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 * 12th Aug 2005 Initial version.
19 * 25th Oct 2005 Working Codec, Interface and Platform registration.
22 * o Add hw rules to enforce rates, etc.
23 * o More testing with other codecs/machines.
24 * o Add more codecs and platforms to ensure good API coverage.
25 * o Support TDM on PCM and I2S
28 #include <linux/module.h>
29 #include <linux/moduleparam.h>
30 #include <linux/init.h>
31 #include <linux/delay.h>
33 #include <linux/bitops.h>
34 #include <linux/platform_device.h>
35 #include <sound/core.h>
36 #include <sound/pcm.h>
37 #include <sound/pcm_params.h>
38 #include <sound/soc.h>
39 #include <sound/soc-dapm.h>
40 #include <sound/initval.h>
45 #define dbg(format, arg...) printk(format, ## arg)
47 #define dbg(format, arg...)
50 static DEFINE_MUTEX(pcm_mutex
);
51 static DEFINE_MUTEX(io_mutex
);
52 static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq
);
55 * This is a timeout to do a DAPM powerdown after a stream is closed().
56 * It can be used to eliminate pops between different playback streams, e.g.
57 * between two audio tracks.
59 static int pmdown_time
= 5000;
60 module_param(pmdown_time
, int, 0);
61 MODULE_PARM_DESC(pmdown_time
, "DAPM stream powerdown time (msecs)");
64 * This function forces any delayed work to be queued and run.
66 static int run_delayed_work(struct delayed_work
*dwork
)
70 /* cancel any work waiting to be queued. */
71 ret
= cancel_delayed_work(dwork
);
73 /* if there was any work waiting then we run it now and
74 * wait for it's completion */
76 schedule_delayed_work(dwork
, 0);
77 flush_scheduled_work();
82 #ifdef CONFIG_SND_SOC_AC97_BUS
83 /* unregister ac97 codec */
84 static int soc_ac97_dev_unregister(struct snd_soc_codec
*codec
)
86 if (codec
->ac97
->dev
.bus
)
87 device_unregister(&codec
->ac97
->dev
);
91 /* stop no dev release warning */
92 static void soc_ac97_device_release(struct device
*dev
){}
94 /* register ac97 codec to bus */
95 static int soc_ac97_dev_register(struct snd_soc_codec
*codec
)
99 codec
->ac97
->dev
.bus
= &ac97_bus_type
;
100 codec
->ac97
->dev
.parent
= NULL
;
101 codec
->ac97
->dev
.release
= soc_ac97_device_release
;
103 snprintf(codec
->ac97
->dev
.bus_id
, BUS_ID_SIZE
, "%d-%d:%s",
104 codec
->card
->number
, 0, codec
->name
);
105 err
= device_register(&codec
->ac97
->dev
);
107 snd_printk(KERN_ERR
"Can't register ac97 bus\n");
108 codec
->ac97
->dev
.bus
= NULL
;
115 static inline const char* get_dai_name(int type
)
118 case SND_SOC_DAI_AC97_BUS
:
119 case SND_SOC_DAI_AC97
:
121 case SND_SOC_DAI_I2S
:
123 case SND_SOC_DAI_PCM
:
130 * Called by ALSA when a PCM substream is opened, the runtime->hw record is
131 * then initialized and any private data can be allocated. This also calls
132 * startup for the cpu DAI, platform, machine and codec DAI.
134 static int soc_pcm_open(struct snd_pcm_substream
*substream
)
136 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
137 struct snd_soc_device
*socdev
= rtd
->socdev
;
138 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
139 struct snd_soc_dai_link
*machine
= rtd
->dai
;
140 struct snd_soc_platform
*platform
= socdev
->platform
;
141 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
142 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
145 mutex_lock(&pcm_mutex
);
147 /* startup the audio subsystem */
148 if (cpu_dai
->ops
.startup
) {
149 ret
= cpu_dai
->ops
.startup(substream
);
151 printk(KERN_ERR
"asoc: can't open interface %s\n",
157 if (platform
->pcm_ops
->open
) {
158 ret
= platform
->pcm_ops
->open(substream
);
160 printk(KERN_ERR
"asoc: can't open platform %s\n", platform
->name
);
165 if (codec_dai
->ops
.startup
) {
166 ret
= codec_dai
->ops
.startup(substream
);
168 printk(KERN_ERR
"asoc: can't open codec %s\n",
174 if (machine
->ops
&& machine
->ops
->startup
) {
175 ret
= machine
->ops
->startup(substream
);
177 printk(KERN_ERR
"asoc: %s startup failed\n", machine
->name
);
182 /* Check that the codec and cpu DAI's are compatible */
183 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
) {
184 runtime
->hw
.rate_min
=
185 max(codec_dai
->playback
.rate_min
, cpu_dai
->playback
.rate_min
);
186 runtime
->hw
.rate_max
=
187 min(codec_dai
->playback
.rate_max
, cpu_dai
->playback
.rate_max
);
188 runtime
->hw
.channels_min
=
189 max(codec_dai
->playback
.channels_min
,
190 cpu_dai
->playback
.channels_min
);
191 runtime
->hw
.channels_max
=
192 min(codec_dai
->playback
.channels_max
,
193 cpu_dai
->playback
.channels_max
);
194 runtime
->hw
.formats
=
195 codec_dai
->playback
.formats
& cpu_dai
->playback
.formats
;
197 codec_dai
->playback
.rates
& cpu_dai
->playback
.rates
;
199 runtime
->hw
.rate_min
=
200 max(codec_dai
->capture
.rate_min
, cpu_dai
->capture
.rate_min
);
201 runtime
->hw
.rate_max
=
202 min(codec_dai
->capture
.rate_max
, 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 /* power down the codec to D1 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_device_event(socdev
,
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 /* power down the codec power domain 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_device_event(socdev
,
308 SNDRV_CTL_POWER_D3hot
);
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_device_event(socdev
,
369 SNDRV_CTL_POWER_D3hot
);
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
->dapm_state
!= SNDRV_CTL_POWER_D0
) {
444 snd_soc_dapm_device_event(socdev
, SNDRV_CTL_POWER_D1
);
446 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
447 snd_soc_dapm_stream_event(codec
,
448 codec_dai
->playback
.stream_name
,
449 SND_SOC_DAPM_STREAM_START
);
451 snd_soc_dapm_stream_event(codec
,
452 codec_dai
->capture
.stream_name
,
453 SND_SOC_DAPM_STREAM_START
);
455 snd_soc_dapm_device_event(socdev
, SNDRV_CTL_POWER_D0
);
456 if (codec_dai
->dai_ops
.digital_mute
)
457 codec_dai
->dai_ops
.digital_mute(codec_dai
, 0);
460 /* codec already powered - power on widgets */
461 if (substream
->stream
== SNDRV_PCM_STREAM_PLAYBACK
)
462 snd_soc_dapm_stream_event(codec
,
463 codec_dai
->playback
.stream_name
,
464 SND_SOC_DAPM_STREAM_START
);
466 snd_soc_dapm_stream_event(codec
,
467 codec_dai
->capture
.stream_name
,
468 SND_SOC_DAPM_STREAM_START
);
469 if (codec_dai
->dai_ops
.digital_mute
)
470 codec_dai
->dai_ops
.digital_mute(codec_dai
, 0);
475 mutex_unlock(&pcm_mutex
);
480 * Called by ALSA when the hardware params are set by application. This
481 * function can also be called multiple times and can allocate buffers
482 * (using snd_pcm_lib_* ). It's non-atomic.
484 static int soc_pcm_hw_params(struct snd_pcm_substream
*substream
,
485 struct snd_pcm_hw_params
*params
)
487 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
488 struct snd_soc_device
*socdev
= rtd
->socdev
;
489 struct snd_soc_dai_link
*machine
= rtd
->dai
;
490 struct snd_soc_platform
*platform
= socdev
->platform
;
491 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
492 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
495 mutex_lock(&pcm_mutex
);
497 if (machine
->ops
&& machine
->ops
->hw_params
) {
498 ret
= machine
->ops
->hw_params(substream
, params
);
500 printk(KERN_ERR
"asoc: machine hw_params failed\n");
505 if (codec_dai
->ops
.hw_params
) {
506 ret
= codec_dai
->ops
.hw_params(substream
, params
);
508 printk(KERN_ERR
"asoc: can't set codec %s hw params\n",
514 if (cpu_dai
->ops
.hw_params
) {
515 ret
= cpu_dai
->ops
.hw_params(substream
, params
);
517 printk(KERN_ERR
"asoc: can't set interface %s hw params\n",
523 if (platform
->pcm_ops
->hw_params
) {
524 ret
= platform
->pcm_ops
->hw_params(substream
, params
);
526 printk(KERN_ERR
"asoc: can't set platform %s hw params\n",
533 mutex_unlock(&pcm_mutex
);
537 if (cpu_dai
->ops
.hw_free
)
538 cpu_dai
->ops
.hw_free(substream
);
541 if (codec_dai
->ops
.hw_free
)
542 codec_dai
->ops
.hw_free(substream
);
545 if(machine
->ops
&& machine
->ops
->hw_free
)
546 machine
->ops
->hw_free(substream
);
548 mutex_unlock(&pcm_mutex
);
553 * Free's resources allocated by hw_params, can be called multiple times
555 static int soc_pcm_hw_free(struct snd_pcm_substream
*substream
)
557 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
558 struct snd_soc_device
*socdev
= rtd
->socdev
;
559 struct snd_soc_dai_link
*machine
= rtd
->dai
;
560 struct snd_soc_platform
*platform
= socdev
->platform
;
561 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
562 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
563 struct snd_soc_codec
*codec
= socdev
->codec
;
565 mutex_lock(&pcm_mutex
);
567 /* apply codec digital mute */
568 if (!codec
->active
&& codec_dai
->dai_ops
.digital_mute
)
569 codec_dai
->dai_ops
.digital_mute(codec_dai
, 1);
571 /* free any machine hw params */
572 if (machine
->ops
&& machine
->ops
->hw_free
)
573 machine
->ops
->hw_free(substream
);
575 /* free any DMA resources */
576 if (platform
->pcm_ops
->hw_free
)
577 platform
->pcm_ops
->hw_free(substream
);
579 /* now free hw params for the DAI's */
580 if (codec_dai
->ops
.hw_free
)
581 codec_dai
->ops
.hw_free(substream
);
583 if (cpu_dai
->ops
.hw_free
)
584 cpu_dai
->ops
.hw_free(substream
);
586 mutex_unlock(&pcm_mutex
);
590 static int soc_pcm_trigger(struct snd_pcm_substream
*substream
, int cmd
)
592 struct snd_soc_pcm_runtime
*rtd
= substream
->private_data
;
593 struct snd_soc_device
*socdev
= rtd
->socdev
;
594 struct snd_soc_dai_link
*machine
= rtd
->dai
;
595 struct snd_soc_platform
*platform
= socdev
->platform
;
596 struct snd_soc_cpu_dai
*cpu_dai
= machine
->cpu_dai
;
597 struct snd_soc_codec_dai
*codec_dai
= machine
->codec_dai
;
600 if (codec_dai
->ops
.trigger
) {
601 ret
= codec_dai
->ops
.trigger(substream
, cmd
);
606 if (platform
->pcm_ops
->trigger
) {
607 ret
= platform
->pcm_ops
->trigger(substream
, cmd
);
612 if (cpu_dai
->ops
.trigger
) {
613 ret
= cpu_dai
->ops
.trigger(substream
, cmd
);
620 /* ASoC PCM operations */
621 static struct snd_pcm_ops soc_pcm_ops
= {
622 .open
= soc_pcm_open
,
623 .close
= soc_codec_close
,
624 .hw_params
= soc_pcm_hw_params
,
625 .hw_free
= soc_pcm_hw_free
,
626 .prepare
= soc_pcm_prepare
,
627 .trigger
= soc_pcm_trigger
,
631 /* powers down audio subsystem for suspend */
632 static int soc_suspend(struct platform_device
*pdev
, pm_message_t state
)
634 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
635 struct snd_soc_machine
*machine
= socdev
->machine
;
636 struct snd_soc_platform
*platform
= socdev
->platform
;
637 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
638 struct snd_soc_codec
*codec
= socdev
->codec
;
641 /* mute any active DAC's */
642 for(i
= 0; i
< machine
->num_links
; i
++) {
643 struct snd_soc_codec_dai
*dai
= machine
->dai_link
[i
].codec_dai
;
644 if (dai
->dai_ops
.digital_mute
&& dai
->playback
.active
)
645 dai
->dai_ops
.digital_mute(dai
, 1);
648 /* suspend all pcms */
649 for (i
= 0; i
< machine
->num_links
; i
++)
650 snd_pcm_suspend_all(machine
->dai_link
[i
].pcm
);
652 if (machine
->suspend_pre
)
653 machine
->suspend_pre(pdev
, state
);
655 for(i
= 0; i
< machine
->num_links
; i
++) {
656 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
657 if (cpu_dai
->suspend
&& cpu_dai
->type
!= SND_SOC_DAI_AC97
)
658 cpu_dai
->suspend(pdev
, cpu_dai
);
659 if (platform
->suspend
)
660 platform
->suspend(pdev
, cpu_dai
);
663 /* close any waiting streams and save state */
664 run_delayed_work(&socdev
->delayed_work
);
665 codec
->suspend_dapm_state
= codec
->dapm_state
;
667 for(i
= 0; i
< codec
->num_dai
; i
++) {
668 char *stream
= codec
->dai
[i
].playback
.stream_name
;
670 snd_soc_dapm_stream_event(codec
, stream
,
671 SND_SOC_DAPM_STREAM_SUSPEND
);
672 stream
= codec
->dai
[i
].capture
.stream_name
;
674 snd_soc_dapm_stream_event(codec
, stream
,
675 SND_SOC_DAPM_STREAM_SUSPEND
);
678 if (codec_dev
->suspend
)
679 codec_dev
->suspend(pdev
, state
);
681 for(i
= 0; i
< machine
->num_links
; i
++) {
682 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
683 if (cpu_dai
->suspend
&& cpu_dai
->type
== SND_SOC_DAI_AC97
)
684 cpu_dai
->suspend(pdev
, cpu_dai
);
687 if (machine
->suspend_post
)
688 machine
->suspend_post(pdev
, state
);
693 /* powers up audio subsystem after a suspend */
694 static int soc_resume(struct platform_device
*pdev
)
696 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
697 struct snd_soc_machine
*machine
= socdev
->machine
;
698 struct snd_soc_platform
*platform
= socdev
->platform
;
699 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
700 struct snd_soc_codec
*codec
= socdev
->codec
;
703 if (machine
->resume_pre
)
704 machine
->resume_pre(pdev
);
706 for(i
= 0; i
< machine
->num_links
; i
++) {
707 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
708 if (cpu_dai
->resume
&& cpu_dai
->type
== SND_SOC_DAI_AC97
)
709 cpu_dai
->resume(pdev
, cpu_dai
);
712 if (codec_dev
->resume
)
713 codec_dev
->resume(pdev
);
715 for(i
= 0; i
< codec
->num_dai
; i
++) {
716 char* stream
= codec
->dai
[i
].playback
.stream_name
;
718 snd_soc_dapm_stream_event(codec
, stream
,
719 SND_SOC_DAPM_STREAM_RESUME
);
720 stream
= codec
->dai
[i
].capture
.stream_name
;
722 snd_soc_dapm_stream_event(codec
, stream
,
723 SND_SOC_DAPM_STREAM_RESUME
);
726 /* unmute any active DAC's */
727 for(i
= 0; i
< machine
->num_links
; i
++) {
728 struct snd_soc_codec_dai
*dai
= machine
->dai_link
[i
].codec_dai
;
729 if (dai
->dai_ops
.digital_mute
&& dai
->playback
.active
)
730 dai
->dai_ops
.digital_mute(dai
, 0);
733 for(i
= 0; i
< machine
->num_links
; i
++) {
734 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
735 if (cpu_dai
->resume
&& cpu_dai
->type
!= SND_SOC_DAI_AC97
)
736 cpu_dai
->resume(pdev
, cpu_dai
);
737 if (platform
->resume
)
738 platform
->resume(pdev
, cpu_dai
);
741 if (machine
->resume_post
)
742 machine
->resume_post(pdev
);
748 #define soc_suspend NULL
749 #define soc_resume NULL
752 /* probes a new socdev */
753 static int soc_probe(struct platform_device
*pdev
)
756 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
757 struct snd_soc_machine
*machine
= socdev
->machine
;
758 struct snd_soc_platform
*platform
= socdev
->platform
;
759 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
761 if (machine
->probe
) {
762 ret
= machine
->probe(pdev
);
767 for (i
= 0; i
< machine
->num_links
; i
++) {
768 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
769 if (cpu_dai
->probe
) {
770 ret
= cpu_dai
->probe(pdev
);
776 if (codec_dev
->probe
) {
777 ret
= codec_dev
->probe(pdev
);
782 if (platform
->probe
) {
783 ret
= platform
->probe(pdev
);
788 /* DAPM stream work */
789 INIT_DELAYED_WORK(&socdev
->delayed_work
, close_delayed_work
);
793 if (codec_dev
->remove
)
794 codec_dev
->remove(pdev
);
797 for (i
--; i
>= 0; i
--) {
798 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
800 cpu_dai
->remove(pdev
);
804 machine
->remove(pdev
);
809 /* removes a socdev */
810 static int soc_remove(struct platform_device
*pdev
)
813 struct snd_soc_device
*socdev
= platform_get_drvdata(pdev
);
814 struct snd_soc_machine
*machine
= socdev
->machine
;
815 struct snd_soc_platform
*platform
= socdev
->platform
;
816 struct snd_soc_codec_device
*codec_dev
= socdev
->codec_dev
;
818 run_delayed_work(&socdev
->delayed_work
);
820 if (platform
->remove
)
821 platform
->remove(pdev
);
823 if (codec_dev
->remove
)
824 codec_dev
->remove(pdev
);
826 for (i
= 0; i
< machine
->num_links
; i
++) {
827 struct snd_soc_cpu_dai
*cpu_dai
= machine
->dai_link
[i
].cpu_dai
;
829 cpu_dai
->remove(pdev
);
833 machine
->remove(pdev
);
838 /* ASoC platform driver */
839 static struct platform_driver soc_driver
= {
842 .owner
= THIS_MODULE
,
845 .remove
= soc_remove
,
846 .suspend
= soc_suspend
,
847 .resume
= soc_resume
,
850 /* create a new pcm */
851 static int soc_new_pcm(struct snd_soc_device
*socdev
,
852 struct snd_soc_dai_link
*dai_link
, int num
)
854 struct snd_soc_codec
*codec
= socdev
->codec
;
855 struct snd_soc_codec_dai
*codec_dai
= dai_link
->codec_dai
;
856 struct snd_soc_cpu_dai
*cpu_dai
= dai_link
->cpu_dai
;
857 struct snd_soc_pcm_runtime
*rtd
;
860 int ret
= 0, playback
= 0, capture
= 0;
862 rtd
= kzalloc(sizeof(struct snd_soc_pcm_runtime
), GFP_KERNEL
);
867 rtd
->socdev
= socdev
;
868 codec_dai
->codec
= socdev
->codec
;
870 /* check client and interface hw capabilities */
871 sprintf(new_name
, "%s %s-%s-%d",dai_link
->stream_name
, codec_dai
->name
,
872 get_dai_name(cpu_dai
->type
), num
);
874 if (codec_dai
->playback
.channels_min
)
876 if (codec_dai
->capture
.channels_min
)
879 ret
= snd_pcm_new(codec
->card
, new_name
, codec
->pcm_devs
++, playback
,
882 printk(KERN_ERR
"asoc: can't create pcm for codec %s\n", codec
->name
);
888 pcm
->private_data
= rtd
;
889 soc_pcm_ops
.mmap
= socdev
->platform
->pcm_ops
->mmap
;
890 soc_pcm_ops
.pointer
= socdev
->platform
->pcm_ops
->pointer
;
891 soc_pcm_ops
.ioctl
= socdev
->platform
->pcm_ops
->ioctl
;
892 soc_pcm_ops
.copy
= socdev
->platform
->pcm_ops
->copy
;
893 soc_pcm_ops
.silence
= socdev
->platform
->pcm_ops
->silence
;
894 soc_pcm_ops
.ack
= socdev
->platform
->pcm_ops
->ack
;
895 soc_pcm_ops
.page
= socdev
->platform
->pcm_ops
->page
;
898 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_PLAYBACK
, &soc_pcm_ops
);
901 snd_pcm_set_ops(pcm
, SNDRV_PCM_STREAM_CAPTURE
, &soc_pcm_ops
);
903 ret
= socdev
->platform
->pcm_new(codec
->card
, codec_dai
, pcm
);
905 printk(KERN_ERR
"asoc: platform pcm constructor failed\n");
910 pcm
->private_free
= socdev
->platform
->pcm_free
;
911 printk(KERN_INFO
"asoc: %s <-> %s mapping ok\n", codec_dai
->name
,
916 /* codec register dump */
917 static ssize_t
codec_reg_show(struct device
*dev
,
918 struct device_attribute
*attr
, char *buf
)
920 struct snd_soc_device
*devdata
= dev_get_drvdata(dev
);
921 struct snd_soc_codec
*codec
= devdata
->codec
;
922 int i
, step
= 1, count
= 0;
924 if (!codec
->reg_cache_size
)
927 if (codec
->reg_cache_step
)
928 step
= codec
->reg_cache_step
;
930 count
+= sprintf(buf
, "%s registers\n", codec
->name
);
931 for(i
= 0; i
< codec
->reg_cache_size
; i
+= step
)
932 count
+= sprintf(buf
+ count
, "%2x: %4x\n", i
, codec
->read(codec
, i
));
936 static DEVICE_ATTR(codec_reg
, 0444, codec_reg_show
, NULL
);
939 * snd_soc_new_ac97_codec - initailise AC97 device
940 * @codec: audio codec
941 * @ops: AC97 bus operations
942 * @num: AC97 codec number
944 * Initialises AC97 codec resources for use by ad-hoc devices only.
946 int snd_soc_new_ac97_codec(struct snd_soc_codec
*codec
,
947 struct snd_ac97_bus_ops
*ops
, int num
)
949 mutex_lock(&codec
->mutex
);
951 codec
->ac97
= kzalloc(sizeof(struct snd_ac97
), GFP_KERNEL
);
952 if (codec
->ac97
== NULL
) {
953 mutex_unlock(&codec
->mutex
);
957 codec
->ac97
->bus
= kzalloc(sizeof(struct snd_ac97_bus
), GFP_KERNEL
);
958 if (codec
->ac97
->bus
== NULL
) {
961 mutex_unlock(&codec
->mutex
);
965 codec
->ac97
->bus
->ops
= ops
;
966 codec
->ac97
->num
= num
;
967 mutex_unlock(&codec
->mutex
);
970 EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec
);
973 * snd_soc_free_ac97_codec - free AC97 codec device
974 * @codec: audio codec
976 * Frees AC97 codec device resources.
978 void snd_soc_free_ac97_codec(struct snd_soc_codec
*codec
)
980 mutex_lock(&codec
->mutex
);
981 kfree(codec
->ac97
->bus
);
984 mutex_unlock(&codec
->mutex
);
986 EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec
);
989 * snd_soc_update_bits - update codec register bits
990 * @codec: audio codec
991 * @reg: codec register
992 * @mask: register mask
995 * Writes new register value.
997 * Returns 1 for change else 0.
999 int snd_soc_update_bits(struct snd_soc_codec
*codec
, unsigned short reg
,
1000 unsigned short mask
, unsigned short value
)
1003 unsigned short old
, new;
1005 mutex_lock(&io_mutex
);
1006 old
= snd_soc_read(codec
, reg
);
1007 new = (old
& ~mask
) | value
;
1008 change
= old
!= new;
1010 snd_soc_write(codec
, reg
, new);
1012 mutex_unlock(&io_mutex
);
1015 EXPORT_SYMBOL_GPL(snd_soc_update_bits
);
1018 * snd_soc_test_bits - test register for change
1019 * @codec: audio codec
1020 * @reg: codec register
1021 * @mask: register mask
1024 * Tests a register with a new value and checks if the new value is
1025 * different from the old value.
1027 * Returns 1 for change else 0.
1029 int snd_soc_test_bits(struct snd_soc_codec
*codec
, unsigned short reg
,
1030 unsigned short mask
, unsigned short value
)
1033 unsigned short old
, new;
1035 mutex_lock(&io_mutex
);
1036 old
= snd_soc_read(codec
, reg
);
1037 new = (old
& ~mask
) | value
;
1038 change
= old
!= new;
1039 mutex_unlock(&io_mutex
);
1043 EXPORT_SYMBOL_GPL(snd_soc_test_bits
);
1046 * snd_soc_new_pcms - create new sound card and pcms
1047 * @socdev: the SoC audio device
1049 * Create a new sound card based upon the codec and interface pcms.
1051 * Returns 0 for success, else error.
1053 int snd_soc_new_pcms(struct snd_soc_device
*socdev
, int idx
, const char *xid
)
1055 struct snd_soc_codec
*codec
= socdev
->codec
;
1056 struct snd_soc_machine
*machine
= socdev
->machine
;
1059 mutex_lock(&codec
->mutex
);
1061 /* register a sound card */
1062 codec
->card
= snd_card_new(idx
, xid
, codec
->owner
, 0);
1064 printk(KERN_ERR
"asoc: can't create sound card for codec %s\n",
1066 mutex_unlock(&codec
->mutex
);
1070 codec
->card
->dev
= socdev
->dev
;
1071 codec
->card
->private_data
= codec
;
1072 strncpy(codec
->card
->driver
, codec
->name
, sizeof(codec
->card
->driver
));
1074 /* create the pcms */
1075 for(i
= 0; i
< machine
->num_links
; i
++) {
1076 ret
= soc_new_pcm(socdev
, &machine
->dai_link
[i
], i
);
1078 printk(KERN_ERR
"asoc: can't create pcm %s\n",
1079 machine
->dai_link
[i
].stream_name
);
1080 mutex_unlock(&codec
->mutex
);
1085 mutex_unlock(&codec
->mutex
);
1088 EXPORT_SYMBOL_GPL(snd_soc_new_pcms
);
1091 * snd_soc_register_card - register sound card
1092 * @socdev: the SoC audio device
1094 * Register a SoC sound card. Also registers an AC97 device if the
1095 * codec is AC97 for ad hoc devices.
1097 * Returns 0 for success, else error.
1099 int snd_soc_register_card(struct snd_soc_device
*socdev
)
1101 struct snd_soc_codec
*codec
= socdev
->codec
;
1102 struct snd_soc_machine
*machine
= socdev
->machine
;
1103 int ret
= 0, i
, ac97
= 0, err
= 0;
1105 for(i
= 0; i
< machine
->num_links
; i
++) {
1106 if (socdev
->machine
->dai_link
[i
].init
) {
1107 err
= socdev
->machine
->dai_link
[i
].init(codec
);
1109 printk(KERN_ERR
"asoc: failed to init %s\n",
1110 socdev
->machine
->dai_link
[i
].stream_name
);
1114 if (socdev
->machine
->dai_link
[i
].codec_dai
->type
==
1115 SND_SOC_DAI_AC97_BUS
)
1118 snprintf(codec
->card
->shortname
, sizeof(codec
->card
->shortname
),
1119 "%s", machine
->name
);
1120 snprintf(codec
->card
->longname
, sizeof(codec
->card
->longname
),
1121 "%s (%s)", machine
->name
, codec
->name
);
1123 ret
= snd_card_register(codec
->card
);
1125 printk(KERN_ERR
"asoc: failed to register soundcard for codec %s\n",
1130 mutex_lock(&codec
->mutex
);
1131 #ifdef CONFIG_SND_SOC_AC97_BUS
1133 ret
= soc_ac97_dev_register(codec
);
1135 printk(KERN_ERR
"asoc: AC97 device register failed\n");
1136 snd_card_free(codec
->card
);
1137 mutex_unlock(&codec
->mutex
);
1143 err
= snd_soc_dapm_sys_add(socdev
->dev
);
1145 printk(KERN_WARNING
"asoc: failed to add dapm sysfs entries\n");
1147 err
= device_create_file(socdev
->dev
, &dev_attr_codec_reg
);
1149 printk(KERN_WARNING
"asoc: failed to add codec sysfs entries\n");
1151 mutex_unlock(&codec
->mutex
);
1156 EXPORT_SYMBOL_GPL(snd_soc_register_card
);
1159 * snd_soc_free_pcms - free sound card and pcms
1160 * @socdev: the SoC audio device
1162 * Frees sound card and pcms associated with the socdev.
1163 * Also unregister the codec if it is an AC97 device.
1165 void snd_soc_free_pcms(struct snd_soc_device
*socdev
)
1167 struct snd_soc_codec
*codec
= socdev
->codec
;
1168 #ifdef CONFIG_SND_SOC_AC97_BUS
1169 struct snd_soc_codec_dai
*codec_dai
;
1173 mutex_lock(&codec
->mutex
);
1174 #ifdef CONFIG_SND_SOC_AC97_BUS
1175 for(i
= 0; i
< codec
->num_dai
; i
++) {
1176 codec_dai
= &codec
->dai
[i
];
1177 if (codec_dai
->type
== SND_SOC_DAI_AC97_BUS
&& codec
->ac97
) {
1178 soc_ac97_dev_unregister(codec
);
1186 snd_card_free(codec
->card
);
1187 device_remove_file(socdev
->dev
, &dev_attr_codec_reg
);
1188 mutex_unlock(&codec
->mutex
);
1190 EXPORT_SYMBOL_GPL(snd_soc_free_pcms
);
1193 * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
1194 * @substream: the pcm substream
1195 * @hw: the hardware parameters
1197 * Sets the substream runtime hardware parameters.
1199 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream
*substream
,
1200 const struct snd_pcm_hardware
*hw
)
1202 struct snd_pcm_runtime
*runtime
= substream
->runtime
;
1203 runtime
->hw
.info
= hw
->info
;
1204 runtime
->hw
.formats
= hw
->formats
;
1205 runtime
->hw
.period_bytes_min
= hw
->period_bytes_min
;
1206 runtime
->hw
.period_bytes_max
= hw
->period_bytes_max
;
1207 runtime
->hw
.periods_min
= hw
->periods_min
;
1208 runtime
->hw
.periods_max
= hw
->periods_max
;
1209 runtime
->hw
.buffer_bytes_max
= hw
->buffer_bytes_max
;
1210 runtime
->hw
.fifo_size
= hw
->fifo_size
;
1213 EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams
);
1216 * snd_soc_cnew - create new control
1217 * @_template: control template
1218 * @data: control private data
1219 * @lnng_name: control long name
1221 * Create a new mixer control from a template control.
1223 * Returns 0 for success, else error.
1225 struct snd_kcontrol
*snd_soc_cnew(const struct snd_kcontrol_new
*_template
,
1226 void *data
, char *long_name
)
1228 struct snd_kcontrol_new
template;
1230 memcpy(&template, _template
, sizeof(template));
1232 template.name
= long_name
;
1235 return snd_ctl_new1(&template, data
);
1237 EXPORT_SYMBOL_GPL(snd_soc_cnew
);
1240 * snd_soc_info_enum_double - enumerated double mixer info callback
1241 * @kcontrol: mixer control
1242 * @uinfo: control element information
1244 * Callback to provide information about a double enumerated
1247 * Returns 0 for success.
1249 int snd_soc_info_enum_double(struct snd_kcontrol
*kcontrol
,
1250 struct snd_ctl_elem_info
*uinfo
)
1252 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1254 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
1255 uinfo
->count
= e
->shift_l
== e
->shift_r
? 1 : 2;
1256 uinfo
->value
.enumerated
.items
= e
->mask
;
1258 if (uinfo
->value
.enumerated
.item
> e
->mask
- 1)
1259 uinfo
->value
.enumerated
.item
= e
->mask
- 1;
1260 strcpy(uinfo
->value
.enumerated
.name
,
1261 e
->texts
[uinfo
->value
.enumerated
.item
]);
1264 EXPORT_SYMBOL_GPL(snd_soc_info_enum_double
);
1267 * snd_soc_get_enum_double - enumerated double mixer get callback
1268 * @kcontrol: mixer control
1269 * @uinfo: control element information
1271 * Callback to get the value of a double enumerated mixer.
1273 * Returns 0 for success.
1275 int snd_soc_get_enum_double(struct snd_kcontrol
*kcontrol
,
1276 struct snd_ctl_elem_value
*ucontrol
)
1278 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1279 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1280 unsigned short val
, bitmask
;
1282 for (bitmask
= 1; bitmask
< e
->mask
; bitmask
<<= 1)
1284 val
= snd_soc_read(codec
, e
->reg
);
1285 ucontrol
->value
.enumerated
.item
[0] = (val
>> e
->shift_l
) & (bitmask
- 1);
1286 if (e
->shift_l
!= e
->shift_r
)
1287 ucontrol
->value
.enumerated
.item
[1] =
1288 (val
>> e
->shift_r
) & (bitmask
- 1);
1292 EXPORT_SYMBOL_GPL(snd_soc_get_enum_double
);
1295 * snd_soc_put_enum_double - enumerated double mixer put callback
1296 * @kcontrol: mixer control
1297 * @uinfo: control element information
1299 * Callback to set the value of a double enumerated mixer.
1301 * Returns 0 for success.
1303 int snd_soc_put_enum_double(struct snd_kcontrol
*kcontrol
,
1304 struct snd_ctl_elem_value
*ucontrol
)
1306 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1307 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1309 unsigned short mask
, bitmask
;
1311 for (bitmask
= 1; bitmask
< e
->mask
; bitmask
<<= 1)
1313 if (ucontrol
->value
.enumerated
.item
[0] > e
->mask
- 1)
1315 val
= ucontrol
->value
.enumerated
.item
[0] << e
->shift_l
;
1316 mask
= (bitmask
- 1) << e
->shift_l
;
1317 if (e
->shift_l
!= e
->shift_r
) {
1318 if (ucontrol
->value
.enumerated
.item
[1] > e
->mask
- 1)
1320 val
|= ucontrol
->value
.enumerated
.item
[1] << e
->shift_r
;
1321 mask
|= (bitmask
- 1) << e
->shift_r
;
1324 return snd_soc_update_bits(codec
, e
->reg
, mask
, val
);
1326 EXPORT_SYMBOL_GPL(snd_soc_put_enum_double
);
1329 * snd_soc_info_enum_ext - external enumerated single mixer info callback
1330 * @kcontrol: mixer control
1331 * @uinfo: control element information
1333 * Callback to provide information about an external enumerated
1336 * Returns 0 for success.
1338 int snd_soc_info_enum_ext(struct snd_kcontrol
*kcontrol
,
1339 struct snd_ctl_elem_info
*uinfo
)
1341 struct soc_enum
*e
= (struct soc_enum
*)kcontrol
->private_value
;
1343 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_ENUMERATED
;
1345 uinfo
->value
.enumerated
.items
= e
->mask
;
1347 if (uinfo
->value
.enumerated
.item
> e
->mask
- 1)
1348 uinfo
->value
.enumerated
.item
= e
->mask
- 1;
1349 strcpy(uinfo
->value
.enumerated
.name
,
1350 e
->texts
[uinfo
->value
.enumerated
.item
]);
1353 EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext
);
1356 * snd_soc_info_volsw_ext - external single mixer info callback
1357 * @kcontrol: mixer control
1358 * @uinfo: control element information
1360 * Callback to provide information about a single external mixer control.
1362 * Returns 0 for success.
1364 int snd_soc_info_volsw_ext(struct snd_kcontrol
*kcontrol
,
1365 struct snd_ctl_elem_info
*uinfo
)
1367 int max
= kcontrol
->private_value
;
1370 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1372 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1375 uinfo
->value
.integer
.min
= 0;
1376 uinfo
->value
.integer
.max
= max
;
1379 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext
);
1382 * snd_soc_info_volsw - single mixer info callback
1383 * @kcontrol: mixer control
1384 * @uinfo: control element information
1386 * Callback to provide information about a single mixer control.
1388 * Returns 0 for success.
1390 int snd_soc_info_volsw(struct snd_kcontrol
*kcontrol
,
1391 struct snd_ctl_elem_info
*uinfo
)
1393 int max
= (kcontrol
->private_value
>> 16) & 0xff;
1394 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1395 int rshift
= (kcontrol
->private_value
>> 12) & 0x0f;
1398 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1400 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1402 uinfo
->count
= shift
== rshift
? 1 : 2;
1403 uinfo
->value
.integer
.min
= 0;
1404 uinfo
->value
.integer
.max
= max
;
1407 EXPORT_SYMBOL_GPL(snd_soc_info_volsw
);
1410 * snd_soc_get_volsw - single mixer get callback
1411 * @kcontrol: mixer control
1412 * @uinfo: control element information
1414 * Callback to get the value of a single mixer control.
1416 * Returns 0 for success.
1418 int snd_soc_get_volsw(struct snd_kcontrol
*kcontrol
,
1419 struct snd_ctl_elem_value
*ucontrol
)
1421 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1422 int reg
= kcontrol
->private_value
& 0xff;
1423 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1424 int rshift
= (kcontrol
->private_value
>> 12) & 0x0f;
1425 int max
= (kcontrol
->private_value
>> 16) & 0xff;
1426 int mask
= (1 << fls(max
)) - 1;
1427 int invert
= (kcontrol
->private_value
>> 24) & 0x01;
1429 ucontrol
->value
.integer
.value
[0] =
1430 (snd_soc_read(codec
, reg
) >> shift
) & mask
;
1431 if (shift
!= rshift
)
1432 ucontrol
->value
.integer
.value
[1] =
1433 (snd_soc_read(codec
, reg
) >> rshift
) & mask
;
1435 ucontrol
->value
.integer
.value
[0] =
1436 max
- ucontrol
->value
.integer
.value
[0];
1437 if (shift
!= rshift
)
1438 ucontrol
->value
.integer
.value
[1] =
1439 max
- ucontrol
->value
.integer
.value
[1];
1444 EXPORT_SYMBOL_GPL(snd_soc_get_volsw
);
1447 * snd_soc_put_volsw - single mixer put callback
1448 * @kcontrol: mixer control
1449 * @uinfo: control element information
1451 * Callback to set the value of a single mixer control.
1453 * Returns 0 for success.
1455 int snd_soc_put_volsw(struct snd_kcontrol
*kcontrol
,
1456 struct snd_ctl_elem_value
*ucontrol
)
1458 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1459 int reg
= kcontrol
->private_value
& 0xff;
1460 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1461 int rshift
= (kcontrol
->private_value
>> 12) & 0x0f;
1462 int max
= (kcontrol
->private_value
>> 16) & 0xff;
1463 int mask
= (1 << fls(max
)) - 1;
1464 int invert
= (kcontrol
->private_value
>> 24) & 0x01;
1465 unsigned short val
, val2
, val_mask
;
1467 val
= (ucontrol
->value
.integer
.value
[0] & mask
);
1470 val_mask
= mask
<< shift
;
1472 if (shift
!= rshift
) {
1473 val2
= (ucontrol
->value
.integer
.value
[1] & mask
);
1476 val_mask
|= mask
<< rshift
;
1477 val
|= val2
<< rshift
;
1479 return snd_soc_update_bits(codec
, reg
, val_mask
, val
);
1481 EXPORT_SYMBOL_GPL(snd_soc_put_volsw
);
1484 * snd_soc_info_volsw_2r - double mixer info callback
1485 * @kcontrol: mixer control
1486 * @uinfo: control element information
1488 * Callback to provide information about a double mixer control that
1489 * spans 2 codec registers.
1491 * Returns 0 for success.
1493 int snd_soc_info_volsw_2r(struct snd_kcontrol
*kcontrol
,
1494 struct snd_ctl_elem_info
*uinfo
)
1496 int max
= (kcontrol
->private_value
>> 12) & 0xff;
1499 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_BOOLEAN
;
1501 uinfo
->type
= SNDRV_CTL_ELEM_TYPE_INTEGER
;
1504 uinfo
->value
.integer
.min
= 0;
1505 uinfo
->value
.integer
.max
= max
;
1508 EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r
);
1511 * snd_soc_get_volsw_2r - double mixer get callback
1512 * @kcontrol: mixer control
1513 * @uinfo: control element information
1515 * Callback to get the value of a double mixer control that spans 2 registers.
1517 * Returns 0 for success.
1519 int snd_soc_get_volsw_2r(struct snd_kcontrol
*kcontrol
,
1520 struct snd_ctl_elem_value
*ucontrol
)
1522 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1523 int reg
= kcontrol
->private_value
& 0xff;
1524 int reg2
= (kcontrol
->private_value
>> 24) & 0xff;
1525 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1526 int max
= (kcontrol
->private_value
>> 12) & 0xff;
1527 int mask
= (1<<fls(max
))-1;
1528 int invert
= (kcontrol
->private_value
>> 20) & 0x01;
1530 ucontrol
->value
.integer
.value
[0] =
1531 (snd_soc_read(codec
, reg
) >> shift
) & mask
;
1532 ucontrol
->value
.integer
.value
[1] =
1533 (snd_soc_read(codec
, reg2
) >> shift
) & mask
;
1535 ucontrol
->value
.integer
.value
[0] =
1536 max
- ucontrol
->value
.integer
.value
[0];
1537 ucontrol
->value
.integer
.value
[1] =
1538 max
- ucontrol
->value
.integer
.value
[1];
1543 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r
);
1546 * snd_soc_put_volsw_2r - double mixer set callback
1547 * @kcontrol: mixer control
1548 * @uinfo: control element information
1550 * Callback to set the value of a double mixer control that spans 2 registers.
1552 * Returns 0 for success.
1554 int snd_soc_put_volsw_2r(struct snd_kcontrol
*kcontrol
,
1555 struct snd_ctl_elem_value
*ucontrol
)
1557 struct snd_soc_codec
*codec
= snd_kcontrol_chip(kcontrol
);
1558 int reg
= kcontrol
->private_value
& 0xff;
1559 int reg2
= (kcontrol
->private_value
>> 24) & 0xff;
1560 int shift
= (kcontrol
->private_value
>> 8) & 0x0f;
1561 int max
= (kcontrol
->private_value
>> 12) & 0xff;
1562 int mask
= (1 << fls(max
)) - 1;
1563 int invert
= (kcontrol
->private_value
>> 20) & 0x01;
1565 unsigned short val
, val2
, val_mask
;
1567 val_mask
= mask
<< shift
;
1568 val
= (ucontrol
->value
.integer
.value
[0] & mask
);
1569 val2
= (ucontrol
->value
.integer
.value
[1] & mask
);
1577 val2
= val2
<< shift
;
1579 if ((err
= snd_soc_update_bits(codec
, reg
, val_mask
, val
)) < 0)
1582 err
= snd_soc_update_bits(codec
, reg2
, val_mask
, val2
);
1585 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r
);
1587 static int __devinit
snd_soc_init(void)
1589 printk(KERN_INFO
"ASoC version %s\n", SND_SOC_VERSION
);
1590 return platform_driver_register(&soc_driver
);
1593 static void snd_soc_exit(void)
1595 platform_driver_unregister(&soc_driver
);
1598 module_init(snd_soc_init
);
1599 module_exit(snd_soc_exit
);
1601 /* Module information */
1602 MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
1603 MODULE_DESCRIPTION("ALSA SoC Core");
1604 MODULE_LICENSE("GPL");
1605 MODULE_ALIAS("platform:soc-audio");