slub: Drop fallback to page allocator method
[linux-2.6/mini2440.git] / sound / soc / soc-core.c
blobe148db940cfc77ecb4521df7191909f00e672761
1 /*
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.
17 * Revision history
18 * 12th Aug 2005 Initial version.
19 * 25th Oct 2005 Working Codec, Interface and Platform registration.
21 * TODO:
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>
32 #include <linux/pm.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>
42 /* debug */
43 #define SOC_DEBUG 0
44 #if SOC_DEBUG
45 #define dbg(format, arg...) printk(format, ## arg)
46 #else
47 #define dbg(format, arg...)
48 #endif
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)
68 int ret;
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 */
75 if (ret) {
76 schedule_delayed_work(dwork, 0);
77 flush_scheduled_work();
79 return ret;
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);
88 return 0;
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)
97 int err;
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);
106 if (err < 0) {
107 snd_printk(KERN_ERR "Can't register ac97 bus\n");
108 codec->ac97->dev.bus = NULL;
109 return err;
111 return 0;
113 #endif
115 static inline const char* get_dai_name(int type)
117 switch(type) {
118 case SND_SOC_DAI_AC97_BUS:
119 case SND_SOC_DAI_AC97:
120 return "AC97";
121 case SND_SOC_DAI_I2S:
122 return "I2S";
123 case SND_SOC_DAI_PCM:
124 return "PCM";
126 return NULL;
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;
143 int ret = 0;
145 mutex_lock(&pcm_mutex);
147 /* startup the audio subsystem */
148 if (cpu_dai->ops.startup) {
149 ret = cpu_dai->ops.startup(substream);
150 if (ret < 0) {
151 printk(KERN_ERR "asoc: can't open interface %s\n",
152 cpu_dai->name);
153 goto out;
157 if (platform->pcm_ops->open) {
158 ret = platform->pcm_ops->open(substream);
159 if (ret < 0) {
160 printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
161 goto platform_err;
165 if (codec_dai->ops.startup) {
166 ret = codec_dai->ops.startup(substream);
167 if (ret < 0) {
168 printk(KERN_ERR "asoc: can't open codec %s\n",
169 codec_dai->name);
170 goto codec_dai_err;
174 if (machine->ops && machine->ops->startup) {
175 ret = machine->ops->startup(substream);
176 if (ret < 0) {
177 printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
178 goto machine_err;
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;
196 runtime->hw.rates =
197 codec_dai->playback.rates & cpu_dai->playback.rates;
198 } else {
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;
211 runtime->hw.rates =
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);
219 goto machine_err;
221 if (!runtime->hw.formats) {
222 printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
223 codec_dai->name, cpu_dai->name);
224 goto machine_err;
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);
229 goto machine_err;
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;
241 else
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);
247 return 0;
249 machine_err:
250 if (machine->ops && machine->ops->shutdown)
251 machine->ops->shutdown(substream);
253 codec_dai_err:
254 if (platform->pcm_ops->close)
255 platform->pcm_ops->close(substream);
257 platform_err:
258 if (cpu_dai->ops.shutdown)
259 cpu_dai->ops.shutdown(substream);
260 out:
261 mutex_unlock(&pcm_mutex);
262 return ret;
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;
276 int i;
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,
295 SNDRV_CTL_POWER_D1);
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
318 * shutdown.
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;
334 else
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;
341 codec->active--;
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));
361 } else {
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);
373 return 0;
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;
390 int ret = 0;
392 mutex_lock(&pcm_mutex);
394 if (machine->ops && machine->ops->prepare) {
395 ret = machine->ops->prepare(substream);
396 if (ret < 0) {
397 printk(KERN_ERR "asoc: machine prepare error\n");
398 goto out;
402 if (platform->pcm_ops->prepare) {
403 ret = platform->pcm_ops->prepare(substream);
404 if (ret < 0) {
405 printk(KERN_ERR "asoc: platform prepare error\n");
406 goto out;
410 if (codec_dai->ops.prepare) {
411 ret = codec_dai->ops.prepare(substream);
412 if (ret < 0) {
413 printk(KERN_ERR "asoc: codec DAI prepare error\n");
414 goto out;
418 if (cpu_dai->ops.prepare) {
419 ret = cpu_dai->ops.prepare(substream);
420 if (ret < 0) {
421 printk(KERN_ERR "asoc: cpu DAI prepare error\n");
422 goto out;
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);
434 else {
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);
440 } else {
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);
450 else
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);
459 } else {
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);
465 else
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);
474 out:
475 mutex_unlock(&pcm_mutex);
476 return ret;
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;
493 int ret = 0;
495 mutex_lock(&pcm_mutex);
497 if (machine->ops && machine->ops->hw_params) {
498 ret = machine->ops->hw_params(substream, params);
499 if (ret < 0) {
500 printk(KERN_ERR "asoc: machine hw_params failed\n");
501 goto out;
505 if (codec_dai->ops.hw_params) {
506 ret = codec_dai->ops.hw_params(substream, params);
507 if (ret < 0) {
508 printk(KERN_ERR "asoc: can't set codec %s hw params\n",
509 codec_dai->name);
510 goto codec_err;
514 if (cpu_dai->ops.hw_params) {
515 ret = cpu_dai->ops.hw_params(substream, params);
516 if (ret < 0) {
517 printk(KERN_ERR "asoc: can't set interface %s hw params\n",
518 cpu_dai->name);
519 goto interface_err;
523 if (platform->pcm_ops->hw_params) {
524 ret = platform->pcm_ops->hw_params(substream, params);
525 if (ret < 0) {
526 printk(KERN_ERR "asoc: can't set platform %s hw params\n",
527 platform->name);
528 goto platform_err;
532 out:
533 mutex_unlock(&pcm_mutex);
534 return ret;
536 platform_err:
537 if (cpu_dai->ops.hw_free)
538 cpu_dai->ops.hw_free(substream);
540 interface_err:
541 if (codec_dai->ops.hw_free)
542 codec_dai->ops.hw_free(substream);
544 codec_err:
545 if(machine->ops && machine->ops->hw_free)
546 machine->ops->hw_free(substream);
548 mutex_unlock(&pcm_mutex);
549 return ret;
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);
587 return 0;
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;
598 int ret;
600 if (codec_dai->ops.trigger) {
601 ret = codec_dai->ops.trigger(substream, cmd);
602 if (ret < 0)
603 return ret;
606 if (platform->pcm_ops->trigger) {
607 ret = platform->pcm_ops->trigger(substream, cmd);
608 if (ret < 0)
609 return ret;
612 if (cpu_dai->ops.trigger) {
613 ret = cpu_dai->ops.trigger(substream, cmd);
614 if (ret < 0)
615 return ret;
617 return 0;
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,
630 #ifdef CONFIG_PM
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;
639 int i;
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;
669 if (stream != NULL)
670 snd_soc_dapm_stream_event(codec, stream,
671 SND_SOC_DAPM_STREAM_SUSPEND);
672 stream = codec->dai[i].capture.stream_name;
673 if (stream != NULL)
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);
690 return 0;
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;
701 int i;
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;
717 if (stream != NULL)
718 snd_soc_dapm_stream_event(codec, stream,
719 SND_SOC_DAPM_STREAM_RESUME);
720 stream = codec->dai[i].capture.stream_name;
721 if (stream != NULL)
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);
744 return 0;
747 #else
748 #define soc_suspend NULL
749 #define soc_resume NULL
750 #endif
752 /* probes a new socdev */
753 static int soc_probe(struct platform_device *pdev)
755 int ret = 0, i;
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);
763 if(ret < 0)
764 return ret;
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);
771 if(ret < 0)
772 goto cpu_dai_err;
776 if (codec_dev->probe) {
777 ret = codec_dev->probe(pdev);
778 if(ret < 0)
779 goto cpu_dai_err;
782 if (platform->probe) {
783 ret = platform->probe(pdev);
784 if(ret < 0)
785 goto platform_err;
788 /* DAPM stream work */
789 INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
790 return 0;
792 platform_err:
793 if (codec_dev->remove)
794 codec_dev->remove(pdev);
796 cpu_dai_err:
797 for (i--; i >= 0; i--) {
798 struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
799 if (cpu_dai->remove)
800 cpu_dai->remove(pdev);
803 if (machine->remove)
804 machine->remove(pdev);
806 return ret;
809 /* removes a socdev */
810 static int soc_remove(struct platform_device *pdev)
812 int i;
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;
828 if (cpu_dai->remove)
829 cpu_dai->remove(pdev);
832 if (machine->remove)
833 machine->remove(pdev);
835 return 0;
838 /* ASoC platform driver */
839 static struct platform_driver soc_driver = {
840 .driver = {
841 .name = "soc-audio",
842 .owner = THIS_MODULE,
844 .probe = soc_probe,
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;
858 struct snd_pcm *pcm;
859 char new_name[64];
860 int ret = 0, playback = 0, capture = 0;
862 rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
863 if (rtd == NULL)
864 return -ENOMEM;
866 rtd->dai = dai_link;
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)
875 playback = 1;
876 if (codec_dai->capture.channels_min)
877 capture = 1;
879 ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
880 capture, &pcm);
881 if (ret < 0) {
882 printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
883 kfree(rtd);
884 return ret;
887 dai_link->pcm = pcm;
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;
897 if (playback)
898 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);
900 if (capture)
901 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);
903 ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
904 if (ret < 0) {
905 printk(KERN_ERR "asoc: platform pcm constructor failed\n");
906 kfree(rtd);
907 return ret;
910 pcm->private_free = socdev->platform->pcm_free;
911 printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
912 cpu_dai->name);
913 return ret;
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)
925 return 0;
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));
934 return count;
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);
954 return -ENOMEM;
957 codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
958 if (codec->ac97->bus == NULL) {
959 kfree(codec->ac97);
960 codec->ac97 = NULL;
961 mutex_unlock(&codec->mutex);
962 return -ENOMEM;
965 codec->ac97->bus->ops = ops;
966 codec->ac97->num = num;
967 mutex_unlock(&codec->mutex);
968 return 0;
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);
982 kfree(codec->ac97);
983 codec->ac97 = NULL;
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
993 * @value: new value
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)
1002 int change;
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;
1009 if (change)
1010 snd_soc_write(codec, reg, new);
1012 mutex_unlock(&io_mutex);
1013 return change;
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
1022 * @value: new value
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)
1032 int change;
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);
1041 return change;
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;
1057 int ret = 0, i;
1059 mutex_lock(&codec->mutex);
1061 /* register a sound card */
1062 codec->card = snd_card_new(idx, xid, codec->owner, 0);
1063 if (!codec->card) {
1064 printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
1065 codec->name);
1066 mutex_unlock(&codec->mutex);
1067 return -ENODEV;
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);
1077 if (ret < 0) {
1078 printk(KERN_ERR "asoc: can't create pcm %s\n",
1079 machine->dai_link[i].stream_name);
1080 mutex_unlock(&codec->mutex);
1081 return ret;
1085 mutex_unlock(&codec->mutex);
1086 return ret;
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);
1108 if (err < 0) {
1109 printk(KERN_ERR "asoc: failed to init %s\n",
1110 socdev->machine->dai_link[i].stream_name);
1111 continue;
1114 if (socdev->machine->dai_link[i].codec_dai->type ==
1115 SND_SOC_DAI_AC97_BUS)
1116 ac97 = 1;
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);
1124 if (ret < 0) {
1125 printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n",
1126 codec->name);
1127 goto out;
1130 mutex_lock(&codec->mutex);
1131 #ifdef CONFIG_SND_SOC_AC97_BUS
1132 if (ac97) {
1133 ret = soc_ac97_dev_register(codec);
1134 if (ret < 0) {
1135 printk(KERN_ERR "asoc: AC97 device register failed\n");
1136 snd_card_free(codec->card);
1137 mutex_unlock(&codec->mutex);
1138 goto out;
1141 #endif
1143 err = snd_soc_dapm_sys_add(socdev->dev);
1144 if (err < 0)
1145 printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");
1147 err = device_create_file(socdev->dev, &dev_attr_codec_reg);
1148 if (err < 0)
1149 printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n");
1151 mutex_unlock(&codec->mutex);
1153 out:
1154 return ret;
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;
1170 int i;
1171 #endif
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);
1179 goto free_card;
1182 free_card:
1183 #endif
1185 if (codec->card)
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;
1211 return 0;
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));
1231 if (long_name)
1232 template.name = long_name;
1233 template.index = 0;
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
1245 * mixer control.
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]);
1262 return 0;
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);
1290 return 0;
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;
1308 unsigned short val;
1309 unsigned short mask, bitmask;
1311 for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
1313 if (ucontrol->value.enumerated.item[0] > e->mask - 1)
1314 return -EINVAL;
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)
1319 return -EINVAL;
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
1334 * single mixer.
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;
1344 uinfo->count = 1;
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]);
1351 return 0;
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;
1369 if (max == 1)
1370 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1371 else
1372 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1374 uinfo->count = 1;
1375 uinfo->value.integer.min = 0;
1376 uinfo->value.integer.max = max;
1377 return 0;
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;
1397 if (max == 1)
1398 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1399 else
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;
1405 return 0;
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;
1434 if (invert) {
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];
1442 return 0;
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);
1468 if (invert)
1469 val = max - val;
1470 val_mask = mask << shift;
1471 val = val << shift;
1472 if (shift != rshift) {
1473 val2 = (ucontrol->value.integer.value[1] & mask);
1474 if (invert)
1475 val2 = max - val2;
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;
1498 if (max == 1)
1499 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1500 else
1501 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1503 uinfo->count = 2;
1504 uinfo->value.integer.min = 0;
1505 uinfo->value.integer.max = max;
1506 return 0;
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;
1534 if (invert) {
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];
1541 return 0;
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;
1564 int err;
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);
1571 if (invert) {
1572 val = max - val;
1573 val2 = max - val2;
1576 val = val << shift;
1577 val2 = val2 << shift;
1579 if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0)
1580 return err;
1582 err = snd_soc_update_bits(codec, reg2, val_mask, val2);
1583 return err;
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");