soc-core.c 53.6 KB
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/*
 * soc-core.c  --  ALSA SoC Audio Layer
 *
 * Copyright 2005 Wolfson Microelectronics PLC.
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 * Copyright 2005 Openedhand Ltd.
 *
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 * Author: Liam Girdwood <lrg@slimlogic.co.uk>
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 *         with code, comments and ideas from :-
 *         Richard Purdie <richard@openedhand.com>
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 *
 *  This program is free software; you can redistribute  it and/or modify it
 *  under  the terms of  the GNU General  Public License as published by the
 *  Free Software Foundation;  either version 2 of the  License, or (at your
 *  option) any later version.
 *
 *  TODO:
 *   o Add hw rules to enforce rates, etc.
 *   o More testing with other codecs/machines.
 *   o Add more codecs and platforms to ensure good API coverage.
 *   o Support TDM on PCM and I2S
 */

#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
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#include <linux/debugfs.h>
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#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>

static DEFINE_MUTEX(pcm_mutex);
static DEFINE_MUTEX(io_mutex);
static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);

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#ifdef CONFIG_DEBUG_FS
static struct dentry *debugfs_root;
#endif

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static DEFINE_MUTEX(client_mutex);
static LIST_HEAD(card_list);

static int snd_soc_register_card(struct snd_soc_card *card);
static int snd_soc_unregister_card(struct snd_soc_card *card);

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/*
 * This is a timeout to do a DAPM powerdown after a stream is closed().
 * It can be used to eliminate pops between different playback streams, e.g.
 * between two audio tracks.
 */
static int pmdown_time = 5000;
module_param(pmdown_time, int, 0);
MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)");

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/*
 * This function forces any delayed work to be queued and run.
 */
static int run_delayed_work(struct delayed_work *dwork)
{
	int ret;

	/* cancel any work waiting to be queued. */
	ret = cancel_delayed_work(dwork);

	/* if there was any work waiting then we run it now and
	 * wait for it's completion */
	if (ret) {
		schedule_delayed_work(dwork, 0);
		flush_scheduled_work();
	}
	return ret;
}

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#ifdef CONFIG_SND_SOC_AC97_BUS
/* unregister ac97 codec */
static int soc_ac97_dev_unregister(struct snd_soc_codec *codec)
{
	if (codec->ac97->dev.bus)
		device_unregister(&codec->ac97->dev);
	return 0;
}

/* stop no dev release warning */
static void soc_ac97_device_release(struct device *dev){}

/* register ac97 codec to bus */
static int soc_ac97_dev_register(struct snd_soc_codec *codec)
{
	int err;

	codec->ac97->dev.bus = &ac97_bus_type;
	codec->ac97->dev.parent = NULL;
	codec->ac97->dev.release = soc_ac97_device_release;

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	dev_set_name(&codec->ac97->dev, "%d-%d:%s",
		     codec->card->number, 0, codec->name);
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	err = device_register(&codec->ac97->dev);
	if (err < 0) {
		snd_printk(KERN_ERR "Can't register ac97 bus\n");
		codec->ac97->dev.bus = NULL;
		return err;
	}
	return 0;
}
#endif

/*
 * Called by ALSA when a PCM substream is opened, the runtime->hw record is
 * then initialized and any private data can be allocated. This also calls
 * startup for the cpu DAI, platform, machine and codec DAI.
 */
static int soc_pcm_open(struct snd_pcm_substream *substream)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
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	struct snd_soc_card *card = socdev->card;
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	struct snd_pcm_runtime *runtime = substream->runtime;
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	struct snd_soc_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = card->platform;
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	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
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	int ret = 0;

	mutex_lock(&pcm_mutex);

	/* startup the audio subsystem */
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	if (cpu_dai->ops.startup) {
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		ret = cpu_dai->ops.startup(substream, cpu_dai);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: can't open interface %s\n",
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				cpu_dai->name);
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			goto out;
		}
	}

	if (platform->pcm_ops->open) {
		ret = platform->pcm_ops->open(substream);
		if (ret < 0) {
			printk(KERN_ERR "asoc: can't open platform %s\n", platform->name);
			goto platform_err;
		}
	}

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	if (codec_dai->ops.startup) {
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		ret = codec_dai->ops.startup(substream, codec_dai);
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		if (ret < 0) {
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			printk(KERN_ERR "asoc: can't open codec %s\n",
				codec_dai->name);
			goto codec_dai_err;
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		}
	}

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	if (machine->ops && machine->ops->startup) {
		ret = machine->ops->startup(substream);
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		if (ret < 0) {
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			printk(KERN_ERR "asoc: %s startup failed\n", machine->name);
			goto machine_err;
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		}
	}

	/* Check that the codec and cpu DAI's are compatible */
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		runtime->hw.rate_min =
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			max(codec_dai->playback.rate_min,
			    cpu_dai->playback.rate_min);
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		runtime->hw.rate_max =
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			min(codec_dai->playback.rate_max,
			    cpu_dai->playback.rate_max);
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		runtime->hw.channels_min =
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			max(codec_dai->playback.channels_min,
				cpu_dai->playback.channels_min);
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		runtime->hw.channels_max =
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			min(codec_dai->playback.channels_max,
				cpu_dai->playback.channels_max);
		runtime->hw.formats =
			codec_dai->playback.formats & cpu_dai->playback.formats;
		runtime->hw.rates =
			codec_dai->playback.rates & cpu_dai->playback.rates;
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	} else {
		runtime->hw.rate_min =
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			max(codec_dai->capture.rate_min,
			    cpu_dai->capture.rate_min);
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		runtime->hw.rate_max =
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			min(codec_dai->capture.rate_max,
			    cpu_dai->capture.rate_max);
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		runtime->hw.channels_min =
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			max(codec_dai->capture.channels_min,
				cpu_dai->capture.channels_min);
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		runtime->hw.channels_max =
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			min(codec_dai->capture.channels_max,
				cpu_dai->capture.channels_max);
		runtime->hw.formats =
			codec_dai->capture.formats & cpu_dai->capture.formats;
		runtime->hw.rates =
			codec_dai->capture.rates & cpu_dai->capture.rates;
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	}

	snd_pcm_limit_hw_rates(runtime);
	if (!runtime->hw.rates) {
		printk(KERN_ERR "asoc: %s <-> %s No matching rates\n",
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			codec_dai->name, cpu_dai->name);
		goto machine_err;
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	}
	if (!runtime->hw.formats) {
		printk(KERN_ERR "asoc: %s <-> %s No matching formats\n",
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			codec_dai->name, cpu_dai->name);
		goto machine_err;
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	}
	if (!runtime->hw.channels_min || !runtime->hw.channels_max) {
		printk(KERN_ERR "asoc: %s <-> %s No matching channels\n",
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			codec_dai->name, cpu_dai->name);
		goto machine_err;
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	}

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	pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
	pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates);
	pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
		 runtime->hw.channels_max);
	pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
		 runtime->hw.rate_max);
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	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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		cpu_dai->playback.active = codec_dai->playback.active = 1;
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	else
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		cpu_dai->capture.active = codec_dai->capture.active = 1;
	cpu_dai->active = codec_dai->active = 1;
	cpu_dai->runtime = runtime;
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	socdev->codec->active++;
	mutex_unlock(&pcm_mutex);
	return 0;

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machine_err:
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	if (machine->ops && machine->ops->shutdown)
		machine->ops->shutdown(substream);

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codec_dai_err:
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	if (platform->pcm_ops->close)
		platform->pcm_ops->close(substream);

platform_err:
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	if (cpu_dai->ops.shutdown)
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		cpu_dai->ops.shutdown(substream, cpu_dai);
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out:
	mutex_unlock(&pcm_mutex);
	return ret;
}

/*
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 * Power down the audio subsystem pmdown_time msecs after close is called.
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 * This is to ensure there are no pops or clicks in between any music tracks
 * due to DAPM power cycling.
 */
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static void close_delayed_work(struct work_struct *work)
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{
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	struct snd_soc_card *card = container_of(work, struct snd_soc_card,
						 delayed_work.work);
	struct snd_soc_device *socdev = card->socdev;
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	struct snd_soc_codec *codec = socdev->codec;
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	struct snd_soc_dai *codec_dai;
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	int i;

	mutex_lock(&pcm_mutex);
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	for (i = 0; i < codec->num_dai; i++) {
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		codec_dai = &codec->dai[i];

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		pr_debug("pop wq checking: %s status: %s waiting: %s\n",
			 codec_dai->playback.stream_name,
			 codec_dai->playback.active ? "active" : "inactive",
			 codec_dai->pop_wait ? "yes" : "no");
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		/* are we waiting on this codec DAI stream */
		if (codec_dai->pop_wait == 1) {

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			/* Reduce power if no longer active */
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			if (codec->active == 0) {
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				pr_debug("pop wq D1 %s %s\n", codec->name,
					 codec_dai->playback.stream_name);
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				snd_soc_dapm_set_bias_level(socdev,
					SND_SOC_BIAS_PREPARE);
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			}

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			codec_dai->pop_wait = 0;
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			snd_soc_dapm_stream_event(codec,
				codec_dai->playback.stream_name,
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				SND_SOC_DAPM_STREAM_STOP);

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			/* Fall into standby if no longer active */
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			if (codec->active == 0) {
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				pr_debug("pop wq D3 %s %s\n", codec->name,
					 codec_dai->playback.stream_name);
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				snd_soc_dapm_set_bias_level(socdev,
					SND_SOC_BIAS_STANDBY);
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			}
		}
	}
	mutex_unlock(&pcm_mutex);
}

/*
 * Called by ALSA when a PCM substream is closed. Private data can be
 * freed here. The cpu DAI, codec DAI, machine and platform are also
 * shutdown.
 */
static int soc_codec_close(struct snd_pcm_substream *substream)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
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	struct snd_soc_card *card = socdev->card;
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	struct snd_soc_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = card->platform;
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	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
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	struct snd_soc_codec *codec = socdev->codec;

	mutex_lock(&pcm_mutex);

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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		cpu_dai->playback.active = codec_dai->playback.active = 0;
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	else
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		cpu_dai->capture.active = codec_dai->capture.active = 0;
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	if (codec_dai->playback.active == 0 &&
		codec_dai->capture.active == 0) {
		cpu_dai->active = codec_dai->active = 0;
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	}
	codec->active--;

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	/* Muting the DAC suppresses artifacts caused during digital
	 * shutdown, for example from stopping clocks.
	 */
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		snd_soc_dai_digital_mute(codec_dai, 1);

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	if (cpu_dai->ops.shutdown)
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		cpu_dai->ops.shutdown(substream, cpu_dai);
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	if (codec_dai->ops.shutdown)
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		codec_dai->ops.shutdown(substream, codec_dai);
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	if (machine->ops && machine->ops->shutdown)
		machine->ops->shutdown(substream);

	if (platform->pcm_ops->close)
		platform->pcm_ops->close(substream);
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	cpu_dai->runtime = NULL;
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	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		/* start delayed pop wq here for playback streams */
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		codec_dai->pop_wait = 1;
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		schedule_delayed_work(&card->delayed_work,
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			msecs_to_jiffies(pmdown_time));
	} else {
		/* capture streams can be powered down now */
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		snd_soc_dapm_stream_event(codec,
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			codec_dai->capture.stream_name,
			SND_SOC_DAPM_STREAM_STOP);
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		if (codec->active == 0 && codec_dai->pop_wait == 0)
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			snd_soc_dapm_set_bias_level(socdev,
						SND_SOC_BIAS_STANDBY);
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	}

	mutex_unlock(&pcm_mutex);
	return 0;
}

/*
 * Called by ALSA when the PCM substream is prepared, can set format, sample
 * rate, etc.  This function is non atomic and can be called multiple times,
 * it can refer to the runtime info.
 */
static int soc_pcm_prepare(struct snd_pcm_substream *substream)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
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	struct snd_soc_card *card = socdev->card;
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	struct snd_soc_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = card->platform;
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	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
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	struct snd_soc_codec *codec = socdev->codec;
	int ret = 0;

	mutex_lock(&pcm_mutex);
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	if (machine->ops && machine->ops->prepare) {
		ret = machine->ops->prepare(substream);
		if (ret < 0) {
			printk(KERN_ERR "asoc: machine prepare error\n");
			goto out;
		}
	}

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	if (platform->pcm_ops->prepare) {
		ret = platform->pcm_ops->prepare(substream);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: platform prepare error\n");
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			goto out;
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		}
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	}

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	if (codec_dai->ops.prepare) {
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		ret = codec_dai->ops.prepare(substream, codec_dai);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: codec DAI prepare error\n");
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			goto out;
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		}
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	}

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	if (cpu_dai->ops.prepare) {
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		ret = cpu_dai->ops.prepare(substream, cpu_dai);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
			goto out;
		}
	}
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	/* cancel any delayed stream shutdown that is pending */
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
	    codec_dai->pop_wait) {
		codec_dai->pop_wait = 0;
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		cancel_delayed_work(&card->delayed_work);
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	}
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	/* do we need to power up codec */
	if (codec->bias_level != SND_SOC_BIAS_ON) {
		snd_soc_dapm_set_bias_level(socdev,
					    SND_SOC_BIAS_PREPARE);
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		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
			snd_soc_dapm_stream_event(codec,
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					codec_dai->playback.stream_name,
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					SND_SOC_DAPM_STREAM_START);
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		else
			snd_soc_dapm_stream_event(codec,
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					codec_dai->capture.stream_name,
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					SND_SOC_DAPM_STREAM_START);

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		snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
		snd_soc_dai_digital_mute(codec_dai, 0);
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	} else {
		/* codec already powered - power on widgets */
		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
			snd_soc_dapm_stream_event(codec,
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					codec_dai->playback.stream_name,
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					SND_SOC_DAPM_STREAM_START);
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		else
			snd_soc_dapm_stream_event(codec,
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					codec_dai->capture.stream_name,
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					SND_SOC_DAPM_STREAM_START);
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		snd_soc_dai_digital_mute(codec_dai, 0);
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	}

out:
	mutex_unlock(&pcm_mutex);
	return ret;
}

/*
 * Called by ALSA when the hardware params are set by application. This
 * function can also be called multiple times and can allocate buffers
 * (using snd_pcm_lib_* ). It's non-atomic.
 */
static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
				struct snd_pcm_hw_params *params)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
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	struct snd_soc_dai_link *machine = rtd->dai;
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	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
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	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
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	int ret = 0;

	mutex_lock(&pcm_mutex);

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	if (machine->ops && machine->ops->hw_params) {
		ret = machine->ops->hw_params(substream, params);
		if (ret < 0) {
			printk(KERN_ERR "asoc: machine hw_params failed\n");
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			goto out;
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		}
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	}

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	if (codec_dai->ops.hw_params) {
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		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
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				codec_dai->name);
			goto codec_err;
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		}
	}

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	if (cpu_dai->ops.hw_params) {
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		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
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		if (ret < 0) {
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			printk(KERN_ERR "asoc: interface %s hw params failed\n",
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				cpu_dai->name);
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			goto interface_err;
		}
	}

	if (platform->pcm_ops->hw_params) {
		ret = platform->pcm_ops->hw_params(substream, params);
		if (ret < 0) {
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			printk(KERN_ERR "asoc: platform %s hw params failed\n",
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				platform->name);
			goto platform_err;
		}
	}

out:
	mutex_unlock(&pcm_mutex);
	return ret;

platform_err:
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	if (cpu_dai->ops.hw_free)
527
		cpu_dai->ops.hw_free(substream, cpu_dai);
F
Frank Mandarino 已提交
528 529

interface_err:
530
	if (codec_dai->ops.hw_free)
531
		codec_dai->ops.hw_free(substream, codec_dai);
532 533

codec_err:
534
	if (machine->ops && machine->ops->hw_free)
535
		machine->ops->hw_free(substream);
F
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536 537 538 539 540 541 542 543 544 545 546 547

	mutex_unlock(&pcm_mutex);
	return ret;
}

/*
 * Free's resources allocated by hw_params, can be called multiple times
 */
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
548
	struct snd_soc_dai_link *machine = rtd->dai;
549 550
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
551 552
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
F
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553 554 555 556 557
	struct snd_soc_codec *codec = socdev->codec;

	mutex_lock(&pcm_mutex);

	/* apply codec digital mute */
558 559
	if (!codec->active)
		snd_soc_dai_digital_mute(codec_dai, 1);
F
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560 561 562 563 564 565 566 567 568 569

	/* free any machine hw params */
	if (machine->ops && machine->ops->hw_free)
		machine->ops->hw_free(substream);

	/* free any DMA resources */
	if (platform->pcm_ops->hw_free)
		platform->pcm_ops->hw_free(substream);

	/* now free hw params for the DAI's  */
570
	if (codec_dai->ops.hw_free)
571
		codec_dai->ops.hw_free(substream, codec_dai);
F
Frank Mandarino 已提交
572

573
	if (cpu_dai->ops.hw_free)
574
		cpu_dai->ops.hw_free(substream, cpu_dai);
F
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575 576 577 578 579 580 581 582 583

	mutex_unlock(&pcm_mutex);
	return 0;
}

static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
584
	struct snd_soc_card *card= socdev->card;
585
	struct snd_soc_dai_link *machine = rtd->dai;
586
	struct snd_soc_platform *platform = card->platform;
587 588
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
F
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	int ret;

591
	if (codec_dai->ops.trigger) {
592
		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
F
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593 594 595 596 597 598 599 600 601 602
		if (ret < 0)
			return ret;
	}

	if (platform->pcm_ops->trigger) {
		ret = platform->pcm_ops->trigger(substream, cmd);
		if (ret < 0)
			return ret;
	}

603
	if (cpu_dai->ops.trigger) {
604
		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
F
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605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624
		if (ret < 0)
			return ret;
	}
	return 0;
}

/* ASoC PCM operations */
static struct snd_pcm_ops soc_pcm_ops = {
	.open		= soc_pcm_open,
	.close		= soc_codec_close,
	.hw_params	= soc_pcm_hw_params,
	.hw_free	= soc_pcm_hw_free,
	.prepare	= soc_pcm_prepare,
	.trigger	= soc_pcm_trigger,
};

#ifdef CONFIG_PM
/* powers down audio subsystem for suspend */
static int soc_suspend(struct platform_device *pdev, pm_message_t state)
{
625
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
626
	struct snd_soc_card *card = socdev->card;
627
	struct snd_soc_platform *platform = card->platform;
628
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
F
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629 630 631
	struct snd_soc_codec *codec = socdev->codec;
	int i;

632 633 634 635 636 637 638 639 640 641
	/* Due to the resume being scheduled into a workqueue we could
	* suspend before that's finished - wait for it to complete.
	 */
	snd_power_lock(codec->card);
	snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
	snd_power_unlock(codec->card);

	/* we're going to block userspace touching us until resume completes */
	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot);

F
Frank Mandarino 已提交
642
	/* mute any active DAC's */
643 644 645 646
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
		if (dai->ops.digital_mute && dai->playback.active)
			dai->ops.digital_mute(dai, 1);
F
Frank Mandarino 已提交
647 648
	}

649
	/* suspend all pcms */
650 651
	for (i = 0; i < card->num_links; i++)
		snd_pcm_suspend_all(card->dai_link[i].pcm);
652

653 654
	if (card->suspend_pre)
		card->suspend_pre(pdev, state);
F
Frank Mandarino 已提交
655

656 657
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
M
Mark Brown 已提交
658
		if (cpu_dai->suspend && !cpu_dai->ac97_control)
659
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
660
		if (platform->suspend)
661
			platform->suspend(cpu_dai);
F
Frank Mandarino 已提交
662 663 664
	}

	/* close any waiting streams and save state */
665
	run_delayed_work(&card->delayed_work);
666
	codec->suspend_bias_level = codec->bias_level;
F
Frank Mandarino 已提交
667

668
	for (i = 0; i < codec->num_dai; i++) {
F
Frank Mandarino 已提交
669 670 671 672 673 674 675 676 677 678 679 680 681
		char *stream = codec->dai[i].playback.stream_name;
		if (stream != NULL)
			snd_soc_dapm_stream_event(codec, stream,
				SND_SOC_DAPM_STREAM_SUSPEND);
		stream = codec->dai[i].capture.stream_name;
		if (stream != NULL)
			snd_soc_dapm_stream_event(codec, stream,
				SND_SOC_DAPM_STREAM_SUSPEND);
	}

	if (codec_dev->suspend)
		codec_dev->suspend(pdev, state);

682 683
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
M
Mark Brown 已提交
684
		if (cpu_dai->suspend && cpu_dai->ac97_control)
685
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
686 687
	}

688 689
	if (card->suspend_post)
		card->suspend_post(pdev, state);
F
Frank Mandarino 已提交
690 691 692 693

	return 0;
}

694 695 696 697
/* deferred resume work, so resume can complete before we finished
 * setting our codec back up, which can be very slow on I2C
 */
static void soc_resume_deferred(struct work_struct *work)
F
Frank Mandarino 已提交
698
{
699 700 701 702
	struct snd_soc_card *card = container_of(work,
						 struct snd_soc_card,
						 deferred_resume_work);
	struct snd_soc_device *socdev = card->socdev;
703
	struct snd_soc_platform *platform = card->platform;
704
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
F
Frank Mandarino 已提交
705
	struct snd_soc_codec *codec = socdev->codec;
706
	struct platform_device *pdev = to_platform_device(socdev->dev);
F
Frank Mandarino 已提交
707 708
	int i;

709 710 711 712
	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
	 * so userspace apps are blocked from touching us
	 */

713
	dev_dbg(socdev->dev, "starting resume work\n");
714

715 716
	if (card->resume_pre)
		card->resume_pre(pdev);
F
Frank Mandarino 已提交
717

718 719
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
M
Mark Brown 已提交
720
		if (cpu_dai->resume && cpu_dai->ac97_control)
721
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
722 723 724 725 726
	}

	if (codec_dev->resume)
		codec_dev->resume(pdev);

727 728
	for (i = 0; i < codec->num_dai; i++) {
		char *stream = codec->dai[i].playback.stream_name;
F
Frank Mandarino 已提交
729 730 731 732 733 734 735 736 737
		if (stream != NULL)
			snd_soc_dapm_stream_event(codec, stream,
				SND_SOC_DAPM_STREAM_RESUME);
		stream = codec->dai[i].capture.stream_name;
		if (stream != NULL)
			snd_soc_dapm_stream_event(codec, stream,
				SND_SOC_DAPM_STREAM_RESUME);
	}

738
	/* unmute any active DACs */
739 740 741 742
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
		if (dai->ops.digital_mute && dai->playback.active)
			dai->ops.digital_mute(dai, 0);
F
Frank Mandarino 已提交
743 744
	}

745 746
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
M
Mark Brown 已提交
747
		if (cpu_dai->resume && !cpu_dai->ac97_control)
748
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
749
		if (platform->resume)
750
			platform->resume(cpu_dai);
F
Frank Mandarino 已提交
751 752
	}

753 754
	if (card->resume_post)
		card->resume_post(pdev);
F
Frank Mandarino 已提交
755

756
	dev_dbg(socdev->dev, "resume work completed\n");
757 758 759 760 761 762 763 764 765

	/* userspace can access us now we are back as we were before */
	snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0);
}

/* powers up audio subsystem after a suspend */
static int soc_resume(struct platform_device *pdev)
{
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
766
	struct snd_soc_card *card = socdev->card;
767

768
	dev_dbg(socdev->dev, "scheduling resume work\n");
769

770
	if (!schedule_work(&card->deferred_resume_work))
771
		dev_err(socdev->dev, "resume work item may be lost\n");
772

F
Frank Mandarino 已提交
773 774 775 776 777 778 779 780 781 782 783 784 785
	return 0;
}

#else
#define soc_suspend	NULL
#define soc_resume	NULL
#endif

/* probes a new socdev */
static int soc_probe(struct platform_device *pdev)
{
	int ret = 0, i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
786
	struct snd_soc_card *card = socdev->card;
787
	struct snd_soc_platform *platform = card->platform;
F
Frank Mandarino 已提交
788 789
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

790 791 792
	/* Bodge while we push things out of socdev */
	card->socdev = socdev;

M
Mark Brown 已提交
793 794 795 796 797 798 799 800
	/* Bodge while we unpick instantiation */
	card->dev = &pdev->dev;
	ret = snd_soc_register_card(card);
	if (ret != 0) {
		dev_err(&pdev->dev, "Failed to register card\n");
		return ret;
	}

801 802
	if (card->probe) {
		ret = card->probe(pdev);
803
		if (ret < 0)
F
Frank Mandarino 已提交
804 805 806
			return ret;
	}

807 808
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
809
		if (cpu_dai->probe) {
810
			ret = cpu_dai->probe(pdev, cpu_dai);
811
			if (ret < 0)
F
Frank Mandarino 已提交
812 813 814 815 816 817
				goto cpu_dai_err;
		}
	}

	if (codec_dev->probe) {
		ret = codec_dev->probe(pdev);
818
		if (ret < 0)
F
Frank Mandarino 已提交
819 820 821 822 823
			goto cpu_dai_err;
	}

	if (platform->probe) {
		ret = platform->probe(pdev);
824
		if (ret < 0)
F
Frank Mandarino 已提交
825 826 827 828
			goto platform_err;
	}

	/* DAPM stream work */
829
	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
R
Randy Dunlap 已提交
830
#ifdef CONFIG_PM
831
	/* deferred resume work */
832
	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
R
Randy Dunlap 已提交
833
#endif
834

F
Frank Mandarino 已提交
835 836 837 838 839 840 841
	return 0;

platform_err:
	if (codec_dev->remove)
		codec_dev->remove(pdev);

cpu_dai_err:
842
	for (i--; i >= 0; i--) {
843
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
844
		if (cpu_dai->remove)
845
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
846 847
	}

848 849
	if (card->remove)
		card->remove(pdev);
F
Frank Mandarino 已提交
850 851 852 853 854 855 856 857 858

	return ret;
}

/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
	int i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
859
	struct snd_soc_card *card = socdev->card;
860
	struct snd_soc_platform *platform = card->platform;
F
Frank Mandarino 已提交
861 862
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

863
	run_delayed_work(&card->delayed_work);
864

F
Frank Mandarino 已提交
865 866 867 868 869 870
	if (platform->remove)
		platform->remove(pdev);

	if (codec_dev->remove)
		codec_dev->remove(pdev);

871 872
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
873
		if (cpu_dai->remove)
874
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
875 876
	}

877 878
	if (card->remove)
		card->remove(pdev);
F
Frank Mandarino 已提交
879

M
Mark Brown 已提交
880 881
	snd_soc_unregister_card(card);

F
Frank Mandarino 已提交
882 883 884 885 886 887 888
	return 0;
}

/* ASoC platform driver */
static struct platform_driver soc_driver = {
	.driver		= {
		.name		= "soc-audio",
889
		.owner		= THIS_MODULE,
F
Frank Mandarino 已提交
890 891 892 893 894 895 896 897 898 899 900 901
	},
	.probe		= soc_probe,
	.remove		= soc_remove,
	.suspend	= soc_suspend,
	.resume		= soc_resume,
};

/* create a new pcm */
static int soc_new_pcm(struct snd_soc_device *socdev,
	struct snd_soc_dai_link *dai_link, int num)
{
	struct snd_soc_codec *codec = socdev->codec;
902 903
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
904 905
	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
F
Frank Mandarino 已提交
906 907 908 909 910 911 912 913
	struct snd_soc_pcm_runtime *rtd;
	struct snd_pcm *pcm;
	char new_name[64];
	int ret = 0, playback = 0, capture = 0;

	rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL);
	if (rtd == NULL)
		return -ENOMEM;
914 915

	rtd->dai = dai_link;
F
Frank Mandarino 已提交
916
	rtd->socdev = socdev;
917
	codec_dai->codec = socdev->codec;
F
Frank Mandarino 已提交
918 919

	/* check client and interface hw capabilities */
M
Mark Brown 已提交
920 921
	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
		num);
F
Frank Mandarino 已提交
922 923 924 925 926 927 928 929 930

	if (codec_dai->playback.channels_min)
		playback = 1;
	if (codec_dai->capture.channels_min)
		capture = 1;

	ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback,
		capture, &pcm);
	if (ret < 0) {
931 932
		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
			codec->name);
F
Frank Mandarino 已提交
933 934 935 936
		kfree(rtd);
		return ret;
	}

937
	dai_link->pcm = pcm;
F
Frank Mandarino 已提交
938
	pcm->private_data = rtd;
939 940 941 942 943 944 945
	soc_pcm_ops.mmap = platform->pcm_ops->mmap;
	soc_pcm_ops.pointer = platform->pcm_ops->pointer;
	soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
	soc_pcm_ops.copy = platform->pcm_ops->copy;
	soc_pcm_ops.silence = platform->pcm_ops->silence;
	soc_pcm_ops.ack = platform->pcm_ops->ack;
	soc_pcm_ops.page = platform->pcm_ops->page;
F
Frank Mandarino 已提交
946 947 948 949 950 951 952

	if (playback)
		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops);

	if (capture)
		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops);

953
	ret = platform->pcm_new(codec->card, codec_dai, pcm);
F
Frank Mandarino 已提交
954 955 956 957 958 959
	if (ret < 0) {
		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
		kfree(rtd);
		return ret;
	}

960
	pcm->private_free = platform->pcm_free;
F
Frank Mandarino 已提交
961 962 963 964 965 966
	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
		cpu_dai->name);
	return ret;
}

/* codec register dump */
967
static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
F
Frank Mandarino 已提交
968 969 970 971 972 973 974 975 976 977 978
{
	struct snd_soc_codec *codec = devdata->codec;
	int i, step = 1, count = 0;

	if (!codec->reg_cache_size)
		return 0;

	if (codec->reg_cache_step)
		step = codec->reg_cache_step;

	count += sprintf(buf, "%s registers\n", codec->name);
979 980 981 982 983 984 985 986 987 988 989 990 991 992 993 994 995 996 997 998 999 1000 1001
	for (i = 0; i < codec->reg_cache_size; i += step) {
		count += sprintf(buf + count, "%2x: ", i);
		if (count >= PAGE_SIZE - 1)
			break;

		if (codec->display_register)
			count += codec->display_register(codec, buf + count,
							 PAGE_SIZE - count, i);
		else
			count += snprintf(buf + count, PAGE_SIZE - count,
					  "%4x", codec->read(codec, i));

		if (count >= PAGE_SIZE - 1)
			break;

		count += snprintf(buf + count, PAGE_SIZE - count, "\n");
		if (count >= PAGE_SIZE - 1)
			break;
	}

	/* Truncate count; min() would cause a warning */
	if (count >= PAGE_SIZE)
		count = PAGE_SIZE - 1;
F
Frank Mandarino 已提交
1002 1003 1004

	return count;
}
1005 1006 1007 1008 1009 1010 1011
static ssize_t codec_reg_show(struct device *dev,
	struct device_attribute *attr, char *buf)
{
	struct snd_soc_device *devdata = dev_get_drvdata(dev);
	return soc_codec_reg_show(devdata, buf);
}

F
Frank Mandarino 已提交
1012 1013
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);

1014 1015 1016 1017 1018 1019 1020 1021 1022 1023 1024
#ifdef CONFIG_DEBUG_FS
static int codec_reg_open_file(struct inode *inode, struct file *file)
{
	file->private_data = inode->i_private;
	return 0;
}

static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
			       size_t count, loff_t *ppos)
{
	ssize_t ret;
1025 1026 1027
	struct snd_soc_codec *codec = file->private_data;
	struct device *card_dev = codec->card->dev;
	struct snd_soc_device *devdata = card_dev->driver_data;
1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045
	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
	if (!buf)
		return -ENOMEM;
	ret = soc_codec_reg_show(devdata, buf);
	if (ret >= 0)
		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
	kfree(buf);
	return ret;
}

static ssize_t codec_reg_write_file(struct file *file,
		const char __user *user_buf, size_t count, loff_t *ppos)
{
	char buf[32];
	int buf_size;
	char *start = buf;
	unsigned long reg, value;
	int step = 1;
1046
	struct snd_soc_codec *codec = file->private_data;
1047 1048 1049 1050 1051 1052 1053 1054 1055 1056 1057 1058 1059 1060 1061 1062 1063 1064 1065 1066 1067 1068 1069 1070 1071 1072 1073 1074

	buf_size = min(count, (sizeof(buf)-1));
	if (copy_from_user(buf, user_buf, buf_size))
		return -EFAULT;
	buf[buf_size] = 0;

	if (codec->reg_cache_step)
		step = codec->reg_cache_step;

	while (*start == ' ')
		start++;
	reg = simple_strtoul(start, &start, 16);
	if ((reg >= codec->reg_cache_size) || (reg % step))
		return -EINVAL;
	while (*start == ' ')
		start++;
	if (strict_strtoul(start, 16, &value))
		return -EINVAL;
	codec->write(codec, reg, value);
	return buf_size;
}

static const struct file_operations codec_reg_fops = {
	.open = codec_reg_open_file,
	.read = codec_reg_read_file,
	.write = codec_reg_write_file,
};

1075
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1076
{
1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089
	codec->debugfs_reg = debugfs_create_file("codec_reg", 0644,
						 debugfs_root, codec,
						 &codec_reg_fops);
	if (!codec->debugfs_reg)
		printk(KERN_WARNING
		       "ASoC: Failed to create codec register debugfs file\n");

	codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744,
						     debugfs_root,
						     &codec->pop_time);
	if (!codec->debugfs_pop_time)
		printk(KERN_WARNING
		       "Failed to create pop time debugfs file\n");
1090 1091
}

1092
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1093
{
1094 1095
	debugfs_remove(codec->debugfs_pop_time);
	debugfs_remove(codec->debugfs_reg);
1096 1097 1098 1099
}

#else

1100
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1101 1102 1103
{
}

1104
static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1105 1106 1107 1108
{
}
#endif

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1109 1110 1111 1112 1113 1114 1115 1116 1117 1118 1119 1120 1121 1122 1123 1124 1125 1126 1127 1128 1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153 1154 1155 1156 1157 1158 1159 1160 1161 1162 1163 1164 1165 1166 1167 1168 1169 1170 1171 1172 1173 1174 1175 1176 1177 1178 1179 1180 1181 1182 1183 1184 1185 1186 1187 1188 1189 1190 1191 1192 1193 1194 1195 1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206 1207 1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222 1223
/**
 * snd_soc_new_ac97_codec - initailise AC97 device
 * @codec: audio codec
 * @ops: AC97 bus operations
 * @num: AC97 codec number
 *
 * Initialises AC97 codec resources for use by ad-hoc devices only.
 */
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
	struct snd_ac97_bus_ops *ops, int num)
{
	mutex_lock(&codec->mutex);

	codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL);
	if (codec->ac97 == NULL) {
		mutex_unlock(&codec->mutex);
		return -ENOMEM;
	}

	codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL);
	if (codec->ac97->bus == NULL) {
		kfree(codec->ac97);
		codec->ac97 = NULL;
		mutex_unlock(&codec->mutex);
		return -ENOMEM;
	}

	codec->ac97->bus->ops = ops;
	codec->ac97->num = num;
	mutex_unlock(&codec->mutex);
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec);

/**
 * snd_soc_free_ac97_codec - free AC97 codec device
 * @codec: audio codec
 *
 * Frees AC97 codec device resources.
 */
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec)
{
	mutex_lock(&codec->mutex);
	kfree(codec->ac97->bus);
	kfree(codec->ac97);
	codec->ac97 = NULL;
	mutex_unlock(&codec->mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec);

/**
 * snd_soc_update_bits - update codec register bits
 * @codec: audio codec
 * @reg: codec register
 * @mask: register mask
 * @value: new value
 *
 * Writes new register value.
 *
 * Returns 1 for change else 0.
 */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
				unsigned short mask, unsigned short value)
{
	int change;
	unsigned short old, new;

	mutex_lock(&io_mutex);
	old = snd_soc_read(codec, reg);
	new = (old & ~mask) | value;
	change = old != new;
	if (change)
		snd_soc_write(codec, reg, new);

	mutex_unlock(&io_mutex);
	return change;
}
EXPORT_SYMBOL_GPL(snd_soc_update_bits);

/**
 * snd_soc_test_bits - test register for change
 * @codec: audio codec
 * @reg: codec register
 * @mask: register mask
 * @value: new value
 *
 * Tests a register with a new value and checks if the new value is
 * different from the old value.
 *
 * Returns 1 for change else 0.
 */
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
				unsigned short mask, unsigned short value)
{
	int change;
	unsigned short old, new;

	mutex_lock(&io_mutex);
	old = snd_soc_read(codec, reg);
	new = (old & ~mask) | value;
	change = old != new;
	mutex_unlock(&io_mutex);

	return change;
}
EXPORT_SYMBOL_GPL(snd_soc_test_bits);

/**
 * snd_soc_new_pcms - create new sound card and pcms
 * @socdev: the SoC audio device
 *
 * Create a new sound card based upon the codec and interface pcms.
 *
 * Returns 0 for success, else error.
 */
1224
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
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1225 1226
{
	struct snd_soc_codec *codec = socdev->codec;
1227
	struct snd_soc_card *card = socdev->card;
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1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245
	int ret = 0, i;

	mutex_lock(&codec->mutex);

	/* register a sound card */
	codec->card = snd_card_new(idx, xid, codec->owner, 0);
	if (!codec->card) {
		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
			codec->name);
		mutex_unlock(&codec->mutex);
		return -ENODEV;
	}

	codec->card->dev = socdev->dev;
	codec->card->private_data = codec;
	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));

	/* create the pcms */
1246 1247
	for (i = 0; i < card->num_links; i++) {
		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: can't create pcm %s\n",
1250
				card->dai_link[i].stream_name);
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1251 1252 1253 1254 1255 1256 1257 1258 1259 1260 1261
			mutex_unlock(&codec->mutex);
			return ret;
		}
	}

	mutex_unlock(&codec->mutex);
	return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_new_pcms);

/**
1262
 * snd_soc_init_card - register sound card
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 * @socdev: the SoC audio device
 *
 * Register a SoC sound card. Also registers an AC97 device if the
 * codec is AC97 for ad hoc devices.
 *
 * Returns 0 for success, else error.
 */
1270
int snd_soc_init_card(struct snd_soc_device *socdev)
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1271 1272
{
	struct snd_soc_codec *codec = socdev->codec;
1273
	struct snd_soc_card *card = socdev->card;
1274
	int ret = 0, i, ac97 = 0, err = 0;
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1276 1277 1278
	for (i = 0; i < card->num_links; i++) {
		if (card->dai_link[i].init) {
			err = card->dai_link[i].init(codec);
1279 1280
			if (err < 0) {
				printk(KERN_ERR "asoc: failed to init %s\n",
1281
					card->dai_link[i].stream_name);
1282 1283 1284
				continue;
			}
		}
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		if (card->dai_link[i].codec_dai->ac97_control)
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1286 1287 1288
			ac97 = 1;
	}
	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
1289
		 "%s",  card->name);
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	snprintf(codec->card->longname, sizeof(codec->card->longname),
1291
		 "%s (%s)", card->name, codec->name);
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1292 1293 1294

	ret = snd_card_register(codec->card);
	if (ret < 0) {
1295
		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
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1296
				codec->name);
1297
		goto out;
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1298 1299
	}

1300
	mutex_lock(&codec->mutex);
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1301
#ifdef CONFIG_SND_SOC_AC97_BUS
1302 1303 1304 1305 1306
	if (ac97) {
		ret = soc_ac97_dev_register(codec);
		if (ret < 0) {
			printk(KERN_ERR "asoc: AC97 device register failed\n");
			snd_card_free(codec->card);
1307
			mutex_unlock(&codec->mutex);
1308 1309 1310
			goto out;
		}
	}
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#endif

1313 1314 1315 1316 1317 1318
	err = snd_soc_dapm_sys_add(socdev->dev);
	if (err < 0)
		printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n");

	err = device_create_file(socdev->dev, &dev_attr_codec_reg);
	if (err < 0)
1319
		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
1320

1321
	soc_init_codec_debugfs(socdev->codec);
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	mutex_unlock(&codec->mutex);
1323 1324

out:
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1325 1326
	return ret;
}
1327
EXPORT_SYMBOL_GPL(snd_soc_init_card);
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/**
 * snd_soc_free_pcms - free sound card and pcms
 * @socdev: the SoC audio device
 *
 * Frees sound card and pcms associated with the socdev.
 * Also unregister the codec if it is an AC97 device.
 */
void snd_soc_free_pcms(struct snd_soc_device *socdev)
{
	struct snd_soc_codec *codec = socdev->codec;
1339
#ifdef CONFIG_SND_SOC_AC97_BUS
1340
	struct snd_soc_dai *codec_dai;
1341 1342
	int i;
#endif
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	mutex_lock(&codec->mutex);
1345
	soc_cleanup_codec_debugfs(socdev->codec);
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#ifdef CONFIG_SND_SOC_AC97_BUS
1347
	for (i = 0; i < codec->num_dai; i++) {
1348
		codec_dai = &codec->dai[i];
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		if (codec_dai->ac97_control && codec->ac97) {
1350 1351 1352 1353 1354
			soc_ac97_dev_unregister(codec);
			goto free_card;
		}
	}
free_card:
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#endif

	if (codec->card)
		snd_card_free(codec->card);
	device_remove_file(socdev->dev, &dev_attr_codec_reg);
	mutex_unlock(&codec->mutex);
}
EXPORT_SYMBOL_GPL(snd_soc_free_pcms);

/**
 * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
 * @substream: the pcm substream
 * @hw: the hardware parameters
 *
 * Sets the substream runtime hardware parameters.
 */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
	const struct snd_pcm_hardware *hw)
{
	struct snd_pcm_runtime *runtime = substream->runtime;
	runtime->hw.info = hw->info;
	runtime->hw.formats = hw->formats;
	runtime->hw.period_bytes_min = hw->period_bytes_min;
	runtime->hw.period_bytes_max = hw->period_bytes_max;
	runtime->hw.periods_min = hw->periods_min;
	runtime->hw.periods_max = hw->periods_max;
	runtime->hw.buffer_bytes_max = hw->buffer_bytes_max;
	runtime->hw.fifo_size = hw->fifo_size;
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams);

/**
 * snd_soc_cnew - create new control
 * @_template: control template
 * @data: control private data
 * @lnng_name: control long name
 *
 * Create a new mixer control from a template control.
 *
 * Returns 0 for success, else error.
 */
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
	void *data, char *long_name)
{
	struct snd_kcontrol_new template;

	memcpy(&template, _template, sizeof(template));
	if (long_name)
		template.name = long_name;
	template.index = 0;

	return snd_ctl_new1(&template, data);
}
EXPORT_SYMBOL_GPL(snd_soc_cnew);

/**
 * snd_soc_info_enum_double - enumerated double mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a double enumerated
 * mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;

	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
	uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
1428
	uinfo->value.enumerated.items = e->max;
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1430 1431
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
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1432 1433 1434 1435 1436 1437 1438 1439 1440 1441 1442 1443 1444 1445 1446 1447 1448 1449 1450 1451 1452 1453
	strcpy(uinfo->value.enumerated.name,
		e->texts[uinfo->value.enumerated.item]);
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);

/**
 * snd_soc_get_enum_double - enumerated double mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a double enumerated mixer.
 *
 * Returns 0 for success.
 */
int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
	unsigned short val, bitmask;

1454
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
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1455 1456
		;
	val = snd_soc_read(codec, e->reg);
1457 1458
	ucontrol->value.enumerated.item[0]
		= (val >> e->shift_l) & (bitmask - 1);
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1459 1460 1461 1462 1463 1464 1465 1466 1467 1468 1469 1470 1471 1472 1473 1474 1475 1476 1477 1478 1479 1480 1481 1482 1483
	if (e->shift_l != e->shift_r)
		ucontrol->value.enumerated.item[1] =
			(val >> e->shift_r) & (bitmask - 1);

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);

/**
 * snd_soc_put_enum_double - enumerated double mixer put callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a double enumerated mixer.
 *
 * Returns 0 for success.
 */
int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
	unsigned short val;
	unsigned short mask, bitmask;

1484
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1485
		;
1486
	if (ucontrol->value.enumerated.item[0] > e->max - 1)
F
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1487 1488 1489 1490
		return -EINVAL;
	val = ucontrol->value.enumerated.item[0] << e->shift_l;
	mask = (bitmask - 1) << e->shift_l;
	if (e->shift_l != e->shift_r) {
1491
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
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1492 1493 1494 1495 1496 1497 1498 1499 1500 1501 1502 1503 1504 1505 1506 1507 1508 1509 1510 1511 1512 1513 1514 1515 1516 1517
			return -EINVAL;
		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
		mask |= (bitmask - 1) << e->shift_r;
	}

	return snd_soc_update_bits(codec, e->reg, mask, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);

/**
 * snd_soc_info_enum_ext - external enumerated single mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about an external enumerated
 * single mixer.
 *
 * Returns 0 for success.
 */
int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;

	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
	uinfo->count = 1;
1518
	uinfo->value.enumerated.items = e->max;
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1520 1521
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
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1522 1523 1524 1525 1526 1527 1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539
	strcpy(uinfo->value.enumerated.name,
		e->texts[uinfo->value.enumerated.item]);
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);

/**
 * snd_soc_info_volsw_ext - external single mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a single external mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
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1540 1541 1542 1543 1544 1545
	int max = kcontrol->private_value;

	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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1546 1547 1548

	uinfo->count = 1;
	uinfo->value.integer.min = 0;
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1549
	uinfo->value.integer.max = max;
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1550 1551 1552 1553 1554 1555 1556 1557 1558 1559 1560 1561 1562 1563 1564 1565
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);

/**
 * snd_soc_info_volsw - single mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
1566 1567 1568
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
1569
	unsigned int shift = mc->shift;
1570
	unsigned int rshift = mc->rshift;
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1571

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1572 1573 1574 1575 1576
	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;

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	uinfo->count = shift == rshift ? 1 : 2;
	uinfo->value.integer.min = 0;
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1579
	uinfo->value.integer.max = max;
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1580 1581 1582 1583 1584 1585 1586 1587 1588 1589 1590 1591 1592 1593 1594 1595
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);

/**
 * snd_soc_get_volsw - single mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1596 1597
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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1598
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1599 1600 1601
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1602
	int max = mc->max;
1603 1604
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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1605 1606 1607 1608 1609 1610 1611 1612

	ucontrol->value.integer.value[0] =
		(snd_soc_read(codec, reg) >> shift) & mask;
	if (shift != rshift)
		ucontrol->value.integer.value[1] =
			(snd_soc_read(codec, reg) >> rshift) & mask;
	if (invert) {
		ucontrol->value.integer.value[0] =
P
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1613
			max - ucontrol->value.integer.value[0];
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1614 1615
		if (shift != rshift)
			ucontrol->value.integer.value[1] =
P
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1616
				max - ucontrol->value.integer.value[1];
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1617 1618 1619 1620 1621 1622 1623 1624 1625 1626 1627 1628 1629 1630 1631 1632 1633 1634
	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw);

/**
 * snd_soc_put_volsw - single mixer put callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1635 1636
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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1637
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1638 1639 1640
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1641
	int max = mc->max;
1642 1643
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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	unsigned short val, val2, val_mask;

	val = (ucontrol->value.integer.value[0] & mask);
	if (invert)
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		val = max - val;
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	val_mask = mask << shift;
	val = val << shift;
	if (shift != rshift) {
		val2 = (ucontrol->value.integer.value[1] & mask);
		if (invert)
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			val2 = max - val2;
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		val_mask |= mask << rshift;
		val |= val2 << rshift;
	}
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	return snd_soc_update_bits(codec, reg, val_mask, val);
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}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);

/**
 * snd_soc_info_volsw_2r - double mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a double mixer control that
 * spans 2 codec registers.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
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	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
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	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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	uinfo->count = 2;
	uinfo->value.integer.min = 0;
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	uinfo->value.integer.max = max;
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	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);

/**
 * snd_soc_get_volsw_2r - double mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a double mixer control that spans 2 registers.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
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	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1709
	int max = mc->max;
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	unsigned int mask = (1<<fls(max))-1;
	unsigned int invert = mc->invert;
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	ucontrol->value.integer.value[0] =
		(snd_soc_read(codec, reg) >> shift) & mask;
	ucontrol->value.integer.value[1] =
		(snd_soc_read(codec, reg2) >> shift) & mask;
	if (invert) {
		ucontrol->value.integer.value[0] =
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			max - ucontrol->value.integer.value[0];
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		ucontrol->value.integer.value[1] =
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			max - ucontrol->value.integer.value[1];
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	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);

/**
 * snd_soc_put_volsw_2r - double mixer set callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a double mixer control that spans 2 registers.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
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	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1746
	int max = mc->max;
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	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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	int err;
	unsigned short val, val2, val_mask;

	val_mask = mask << shift;
	val = (ucontrol->value.integer.value[0] & mask);
	val2 = (ucontrol->value.integer.value[1] & mask);

	if (invert) {
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		val = max - val;
		val2 = max - val2;
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	}

	val = val << shift;
	val2 = val2 << shift;

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	err = snd_soc_update_bits(codec, reg, val_mask, val);
	if (err < 0)
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		return err;

	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
	return err;
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r);

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/**
 * snd_soc_info_volsw_s8 - signed mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a signed mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
1785 1786 1787 1788
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
	int min = mc->min;
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	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
	uinfo->count = 2;
	uinfo->value.integer.min = 0;
	uinfo->value.integer.max = max-min;
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);

/**
 * snd_soc_get_volsw_s8 - signed mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a signed mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1810 1811
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
1812
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1813
	unsigned int reg = mc->reg;
1814
	int min = mc->min;
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	int val = snd_soc_read(codec, reg);

	ucontrol->value.integer.value[0] =
		((signed char)(val & 0xff))-min;
	ucontrol->value.integer.value[1] =
		((signed char)((val >> 8) & 0xff))-min;
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8);

/**
 * snd_soc_put_volsw_sgn - signed mixer put callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a signed mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1837 1838
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
1839
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1840
	unsigned int reg = mc->reg;
1841
	int min = mc->min;
1842 1843 1844 1845 1846 1847 1848 1849 1850
	unsigned short val;

	val = (ucontrol->value.integer.value[0]+min) & 0xff;
	val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8;

	return snd_soc_update_bits(codec, reg, 0xffff, val);
}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);

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/**
 * snd_soc_dai_set_sysclk - configure DAI system or master clock.
 * @dai: DAI
 * @clk_id: DAI specific clock ID
 * @freq: new clock frequency in Hz
 * @dir: new clock direction - input/output.
 *
 * Configures the DAI master (MCLK) or system (SYSCLK) clocking.
 */
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
	unsigned int freq, int dir)
{
1863 1864
	if (dai->ops.set_sysclk)
		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
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	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);

/**
 * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
 * @dai: DAI
 * @clk_id: DAI specific clock divider ID
 * @div: new clock divisor.
 *
 * Configures the clock dividers. This is used to derive the best DAI bit and
 * frame clocks from the system or master clock. It's best to set the DAI bit
 * and frame clocks as low as possible to save system power.
 */
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	int div_id, int div)
{
1883 1884
	if (dai->ops.set_clkdiv)
		return dai->ops.set_clkdiv(dai, div_id, div);
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	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);

/**
 * snd_soc_dai_set_pll - configure DAI PLL.
 * @dai: DAI
 * @pll_id: DAI specific PLL ID
 * @freq_in: PLL input clock frequency in Hz
 * @freq_out: requested PLL output clock frequency in Hz
 *
 * Configures and enables PLL to generate output clock based on input clock.
 */
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
	int pll_id, unsigned int freq_in, unsigned int freq_out)
{
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	if (dai->ops.set_pll)
		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
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	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);

/**
 * snd_soc_dai_set_fmt - configure DAI hardware audio format.
 * @dai: DAI
 * @fmt: SND_SOC_DAIFMT_ format value.
 *
 * Configures the DAI hardware format and clocking.
 */
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
1918 1919
	if (dai->ops.set_fmt)
		return dai->ops.set_fmt(dai, fmt);
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	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);

/**
 * snd_soc_dai_set_tdm_slot - configure DAI TDM.
 * @dai: DAI
 * @mask: DAI specific mask representing used slots.
 * @slots: Number of slots in use.
 *
 * Configures a DAI for TDM operation. Both mask and slots are codec and DAI
 * specific.
 */
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
	unsigned int mask, int slots)
{
1937 1938
	if (dai->ops.set_sysclk)
		return dai->ops.set_tdm_slot(dai, mask, slots);
1939 1940 1941 1942 1943 1944 1945 1946 1947 1948 1949 1950 1951 1952
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);

/**
 * snd_soc_dai_set_tristate - configure DAI system or master clock.
 * @dai: DAI
 * @tristate: tristate enable
 *
 * Tristates the DAI so that others can use it.
 */
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
{
1953 1954
	if (dai->ops.set_sysclk)
		return dai->ops.set_tristate(dai, tristate);
1955 1956 1957 1958 1959 1960 1961 1962 1963 1964 1965 1966 1967 1968
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);

/**
 * snd_soc_dai_digital_mute - configure DAI system or master clock.
 * @dai: DAI
 * @mute: mute enable
 *
 * Mutes the DAI DAC.
 */
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
{
1969 1970
	if (dai->ops.digital_mute)
		return dai->ops.digital_mute(dai, mute);
1971 1972 1973 1974 1975
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);

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/**
 * snd_soc_register_card - Register a card with the ASoC core
 *
 * @param card Card to register
 *
 * Note that currently this is an internal only function: it will be
 * exposed to machine drivers after further backporting of ASoC v2
 * registration APIs.
 */
static int snd_soc_register_card(struct snd_soc_card *card)
{
	if (!card->name || !card->dev)
		return -EINVAL;

	INIT_LIST_HEAD(&card->list);
	card->instantiated = 0;

	mutex_lock(&client_mutex);
	list_add(&card->list, &card_list);
	mutex_unlock(&client_mutex);

	dev_dbg(card->dev, "Registered card '%s'\n", card->name);

	return 0;
}

/**
 * snd_soc_unregister_card - Unregister a card with the ASoC core
 *
 * @param card Card to unregister
 *
 * Note that currently this is an internal only function: it will be
 * exposed to machine drivers after further backporting of ASoC v2
 * registration APIs.
 */
static int snd_soc_unregister_card(struct snd_soc_card *card)
{
	mutex_lock(&client_mutex);
	list_del(&card->list);
	mutex_unlock(&client_mutex);

	dev_dbg(card->dev, "Unregistered card '%s'\n", card->name);

	return 0;
}

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static int __devinit snd_soc_init(void)
{
2024 2025 2026 2027 2028 2029 2030 2031 2032
#ifdef CONFIG_DEBUG_FS
	debugfs_root = debugfs_create_dir("asoc", NULL);
	if (IS_ERR(debugfs_root) || !debugfs_root) {
		printk(KERN_WARNING
		       "ASoC: Failed to create debugfs directory\n");
		debugfs_root = NULL;
	}
#endif

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	return platform_driver_register(&soc_driver);
}

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static void __exit snd_soc_exit(void)
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{
2038 2039 2040
#ifdef CONFIG_DEBUG_FS
	debugfs_remove_recursive(debugfs_root);
#endif
2041
	platform_driver_unregister(&soc_driver);
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}

module_init(snd_soc_init);
module_exit(snd_soc_exit);

/* Module information */
2048
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
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MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
2051
MODULE_ALIAS("platform:soc-audio");