soc-core.c 64.5 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);
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static LIST_HEAD(dai_list);
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static LIST_HEAD(platform_list);
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static LIST_HEAD(codec_list);
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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;
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	codec->ac97->dev.parent = codec->card->dev;
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	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

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static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_dai_link *machine = rtd->dai;
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
	int ret;

	if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates ||
	    machine->symmetric_rates) {
		dev_dbg(card->dev, "Symmetry forces %dHz rate\n", 
			machine->rate);

		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
						   SNDRV_PCM_HW_PARAM_RATE,
						   machine->rate,
						   machine->rate);
		if (ret < 0) {
			dev_err(card->dev,
				"Unable to apply rate symmetry constraint: %d\n", ret);
			return ret;
		}
	}

	return 0;
}

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/*
 * 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) {
		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) {
		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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	/* Symmetry only applies if we've already got an active stream. */
	if (cpu_dai->active || codec_dai->active) {
		ret = soc_pcm_apply_symmetry(substream);
		if (ret != 0)
			goto machine_err;
	}

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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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	card->codec->active++;
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	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)
		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);
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	struct snd_soc_codec *codec = card->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) {
			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);
		}
	}
	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 = card->codec;
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	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)
		cpu_dai->ops->shutdown(substream, cpu_dai);
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	if (codec_dai->ops->shutdown)
		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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	}

	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 = card->codec;
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	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) {
		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) {
		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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	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		snd_soc_dapm_stream_event(codec,
					  codec_dai->playback.stream_name,
					  SND_SOC_DAPM_STREAM_START);
	else
		snd_soc_dapm_stream_event(codec,
					  codec_dai->capture.stream_name,
					  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) {
		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) {
		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;
		}
	}

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	machine->rate = params_rate(params);

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out:
	mutex_unlock(&pcm_mutex);
	return ret;

platform_err:
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	if (cpu_dai->ops->hw_free)
		cpu_dai->ops->hw_free(substream, cpu_dai);
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interface_err:
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	if (codec_dai->ops->hw_free)
		codec_dai->ops->hw_free(substream, codec_dai);
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codec_err:
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	if (machine->ops && machine->ops->hw_free)
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		machine->ops->hw_free(substream);
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	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;
547
	struct snd_soc_dai_link *machine = rtd->dai;
548 549
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
550 551
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
552
	struct snd_soc_codec *codec = card->codec;
F
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553 554 555 556

	mutex_lock(&pcm_mutex);

	/* apply codec digital mute */
557 558
	if (!codec->active)
		snd_soc_dai_digital_mute(codec_dai, 1);
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559 560 561 562 563 564 565 566 567 568

	/* 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  */
569 570
	if (codec_dai->ops->hw_free)
		codec_dai->ops->hw_free(substream, codec_dai);
F
Frank Mandarino 已提交
571

572 573
	if (cpu_dai->ops->hw_free)
		cpu_dai->ops->hw_free(substream, cpu_dai);
F
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574 575 576 577 578 579 580 581 582

	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;
583
	struct snd_soc_card *card= socdev->card;
584
	struct snd_soc_dai_link *machine = rtd->dai;
585
	struct snd_soc_platform *platform = card->platform;
586 587
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
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588 589
	int ret;

590 591
	if (codec_dai->ops->trigger) {
		ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
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592 593 594 595 596 597 598 599 600 601
		if (ret < 0)
			return ret;
	}

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

602 603
	if (cpu_dai->ops->trigger) {
		ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
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604 605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623
		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)
{
624
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
625
	struct snd_soc_card *card = socdev->card;
626
	struct snd_soc_platform *platform = card->platform;
627
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
628
	struct snd_soc_codec *codec = card->codec;
F
Frank Mandarino 已提交
629 630
	int i;

631 632 633 634 635 636
	/* If the initialization of this soc device failed, there is no codec
	 * associated with it. Just bail out in this case.
	 */
	if (!codec)
		return 0;

637 638 639 640 641 642 643 644 645 646
	/* 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
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647
	/* mute any active DAC's */
648 649
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
650 651
		if (dai->ops->digital_mute && dai->playback.active)
			dai->ops->digital_mute(dai, 1);
F
Frank Mandarino 已提交
652 653
	}

654
	/* suspend all pcms */
655 656
	for (i = 0; i < card->num_links; i++)
		snd_pcm_suspend_all(card->dai_link[i].pcm);
657

658 659
	if (card->suspend_pre)
		card->suspend_pre(pdev, state);
F
Frank Mandarino 已提交
660

661 662
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai  *cpu_dai = card->dai_link[i].cpu_dai;
M
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663
		if (cpu_dai->suspend && !cpu_dai->ac97_control)
664
			cpu_dai->suspend(cpu_dai);
F
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665
		if (platform->suspend)
666
			platform->suspend(cpu_dai);
F
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667 668 669
	}

	/* close any waiting streams and save state */
670
	run_delayed_work(&card->delayed_work);
671
	codec->suspend_bias_level = codec->bias_level;
F
Frank Mandarino 已提交
672

673
	for (i = 0; i < codec->num_dai; i++) {
F
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674 675 676 677 678 679 680 681 682 683 684 685 686
		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);

687 688
	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
689
		if (cpu_dai->suspend && cpu_dai->ac97_control)
690
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
691 692
	}

693 694
	if (card->suspend_post)
		card->suspend_post(pdev, state);
F
Frank Mandarino 已提交
695 696 697 698

	return 0;
}

699 700 701 702
/* 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
703
{
704 705 706 707
	struct snd_soc_card *card = container_of(work,
						 struct snd_soc_card,
						 deferred_resume_work);
	struct snd_soc_device *socdev = card->socdev;
708
	struct snd_soc_platform *platform = card->platform;
709
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
710
	struct snd_soc_codec *codec = card->codec;
711
	struct platform_device *pdev = to_platform_device(socdev->dev);
F
Frank Mandarino 已提交
712 713
	int i;

714 715 716 717
	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
	 * so userspace apps are blocked from touching us
	 */

718
	dev_dbg(socdev->dev, "starting resume work\n");
719

720 721
	if (card->resume_pre)
		card->resume_pre(pdev);
F
Frank Mandarino 已提交
722

723 724
	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
725
		if (cpu_dai->resume && cpu_dai->ac97_control)
726
			cpu_dai->resume(cpu_dai);
F
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727 728 729 730 731
	}

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

732 733
	for (i = 0; i < codec->num_dai; i++) {
		char *stream = codec->dai[i].playback.stream_name;
F
Frank Mandarino 已提交
734 735 736 737 738 739 740 741 742
		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);
	}

743
	/* unmute any active DACs */
744 745
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
746 747
		if (dai->ops->digital_mute && dai->playback.active)
			dai->ops->digital_mute(dai, 0);
F
Frank Mandarino 已提交
748 749
	}

750 751
	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
752
		if (cpu_dai->resume && !cpu_dai->ac97_control)
753
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
754
		if (platform->resume)
755
			platform->resume(cpu_dai);
F
Frank Mandarino 已提交
756 757
	}

758 759
	if (card->resume_post)
		card->resume_post(pdev);
F
Frank Mandarino 已提交
760

761
	dev_dbg(socdev->dev, "resume work completed\n");
762 763 764 765 766 767 768 769 770

	/* 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);
771
	struct snd_soc_card *card = socdev->card;
772
	struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
773

774 775 776 777 778 779 780 781 782 783 784 785 786
	/* AC97 devices might have other drivers hanging off them so
	 * need to resume immediately.  Other drivers don't have that
	 * problem and may take a substantial amount of time to resume
	 * due to I/O costs and anti-pop so handle them out of line.
	 */
	if (cpu_dai->ac97_control) {
		dev_dbg(socdev->dev, "Resuming AC97 immediately\n");
		soc_resume_deferred(&card->deferred_resume_work);
	} else {
		dev_dbg(socdev->dev, "Scheduling resume work\n");
		if (!schedule_work(&card->deferred_resume_work))
			dev_err(socdev->dev, "resume work item may be lost\n");
	}
787

F
Frank Mandarino 已提交
788 789 790 791 792 793 794 795
	return 0;
}

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

796
static void snd_soc_instantiate_card(struct snd_soc_card *card)
F
Frank Mandarino 已提交
797
{
798 799 800 801 802 803
	struct platform_device *pdev = container_of(card->dev,
						    struct platform_device,
						    dev);
	struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev;
	struct snd_soc_platform *platform;
	struct snd_soc_dai *dai;
804
	int i, found, ret, ac97;
805 806 807 808 809 810 811 812 813 814 815 816 817 818 819

	if (card->instantiated)
		return;

	found = 0;
	list_for_each_entry(platform, &platform_list, list)
		if (card->platform == platform) {
			found = 1;
			break;
		}
	if (!found) {
		dev_dbg(card->dev, "Platform %s not registered\n",
			card->platform->name);
		return;
	}
820

821
	ac97 = 0;
822 823 824 825 826 827 828 829 830 831 832 833
	for (i = 0; i < card->num_links; i++) {
		found = 0;
		list_for_each_entry(dai, &dai_list, list)
			if (card->dai_link[i].cpu_dai == dai) {
				found = 1;
				break;
			}
		if (!found) {
			dev_dbg(card->dev, "DAI %s not registered\n",
				card->dai_link[i].cpu_dai->name);
			return;
		}
834 835 836

		if (card->dai_link[i].cpu_dai->ac97_control)
			ac97 = 1;
M
Mark Brown 已提交
837 838
	}

839 840 841 842 843 844 845 846 847 848 849 850 851 852 853 854 855 856 857 858 859
	/* If we have AC97 in the system then don't wait for the
	 * codec.  This will need revisiting if we have to handle
	 * systems with mixed AC97 and non-AC97 parts.  Only check for
	 * DAIs currently; we can't do this per link since some AC97
	 * codecs have non-AC97 DAIs.
	 */
	if (!ac97)
		for (i = 0; i < card->num_links; i++) {
			found = 0;
			list_for_each_entry(dai, &dai_list, list)
				if (card->dai_link[i].codec_dai == dai) {
					found = 1;
					break;
				}
			if (!found) {
				dev_dbg(card->dev, "DAI %s not registered\n",
					card->dai_link[i].codec_dai->name);
				return;
			}
		}

860 861 862 863 864
	/* Note that we do not current check for codec components */

	dev_dbg(card->dev, "All components present, instantiating\n");

	/* Found everything, bring it up */
865 866
	if (card->probe) {
		ret = card->probe(pdev);
867
		if (ret < 0)
868
			return;
F
Frank Mandarino 已提交
869 870
	}

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->probe) {
874
			ret = cpu_dai->probe(pdev, cpu_dai);
875
			if (ret < 0)
F
Frank Mandarino 已提交
876 877 878 879 880 881
				goto cpu_dai_err;
		}
	}

	if (codec_dev->probe) {
		ret = codec_dev->probe(pdev);
882
		if (ret < 0)
F
Frank Mandarino 已提交
883 884 885 886 887
			goto cpu_dai_err;
	}

	if (platform->probe) {
		ret = platform->probe(pdev);
888
		if (ret < 0)
F
Frank Mandarino 已提交
889 890 891 892
			goto platform_err;
	}

	/* DAPM stream work */
893
	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
R
Randy Dunlap 已提交
894
#ifdef CONFIG_PM
895
	/* deferred resume work */
896
	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
R
Randy Dunlap 已提交
897
#endif
898

899 900 901
	card->instantiated = 1;

	return;
F
Frank Mandarino 已提交
902 903 904 905 906 907

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

cpu_dai_err:
908
	for (i--; i >= 0; i--) {
909
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
910
		if (cpu_dai->remove)
911
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
912 913
	}

914 915
	if (card->remove)
		card->remove(pdev);
916
}
F
Frank Mandarino 已提交
917

918 919 920 921 922 923 924 925 926 927 928 929 930 931 932 933 934 935 936 937 938 939 940 941 942 943 944 945 946 947
/*
 * Attempt to initialise any uninitalised cards.  Must be called with
 * client_mutex.
 */
static void snd_soc_instantiate_cards(void)
{
	struct snd_soc_card *card;
	list_for_each_entry(card, &card_list, list)
		snd_soc_instantiate_card(card);
}

/* probes a new socdev */
static int soc_probe(struct platform_device *pdev)
{
	int ret = 0;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_card *card = socdev->card;

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

	/* 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;
	}

	return 0;
F
Frank Mandarino 已提交
948 949 950 951 952 953 954
}

/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
	int i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
955
	struct snd_soc_card *card = socdev->card;
956
	struct snd_soc_platform *platform = card->platform;
F
Frank Mandarino 已提交
957 958
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

959 960 961
	if (!card->instantiated)
		return 0;

962
	run_delayed_work(&card->delayed_work);
963

F
Frank Mandarino 已提交
964 965 966 967 968 969
	if (platform->remove)
		platform->remove(pdev);

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

970 971
	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
972
		if (cpu_dai->remove)
973
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
974 975
	}

976 977
	if (card->remove)
		card->remove(pdev);
F
Frank Mandarino 已提交
978

M
Mark Brown 已提交
979 980
	snd_soc_unregister_card(card);

F
Frank Mandarino 已提交
981 982 983 984 985 986 987
	return 0;
}

/* ASoC platform driver */
static struct platform_driver soc_driver = {
	.driver		= {
		.name		= "soc-audio",
988
		.owner		= THIS_MODULE,
F
Frank Mandarino 已提交
989 990 991 992 993 994 995 996 997 998 999
	},
	.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)
{
1000
	struct snd_soc_card *card = socdev->card;
1001
	struct snd_soc_codec *codec = card->codec;
1002
	struct snd_soc_platform *platform = card->platform;
1003 1004
	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
1005 1006 1007 1008 1009 1010 1011 1012
	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;
1013 1014

	rtd->dai = dai_link;
F
Frank Mandarino 已提交
1015
	rtd->socdev = socdev;
1016
	codec_dai->codec = card->codec;
F
Frank Mandarino 已提交
1017 1018

	/* check client and interface hw capabilities */
M
Mark Brown 已提交
1019 1020
	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
		num);
F
Frank Mandarino 已提交
1021 1022 1023 1024 1025 1026 1027 1028 1029

	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) {
1030 1031
		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
			codec->name);
F
Frank Mandarino 已提交
1032 1033 1034 1035
		kfree(rtd);
		return ret;
	}

1036
	dai_link->pcm = pcm;
F
Frank Mandarino 已提交
1037
	pcm->private_data = rtd;
1038 1039 1040 1041 1042 1043 1044
	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
1045 1046 1047 1048 1049 1050 1051

	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);

1052
	ret = platform->pcm_new(codec->card, codec_dai, pcm);
F
Frank Mandarino 已提交
1053 1054 1055 1056 1057 1058
	if (ret < 0) {
		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
		kfree(rtd);
		return ret;
	}

1059
	pcm->private_free = platform->pcm_free;
F
Frank Mandarino 已提交
1060 1061 1062 1063 1064 1065
	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
		cpu_dai->name);
	return ret;
}

/* codec register dump */
1066
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
F
Frank Mandarino 已提交
1067 1068 1069 1070 1071 1072 1073 1074 1075 1076
{
	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);
1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089 1090 1091 1092 1093 1094 1095 1096 1097 1098 1099
	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;
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	return count;
}
1103 1104 1105 1106
static ssize_t codec_reg_show(struct device *dev,
	struct device_attribute *attr, char *buf)
{
	struct snd_soc_device *devdata = dev_get_drvdata(dev);
1107
	return soc_codec_reg_show(devdata->card->codec, buf);
1108 1109
}

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1110 1111
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);

1112 1113 1114 1115 1116 1117 1118 1119 1120 1121 1122
#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;
1123
	struct snd_soc_codec *codec = file->private_data;
1124 1125 1126
	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
	if (!buf)
		return -ENOMEM;
1127
	ret = soc_codec_reg_show(codec, buf);
1128 1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141
	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;
1142
	struct snd_soc_codec *codec = file->private_data;
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

	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,
};

1171
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1172
{
1173 1174 1175 1176 1177 1178 1179 1180 1181 1182 1183 1184 1185
	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");
1186 1187
}

1188
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1189
{
1190 1191
	debugfs_remove(codec->debugfs_pop_time);
	debugfs_remove(codec->debugfs_reg);
1192 1193 1194 1195
}

#else

1196
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1197 1198 1199
{
}

1200
static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1201 1202 1203 1204
{
}
#endif

F
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1205 1206 1207 1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245 1246 1247 1248 1249 1250 1251 1252 1253 1254 1255 1256 1257 1258 1259 1260 1261 1262 1263 1264 1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275 1276 1277 1278 1279 1280 1281 1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298 1299 1300 1301 1302 1303 1304 1305 1306 1307 1308 1309 1310 1311 1312 1313 1314
/**
 * 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
M
Mark Brown 已提交
1315 1316
 * @idx: ALSA card index
 * @xid: card identification
F
Frank Mandarino 已提交
1317 1318 1319 1320 1321
 *
 * Create a new sound card based upon the codec and interface pcms.
 *
 * Returns 0 for success, else error.
 */
1322
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
F
Frank Mandarino 已提交
1323
{
1324
	struct snd_soc_card *card = socdev->card;
1325
	struct snd_soc_codec *codec = card->codec;
1326
	int ret, i;
F
Frank Mandarino 已提交
1327 1328 1329 1330

	mutex_lock(&codec->mutex);

	/* register a sound card */
1331 1332
	ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
	if (ret < 0) {
F
Frank Mandarino 已提交
1333 1334 1335
		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
			codec->name);
		mutex_unlock(&codec->mutex);
1336
		return ret;
F
Frank Mandarino 已提交
1337 1338
	}

1339
	codec->socdev = socdev;
F
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1340 1341 1342 1343 1344
	codec->card->dev = socdev->dev;
	codec->card->private_data = codec;
	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));

	/* create the pcms */
1345 1346
	for (i = 0; i < card->num_links; i++) {
		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
F
Frank Mandarino 已提交
1347 1348
		if (ret < 0) {
			printk(KERN_ERR "asoc: can't create pcm %s\n",
1349
				card->dai_link[i].stream_name);
F
Frank Mandarino 已提交
1350 1351 1352 1353 1354 1355 1356 1357 1358 1359 1360
			mutex_unlock(&codec->mutex);
			return ret;
		}
	}

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

/**
1361
 * snd_soc_init_card - register sound card
F
Frank Mandarino 已提交
1362 1363 1364 1365 1366 1367 1368
 * @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.
 */
1369
int snd_soc_init_card(struct snd_soc_device *socdev)
F
Frank Mandarino 已提交
1370
{
1371
	struct snd_soc_card *card = socdev->card;
1372
	struct snd_soc_codec *codec = card->codec;
1373
	int ret = 0, i, ac97 = 0, err = 0;
F
Frank Mandarino 已提交
1374

1375 1376 1377
	for (i = 0; i < card->num_links; i++) {
		if (card->dai_link[i].init) {
			err = card->dai_link[i].init(codec);
1378 1379
			if (err < 0) {
				printk(KERN_ERR "asoc: failed to init %s\n",
1380
					card->dai_link[i].stream_name);
1381 1382 1383
				continue;
			}
		}
M
Mark Brown 已提交
1384
		if (card->dai_link[i].codec_dai->ac97_control)
F
Frank Mandarino 已提交
1385 1386 1387
			ac97 = 1;
	}
	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
1388
		 "%s",  card->name);
F
Frank Mandarino 已提交
1389
	snprintf(codec->card->longname, sizeof(codec->card->longname),
1390
		 "%s (%s)", card->name, codec->name);
F
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1391 1392 1393

	ret = snd_card_register(codec->card);
	if (ret < 0) {
1394
		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
F
Frank Mandarino 已提交
1395
				codec->name);
1396
		goto out;
F
Frank Mandarino 已提交
1397 1398
	}

1399
	mutex_lock(&codec->mutex);
F
Frank Mandarino 已提交
1400
#ifdef CONFIG_SND_SOC_AC97_BUS
1401 1402 1403 1404
	/* Only instantiate AC97 if not already done by the adaptor
	 * for the generic AC97 subsystem.
	 */
	if (ac97 && strcmp(codec->name, "AC97") != 0) {
1405 1406 1407 1408
		ret = soc_ac97_dev_register(codec);
		if (ret < 0) {
			printk(KERN_ERR "asoc: AC97 device register failed\n");
			snd_card_free(codec->card);
1409
			mutex_unlock(&codec->mutex);
1410 1411 1412
			goto out;
		}
	}
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Frank Mandarino 已提交
1413 1414
#endif

1415 1416 1417 1418 1419 1420
	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)
1421
		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
1422

1423
	soc_init_codec_debugfs(codec);
F
Frank Mandarino 已提交
1424
	mutex_unlock(&codec->mutex);
1425 1426

out:
F
Frank Mandarino 已提交
1427 1428
	return ret;
}
1429
EXPORT_SYMBOL_GPL(snd_soc_init_card);
F
Frank Mandarino 已提交
1430 1431 1432 1433 1434 1435 1436 1437 1438 1439

/**
 * 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)
{
1440
	struct snd_soc_codec *codec = socdev->card->codec;
1441
#ifdef CONFIG_SND_SOC_AC97_BUS
1442
	struct snd_soc_dai *codec_dai;
1443 1444
	int i;
#endif
F
Frank Mandarino 已提交
1445 1446

	mutex_lock(&codec->mutex);
1447
	soc_cleanup_codec_debugfs(codec);
F
Frank Mandarino 已提交
1448
#ifdef CONFIG_SND_SOC_AC97_BUS
1449
	for (i = 0; i < codec->num_dai; i++) {
1450
		codec_dai = &codec->dai[i];
1451 1452
		if (codec_dai->ac97_control && codec->ac97 &&
		    strcmp(codec->name, "AC97") != 0) {
1453 1454 1455 1456 1457
			soc_ac97_dev_unregister(codec);
			goto free_card;
		}
	}
free_card:
F
Frank Mandarino 已提交
1458 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 1484 1485 1486 1487 1488 1489 1490 1491 1492 1493
#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
M
Mark Brown 已提交
1494
 * @long_name: control long name
F
Frank Mandarino 已提交
1495 1496 1497 1498 1499 1500 1501 1502 1503 1504 1505 1506 1507 1508 1509 1510 1511 1512 1513
 *
 * 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);

I
Ian Molton 已提交
1514 1515 1516 1517 1518 1519 1520 1521 1522 1523 1524 1525 1526 1527 1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544
/**
 * snd_soc_add_controls - add an array of controls to a codec.
 * Convienience function to add a list of controls. Many codecs were
 * duplicating this code.
 *
 * @codec: codec to add controls to
 * @controls: array of controls to add
 * @num_controls: number of elements in the array
 *
 * Return 0 for success, else error.
 */
int snd_soc_add_controls(struct snd_soc_codec *codec,
	const struct snd_kcontrol_new *controls, int num_controls)
{
	struct snd_card *card = codec->card;
	int err, i;

	for (i = 0; i < num_controls; i++) {
		const struct snd_kcontrol_new *control = &controls[i];
		err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
		if (err < 0) {
			dev_err(codec->dev, "%s: Failed to add %s\n",
				codec->name, control->name);
			return err;
		}
	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_add_controls);

F
Frank Mandarino 已提交
1545 1546 1547 1548 1549 1550 1551 1552 1553 1554 1555 1556 1557 1558 1559 1560 1561
/**
 * 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;
1562
	uinfo->value.enumerated.items = e->max;
F
Frank Mandarino 已提交
1563

1564 1565
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
F
Frank Mandarino 已提交
1566 1567 1568 1569 1570 1571 1572 1573 1574
	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
M
Mark Brown 已提交
1575
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1576 1577 1578 1579 1580 1581 1582 1583 1584 1585 1586 1587
 *
 * 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;

1588
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1589 1590
		;
	val = snd_soc_read(codec, e->reg);
1591 1592
	ucontrol->value.enumerated.item[0]
		= (val >> e->shift_l) & (bitmask - 1);
F
Frank Mandarino 已提交
1593 1594 1595 1596 1597 1598 1599 1600 1601 1602 1603
	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
M
Mark Brown 已提交
1604
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1605 1606 1607 1608 1609 1610 1611 1612 1613 1614 1615 1616 1617
 *
 * 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;

1618
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1619
		;
1620
	if (ucontrol->value.enumerated.item[0] > e->max - 1)
F
Frank Mandarino 已提交
1621 1622 1623 1624
		return -EINVAL;
	val = ucontrol->value.enumerated.item[0] << e->shift_l;
	mask = (bitmask - 1) << e->shift_l;
	if (e->shift_l != e->shift_r) {
1625
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
F
Frank Mandarino 已提交
1626 1627 1628 1629 1630 1631 1632 1633 1634
			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);

P
Peter Ujfalusi 已提交
1635 1636 1637 1638 1639 1640 1641 1642 1643 1644 1645 1646 1647 1648 1649 1650
/**
 * snd_soc_get_value_enum_double - semi enumerated double mixer get callback
 * @kcontrol: mixer control
 * @ucontrol: control element information
 *
 * Callback to get the value of a double semi enumerated mixer.
 *
 * Semi enumerated mixer: the enumerated items are referred as values. Can be
 * used for handling bitfield coded enumeration for example.
 *
 * Returns 0 for success.
 */
int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1651
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
P
Peter Ujfalusi 已提交
1652 1653 1654 1655 1656 1657 1658 1659 1660 1661 1662 1663 1664 1665 1666 1667 1668 1669 1670 1671 1672 1673 1674 1675 1676 1677 1678 1679 1680 1681 1682 1683 1684 1685 1686 1687 1688 1689
	unsigned short reg_val, val, mux;

	reg_val = snd_soc_read(codec, e->reg);
	val = (reg_val >> e->shift_l) & e->mask;
	for (mux = 0; mux < e->max; mux++) {
		if (val == e->values[mux])
			break;
	}
	ucontrol->value.enumerated.item[0] = mux;
	if (e->shift_l != e->shift_r) {
		val = (reg_val >> e->shift_r) & e->mask;
		for (mux = 0; mux < e->max; mux++) {
			if (val == e->values[mux])
				break;
		}
		ucontrol->value.enumerated.item[1] = mux;
	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double);

/**
 * snd_soc_put_value_enum_double - semi enumerated double mixer put callback
 * @kcontrol: mixer control
 * @ucontrol: control element information
 *
 * Callback to set the value of a double semi enumerated mixer.
 *
 * Semi enumerated mixer: the enumerated items are referred as values. Can be
 * used for handling bitfield coded enumeration for example.
 *
 * Returns 0 for success.
 */
int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1690
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
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Peter Ujfalusi 已提交
1691 1692 1693 1694 1695 1696 1697 1698 1699 1700 1701 1702 1703 1704 1705 1706 1707 1708
	unsigned short val;
	unsigned short mask;

	if (ucontrol->value.enumerated.item[0] > e->max - 1)
		return -EINVAL;
	val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l;
	mask = e->mask << e->shift_l;
	if (e->shift_l != e->shift_r) {
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
			return -EINVAL;
		val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r;
		mask |= e->mask << e->shift_r;
	}

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

F
Frank Mandarino 已提交
1709 1710 1711 1712 1713 1714 1715 1716 1717 1718 1719 1720 1721 1722 1723 1724 1725
/**
 * 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;
1726
	uinfo->value.enumerated.items = e->max;
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Frank Mandarino 已提交
1727

1728 1729
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
F
Frank Mandarino 已提交
1730 1731 1732 1733 1734 1735 1736 1737 1738 1739 1740 1741 1742 1743 1744 1745 1746 1747
	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)
{
P
Philipp Zabel 已提交
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	int max = kcontrol->private_value;

1750
	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
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		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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1754 1755 1756

	uinfo->count = 1;
	uinfo->value.integer.min = 0;
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Philipp Zabel 已提交
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	uinfo->value.integer.max = max;
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Frank Mandarino 已提交
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	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)
{
1774 1775 1776
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
1777
	unsigned int shift = mc->shift;
1778
	unsigned int rshift = mc->rshift;
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Frank Mandarino 已提交
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1780
	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
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Philipp Zabel 已提交
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		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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	uinfo->value.integer.max = max;
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	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);

/**
 * snd_soc_get_volsw - single mixer get callback
 * @kcontrol: mixer control
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 * @ucontrol: control element information
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 *
 * 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)
{
1804 1805
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1807 1808 1809
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1810
	int max = mc->max;
1811 1812
	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;
	if (shift != rshift)
		ucontrol->value.integer.value[1] =
			(snd_soc_read(codec, reg) >> rshift) & mask;
	if (invert) {
		ucontrol->value.integer.value[0] =
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Philipp Zabel 已提交
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			max - ucontrol->value.integer.value[0];
F
Frank Mandarino 已提交
1822 1823
		if (shift != rshift)
			ucontrol->value.integer.value[1] =
P
Philipp Zabel 已提交
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				max - ucontrol->value.integer.value[1];
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Frank Mandarino 已提交
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	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw);

/**
 * snd_soc_put_volsw - single mixer put callback
 * @kcontrol: mixer control
M
Mark Brown 已提交
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 * @ucontrol: control element information
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 *
 * 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)
{
1843 1844
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
1845
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1846 1847 1848
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1849
	int max = mc->max;
1850 1851
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
F
Frank Mandarino 已提交
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	unsigned short val, val2, val_mask;

	val = (ucontrol->value.integer.value[0] & mask);
	if (invert)
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Philipp Zabel 已提交
1856
		val = max - val;
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Frank Mandarino 已提交
1857 1858 1859 1860 1861
	val_mask = mask << shift;
	val = val << shift;
	if (shift != rshift) {
		val2 = (ucontrol->value.integer.value[1] & mask);
		if (invert)
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Philipp Zabel 已提交
1862
			val2 = max - val2;
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1863 1864 1865
		val_mask |= mask << rshift;
		val |= val2 << rshift;
	}
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Philipp Zabel 已提交
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	return snd_soc_update_bits(codec, reg, val_mask, val);
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Frank Mandarino 已提交
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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)
{
1883 1884 1885
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
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Philipp Zabel 已提交
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1887
	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
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Philipp Zabel 已提交
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		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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Frank Mandarino 已提交
1891 1892 1893

	uinfo->count = 2;
	uinfo->value.integer.min = 0;
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Philipp Zabel 已提交
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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
M
Mark Brown 已提交
1902
 * @ucontrol: control element information
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Frank Mandarino 已提交
1903 1904 1905 1906 1907 1908 1909 1910
 *
 * 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)
{
1911 1912
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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Frank Mandarino 已提交
1913
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1914 1915 1916
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1917
	int max = mc->max;
1918 1919
	unsigned int mask = (1<<fls(max))-1;
	unsigned int invert = mc->invert;
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Frank Mandarino 已提交
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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] =
P
Philipp Zabel 已提交
1927
			max - ucontrol->value.integer.value[0];
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Frank Mandarino 已提交
1928
		ucontrol->value.integer.value[1] =
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Philipp Zabel 已提交
1929
			max - ucontrol->value.integer.value[1];
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Frank Mandarino 已提交
1930 1931 1932 1933 1934 1935 1936 1937 1938
	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);

/**
 * snd_soc_put_volsw_2r - double mixer set callback
 * @kcontrol: mixer control
M
Mark Brown 已提交
1939
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1940 1941 1942 1943 1944 1945 1946 1947
 *
 * 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)
{
1948 1949
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
1950
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1951 1952 1953
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1954
	int max = mc->max;
1955 1956
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
F
Frank Mandarino 已提交
1957 1958 1959 1960 1961 1962 1963 1964
	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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Philipp Zabel 已提交
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		val = max - val;
		val2 = max - val2;
F
Frank Mandarino 已提交
1967 1968 1969 1970 1971
	}

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

1972 1973
	err = snd_soc_update_bits(codec, reg, val_mask, val);
	if (err < 0)
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Frank Mandarino 已提交
1974 1975 1976 1977 1978 1979 1980
		return err;

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

1981 1982 1983 1984 1985 1986 1987 1988 1989 1990 1991 1992
/**
 * 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)
{
1993 1994 1995 1996
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
	int min = mc->min;
1997 1998 1999 2000 2001 2002 2003 2004 2005 2006 2007 2008

	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
M
Mark Brown 已提交
2009
 * @ucontrol: control element information
2010 2011 2012 2013 2014 2015 2016 2017
 *
 * 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)
{
2018 2019
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
2020
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2021
	unsigned int reg = mc->reg;
2022
	int min = mc->min;
2023 2024 2025 2026 2027 2028 2029 2030 2031 2032 2033 2034 2035
	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
M
Mark Brown 已提交
2036
 * @ucontrol: control element information
2037 2038 2039 2040 2041 2042 2043 2044
 *
 * 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)
{
2045 2046
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
2047
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2048
	unsigned int reg = mc->reg;
2049
	int min = mc->min;
2050 2051 2052 2053 2054 2055 2056 2057 2058
	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);

2059 2060 2061 2062 2063 2064 2065 2066 2067 2068 2069 2070
/**
 * 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)
{
2071
	if (dai->ops && dai->ops->set_sysclk)
2072
		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
2073 2074 2075 2076 2077 2078 2079 2080
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);

/**
 * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
 * @dai: DAI
M
Mark Brown 已提交
2081
 * @div_id: DAI specific clock divider ID
2082 2083 2084 2085 2086 2087 2088 2089 2090
 * @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)
{
2091
	if (dai->ops && dai->ops->set_clkdiv)
2092
		return dai->ops->set_clkdiv(dai, div_id, div);
2093 2094 2095 2096 2097 2098 2099 2100 2101 2102 2103 2104 2105 2106 2107 2108 2109
	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)
{
2110
	if (dai->ops && dai->ops->set_pll)
2111
		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
2112 2113 2114 2115 2116 2117 2118 2119 2120 2121 2122 2123 2124 2125
	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)
{
2126
	if (dai->ops && dai->ops->set_fmt)
2127
		return dai->ops->set_fmt(dai, fmt);
2128 2129 2130 2131 2132 2133 2134 2135 2136 2137 2138 2139 2140 2141 2142 2143 2144
	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)
{
2145
	if (dai->ops && dai->ops->set_tdm_slot)
2146
		return dai->ops->set_tdm_slot(dai, mask, slots);
2147 2148 2149 2150 2151 2152 2153 2154 2155 2156 2157 2158 2159 2160
	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)
{
2161
	if (dai->ops && dai->ops->set_tristate)
2162
		return dai->ops->set_tristate(dai, tristate);
2163 2164 2165 2166 2167 2168 2169 2170 2171 2172 2173 2174 2175 2176
	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)
{
2177
	if (dai->ops && dai->ops->digital_mute)
2178
		return dai->ops->digital_mute(dai, mute);
2179 2180 2181 2182 2183
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);

M
Mark Brown 已提交
2184 2185 2186
/**
 * snd_soc_register_card - Register a card with the ASoC core
 *
M
Mark Brown 已提交
2187
 * @card: Card to register
M
Mark Brown 已提交
2188 2189 2190 2191 2192 2193 2194 2195 2196 2197 2198 2199 2200 2201 2202
 *
 * 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);
2203
	snd_soc_instantiate_cards();
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Mark Brown 已提交
2204 2205 2206 2207 2208 2209 2210 2211 2212 2213
	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
 *
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 * @card: Card to unregister
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 *
 * 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 struct snd_soc_dai_ops null_dai_ops = {
};

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/**
 * snd_soc_register_dai - Register a DAI with the ASoC core
 *
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 * @dai: DAI to register
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 */
int snd_soc_register_dai(struct snd_soc_dai *dai)
{
	if (!dai->name)
		return -EINVAL;

	/* The device should become mandatory over time */
	if (!dai->dev)
		printk(KERN_WARNING "No device for DAI %s\n", dai->name);

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	if (!dai->ops)
		dai->ops = &null_dai_ops;

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	INIT_LIST_HEAD(&dai->list);

	mutex_lock(&client_mutex);
	list_add(&dai->list, &dai_list);
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	snd_soc_instantiate_cards();
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	mutex_unlock(&client_mutex);

	pr_debug("Registered DAI '%s'\n", dai->name);

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_dai);

/**
 * snd_soc_unregister_dai - Unregister a DAI from the ASoC core
 *
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 * @dai: DAI to unregister
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 */
void snd_soc_unregister_dai(struct snd_soc_dai *dai)
{
	mutex_lock(&client_mutex);
	list_del(&dai->list);
	mutex_unlock(&client_mutex);

	pr_debug("Unregistered DAI '%s'\n", dai->name);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_dai);

/**
 * snd_soc_register_dais - Register multiple DAIs with the ASoC core
 *
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 * @dai: Array of DAIs to register
 * @count: Number of DAIs
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 */
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count)
{
	int i, ret;

	for (i = 0; i < count; i++) {
		ret = snd_soc_register_dai(&dai[i]);
		if (ret != 0)
			goto err;
	}

	return 0;

err:
	for (i--; i >= 0; i--)
		snd_soc_unregister_dai(&dai[i]);

	return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_register_dais);

/**
 * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core
 *
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 * @dai: Array of DAIs to unregister
 * @count: Number of DAIs
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 */
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count)
{
	int i;

	for (i = 0; i < count; i++)
		snd_soc_unregister_dai(&dai[i]);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_dais);

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/**
 * snd_soc_register_platform - Register a platform with the ASoC core
 *
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 * @platform: platform to register
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 */
int snd_soc_register_platform(struct snd_soc_platform *platform)
{
	if (!platform->name)
		return -EINVAL;

	INIT_LIST_HEAD(&platform->list);

	mutex_lock(&client_mutex);
	list_add(&platform->list, &platform_list);
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	snd_soc_instantiate_cards();
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	mutex_unlock(&client_mutex);

	pr_debug("Registered platform '%s'\n", platform->name);

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_platform);

/**
 * snd_soc_unregister_platform - Unregister a platform from the ASoC core
 *
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 * @platform: platform to unregister
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 */
void snd_soc_unregister_platform(struct snd_soc_platform *platform)
{
	mutex_lock(&client_mutex);
	list_del(&platform->list);
	mutex_unlock(&client_mutex);

	pr_debug("Unregistered platform '%s'\n", platform->name);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);

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static u64 codec_format_map[] = {
	SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
	SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
	SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
	SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
	SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
	SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
	SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
	SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
	SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
	SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
	SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
	SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
	SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
	| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
};

/* Fix up the DAI formats for endianness: codecs don't actually see
 * the endianness of the data but we're using the CPU format
 * definitions which do need to include endianness so we ensure that
 * codec DAIs always have both big and little endian variants set.
 */
static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
{
	int i;

	for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
		if (stream->formats & codec_format_map[i])
			stream->formats |= codec_format_map[i];
}

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/**
 * snd_soc_register_codec - Register a codec with the ASoC core
 *
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 * @codec: codec to register
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 */
int snd_soc_register_codec(struct snd_soc_codec *codec)
{
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	int i;

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	if (!codec->name)
		return -EINVAL;

	/* The device should become mandatory over time */
	if (!codec->dev)
		printk(KERN_WARNING "No device for codec %s\n", codec->name);

	INIT_LIST_HEAD(&codec->list);

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	for (i = 0; i < codec->num_dai; i++) {
		fixup_codec_formats(&codec->dai[i].playback);
		fixup_codec_formats(&codec->dai[i].capture);
	}

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	mutex_lock(&client_mutex);
	list_add(&codec->list, &codec_list);
	snd_soc_instantiate_cards();
	mutex_unlock(&client_mutex);

	pr_debug("Registered codec '%s'\n", codec->name);

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_codec);

/**
 * snd_soc_unregister_codec - Unregister a codec from the ASoC core
 *
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 * @codec: codec to unregister
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 */
void snd_soc_unregister_codec(struct snd_soc_codec *codec)
{
	mutex_lock(&client_mutex);
	list_del(&codec->list);
	mutex_unlock(&client_mutex);

	pr_debug("Unregistered codec '%s'\n", codec->name);
}
EXPORT_SYMBOL_GPL(snd_soc_unregister_codec);

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static int __init snd_soc_init(void)
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{
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#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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{
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#ifdef CONFIG_DEBUG_FS
	debugfs_remove_recursive(debugfs_root);
#endif
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	platform_driver_unregister(&soc_driver);
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}

module_init(snd_soc_init);
module_exit(snd_soc_exit);

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