 unsigned int reg2
= mc
->rreg
;
1682 unsigned int shift
= mc
->shift
;
1684 unsigned int mask
= (1<<fls(max
))-1;
1685 unsigned int invert
= mc
->invert
;
1687 ucontrol
->value
.integer
.value
[0] =
1688 (snd_soc_read(codec
, reg
) >> shift
) & mask
;
1689 ucontrol
->value
.integer
.value
[1] =
1690 (snd_soc_read(codec
, reg2
) >> shift
) & mask
;
1692 ucontrol
->value
.integer
.value
[0] =
1693 max
- ucontrol
->value
.integer
.value
[0];
1694 ucontrol
->value
.integer
.value
[1] =
1695 max
- ucontrol
->value
.integer
.value
[1];
1700 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r
);
1703 * snd_soc_put_volsw_2r - double mixer set callback
1704 * @kcontrol: mixer control
1705 * @uinfo: control element information
1707 * Callback to set the value of a double mixer control that spans 2 registers.
1709 * Returns 0 for success.
1711 int snd_soc_put_volsw_2r(struct snd_kcontrol
*kcontrol
,
1712 struct snd_ctl_elem_value
*ucontrol
)
1714 struct soc_mixer_control
*mc
=
1715 (struct soc_mixer_control
*)kcontrol
->private_value
;
1716 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1717 unsigned int reg
= mc
->reg
;
1718 unsigned int reg2
= mc
->rreg
;
1719 unsigned int shift
= mc
->shift
;
1721 unsigned int mask
= (1 << fls(max
)) - 1;
1722 unsigned int invert
= mc
->invert
;
1724 unsigned short val
, val2
, val_mask
;
1726 val_mask
= mask
<< shift
;
1727 val
= (ucontrol
->value
.integer
.value
[0] & mask
);
1728 val2
= (ucontrol
->value
.integer
.value
[1] & mask
);
1736 val2
= val2
<< shift
;
1738 err
= snd_soc_update_bits(codec
, reg
, val_mask
, val
);
1742 err
= snd_soc_update_bits(codec
, reg2
, val_mask
, val2
);
1745 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r
);
1748 * snd_soc_info_volsw_s8 - signed mixer info callback
1749 * @kcontrol: mixer control
1750 * @uinfo: control element information
1752 * Callback to provide information about a signed mixer control.
1754 * Returns 0 for success.
1756 int snd_soc_info_volsw_s8(struct snd_kcontrol
*kcontrol
,
1757 struct snd_ctl_elem_info
*uinfo
)
1759 struct soc_mixer_control
*mc
=
1760 (struct soc_mixer_control
*)kcontrol
->private_value
;
1764 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1766 uinfo
->value
.integer
.min
= 0;
1767 uinfo
->value
.integer
.max
= max
-min
;
1770 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8
);
1773 * snd_soc_get_volsw_s8 - signed mixer get callback
1774 * @kcontrol: mixer control
1775 * @uinfo: control element information
1777 * Callback to get the value of a signed mixer control.
1779 * Returns 0 for success.
1781 int snd_soc_get_volsw_s8(struct snd_kcontrol
*kcontrol
,
1782 struct snd_ctl_elem_value
*ucontrol
)
1784 struct soc_mixer_control
*mc
=
1785 (struct soc_mixer_control
*)kcontrol
->private_value
;
1786 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1787 unsigned int reg
= mc
->reg
;
1789 int val
= snd_soc_read(codec
, reg
);
1791 ucontrol
->value
.integer
.value
[0] =
1792 ((signed char)(val
& 0xff))-min
;
1793 ucontrol
->value
.integer
.value
[1] =
1794 ((signed char)((val
>> 8) & 0xff))-min
;
1797 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8
);
1800 * snd_soc_put_volsw_sgn - signed mixer put callback
1801 * @kcontrol: mixer control
1802 * @uinfo: control element information
1804 * Callback to set the value of a signed mixer control.
1806 * Returns 0 for success.
1808 int snd_soc_put_volsw_s8(struct snd_kcontrol
*kcontrol
,
1809 struct snd_ctl_elem_value
*ucontrol
)
1811 struct soc_mixer_control
*mc
=
1812 (struct soc_mixer_control
*)kcontrol
->private_value
;
1813 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1814 unsigned int reg
= mc
->reg
;
1818 val
= (ucontrol
->value
.integer
.value
[0]+min
) & 0xff;
1819 val
|= ((ucontrol
->value
.integer
.value
[1]+min
) & 0xff) << 8;
1821 return snd_soc_update_bits(codec
, reg
, 0xffff, val
);
1823 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8
);
1826 * snd_soc_dai_set_sysclk - configure DAI system or master clock.
1828 * @clk_id: DAI specific clock ID
1829 * @freq: new clock frequency in Hz
1830 * @dir: new clock direction - input/output.
1832 * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
1834 int snd_soc_dai_set_sysclk(struct snd_soc_dai
*dai
, int clk_id
,
1835 unsigned int freq
, int dir
)
1837 if (dai
->ops
.set_sysclk
)
1838 return dai
->ops
.set_sysclk(dai
, clk_id
, freq
, dir
);
1842 EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk
);
1845 * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
1847 * @clk_id: DAI specific clock divider ID
1848 * @div: new clock divisor.
1850 * Configures the clock dividers. This is used to derive the best DAI bit and
1851 * frame clocks from the system or master clock. It's best to set the DAI bit
1852 * and frame clocks as low as possible to save system power.
1854 int snd_soc_dai_set_clkdiv(struct snd_soc_dai
*dai
,
1855 int div_id
, int div
)
1857 if (dai
->ops
.set_clkdiv
)
1858 return dai
->ops
.set_clkdiv(dai
, div_id
, div
);
1862 EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv
);
1865 * snd_soc_dai_set_pll - configure DAI PLL.
1867 * @pll_id: DAI specific PLL ID
1868 * @freq_in: PLL input clock frequency in Hz
1869 * @freq_out: requested PLL output clock frequency in Hz
1871 * Configures and enables PLL to generate output clock based on input clock.
1873 int snd_soc_dai_set_pll(struct snd_soc_dai
*dai
,
1874 int pll_id
, unsigned int freq_in
, unsigned int freq_out
)
1876 if (dai
->ops
.set_pll
)
1877 return dai
->ops
.set_pll(dai
, pll_id
, freq_in
, freq_out
);
1881 EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll
);
1884 * snd_soc_dai_set_fmt - configure DAI hardware audio format.
1886 * @fmt: SND_SOC_DAIFMT_ format value.
1888 * Configures the DAI hardware format and clocking.
1890 int snd_soc_dai_set_fmt(struct snd_soc_dai
*dai
, unsigned int fmt
)
1892 if (dai
->ops
.set_fmt
)
1893 return dai
->ops
.set_fmt(dai
, fmt
);
1897 EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt
);
1900 * snd_soc_dai_set_tdm_slot - configure DAI TDM.
1902 * @mask: DAI specific mask representing used slots.
1903 * @slots: Number of slots in use.
1905 * Configures a DAI for TDM operation. Both mask and slots are codec and DAI
1908 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai
*dai
,
1909 unsigned int mask
, int slots
)
1911 if (dai
->ops
.set_sysclk
)
1912 return dai
->ops
.set_tdm_slot(dai
, mask
, slots
);
1916 EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot
);
1919 * snd_soc_dai_set_tristate - configure DAI system or master clock.
1921 * @tristate: tristate enable
1923 * Tristates the DAI so that others can use it.
1925 int snd_soc_dai_set_tristate(struct snd_soc_dai
*dai
, int tristate
)
1927 if (dai
->ops
.set_sysclk
)
1928 return dai
->ops
.set_tristate(dai
, tristate
);
1932 EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate
);
1935 * snd_soc_dai_digital_mute - configure DAI system or master clock.
1937 * @mute: mute enable
1939 * Mutes the DAI DAC.
1941 int snd_soc_dai_digital_mute(struct snd_soc_dai
*dai
, int mute
)
1943 if (dai
->ops
.digital_mute
)
1944 return dai
->ops
.digital_mute(dai
, mute
);
1948 EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute
);
1950 static int __devinit
snd_soc_init(void)
1952 return platform_driver_register(&soc_driver
);
1955 static void snd_soc_exit(void)
1957 platform_driver_unregister(&soc_driver
);
1960 module_init(snd_soc_init
);
1961 module_exit(snd_soc_exit
);
1963 /* Module information */
1964 MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
1965 MODULE_DESCRIPTION("ALSA SoC Core");
1966 MODULE_LICENSE("GPL");
1967 MODULE_ALIAS("platform:soc-audio");