soc-core.c 67.4 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>
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#include <sound/ac97_codec.h>
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#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;
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;
553
	struct snd_soc_codec *codec = card->codec;
F
Frank Mandarino 已提交
554 555 556 557

	mutex_lock(&pcm_mutex);

	/* apply codec digital mute */
558 559
	if (!codec->active)
		snd_soc_dai_digital_mute(codec_dai, 1);
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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  */
570 571
	if (codec_dai->ops->hw_free)
		codec_dai->ops->hw_free(substream, codec_dai);
F
Frank Mandarino 已提交
572

573 574
	if (cpu_dai->ops->hw_free)
		cpu_dai->ops->hw_free(substream, cpu_dai);
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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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589 590
	int ret;

591 592
	if (codec_dai->ops->trigger) {
		ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
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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;
	}

603 604
	if (cpu_dai->ops->trigger) {
		ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
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605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622
		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 */
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623
static int soc_suspend(struct device *dev)
F
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624
{
M
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625
	struct platform_device *pdev = to_platform_device(dev);
626
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
627
	struct snd_soc_card *card = socdev->card;
628
	struct snd_soc_platform *platform = card->platform;
629
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
630
	struct snd_soc_codec *codec = card->codec;
F
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631 632
	int i;

633 634 635 636 637 638
	/* 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;

639 640 641 642 643 644 645 646 647 648
	/* 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);

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649
	/* mute any active DAC's */
650 651
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
652 653
		if (dai->ops->digital_mute && dai->playback.active)
			dai->ops->digital_mute(dai, 1);
F
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654 655
	}

656
	/* suspend all pcms */
657 658
	for (i = 0; i < card->num_links; i++)
		snd_pcm_suspend_all(card->dai_link[i].pcm);
659

660
	if (card->suspend_pre)
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661
		card->suspend_pre(pdev, PMSG_SUSPEND);
F
Frank Mandarino 已提交
662

663 664
	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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665
		if (cpu_dai->suspend && !cpu_dai->ac97_control)
666
			cpu_dai->suspend(cpu_dai);
F
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667
		if (platform->suspend)
668
			platform->suspend(cpu_dai);
F
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669 670 671
	}

	/* close any waiting streams and save state */
672
	run_delayed_work(&card->delayed_work);
673
	codec->suspend_bias_level = codec->bias_level;
F
Frank Mandarino 已提交
674

675
	for (i = 0; i < codec->num_dai; i++) {
F
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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)
M
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687
		codec_dev->suspend(pdev, PMSG_SUSPEND);
F
Frank Mandarino 已提交
688

689 690
	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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691
		if (cpu_dai->suspend && cpu_dai->ac97_control)
692
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
693 694
	}

695
	if (card->suspend_post)
M
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696
		card->suspend_post(pdev, PMSG_SUSPEND);
F
Frank Mandarino 已提交
697 698 699 700

	return 0;
}

701 702 703 704
/* 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
705
{
706 707 708 709
	struct snd_soc_card *card = container_of(work,
						 struct snd_soc_card,
						 deferred_resume_work);
	struct snd_soc_device *socdev = card->socdev;
710
	struct snd_soc_platform *platform = card->platform;
711
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
712
	struct snd_soc_codec *codec = card->codec;
713
	struct platform_device *pdev = to_platform_device(socdev->dev);
F
Frank Mandarino 已提交
714 715
	int i;

716 717 718 719
	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
	 * so userspace apps are blocked from touching us
	 */

720
	dev_dbg(socdev->dev, "starting resume work\n");
721

722 723
	if (card->resume_pre)
		card->resume_pre(pdev);
F
Frank Mandarino 已提交
724

725 726
	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
727
		if (cpu_dai->resume && cpu_dai->ac97_control)
728
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
729 730 731 732 733
	}

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

734 735
	for (i = 0; i < codec->num_dai; i++) {
		char *stream = codec->dai[i].playback.stream_name;
F
Frank Mandarino 已提交
736 737 738 739 740 741 742 743 744
		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);
	}

745
	/* unmute any active DACs */
746 747
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
748 749
		if (dai->ops->digital_mute && dai->playback.active)
			dai->ops->digital_mute(dai, 0);
F
Frank Mandarino 已提交
750 751
	}

752 753
	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
754
		if (cpu_dai->resume && !cpu_dai->ac97_control)
755
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
756
		if (platform->resume)
757
			platform->resume(cpu_dai);
F
Frank Mandarino 已提交
758 759
	}

760 761
	if (card->resume_post)
		card->resume_post(pdev);
F
Frank Mandarino 已提交
762

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

	/* 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 */
M
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770
static int soc_resume(struct device *dev)
771
{
M
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772
	struct platform_device *pdev = to_platform_device(dev);
773
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
774
	struct snd_soc_card *card = socdev->card;
775
	struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai;
776

777 778 779 780 781 782 783 784 785 786 787 788 789
	/* 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");
	}
790

F
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791 792 793
	return 0;
}

794 795 796 797 798 799 800 801 802 803 804 805 806 807 808 809 810 811 812 813 814 815 816 817 818 819 820 821 822 823 824 825 826 827 828 829 830 831
/**
 * snd_soc_suspend_device: Notify core of device suspend
 *
 * @dev: Device being suspended.
 *
 * In order to ensure that the entire audio subsystem is suspended in a
 * coordinated fashion ASoC devices should suspend themselves when
 * called by ASoC.  When the standard kernel suspend process asks the
 * device to suspend it should call this function to initiate a suspend
 * of the entire ASoC card.
 *
 * \note Currently this function is stubbed out.
 */
int snd_soc_suspend_device(struct device *dev)
{
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_suspend_device);

/**
 * snd_soc_resume_device: Notify core of device resume
 *
 * @dev: Device being resumed.
 *
 * In order to ensure that the entire audio subsystem is resumed in a
 * coordinated fashion ASoC devices should resume themselves when called
 * by ASoC.  When the standard kernel resume process asks the device
 * to resume it should call this function.  Once all the components of
 * the card have notified that they are ready to be resumed the card
 * will be resumed.
 *
 * \note Currently this function is stubbed out.
 */
int snd_soc_resume_device(struct device *dev)
{
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_resume_device);
F
Frank Mandarino 已提交
832 833 834 835 836
#else
#define soc_suspend	NULL
#define soc_resume	NULL
#endif

837
static void snd_soc_instantiate_card(struct snd_soc_card *card)
F
Frank Mandarino 已提交
838
{
839 840 841 842 843 844
	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;
845
	int i, found, ret, ac97;
846 847 848 849 850 851 852 853 854 855 856 857 858 859 860

	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;
	}
861

862
	ac97 = 0;
863 864 865 866 867 868 869 870 871 872 873 874
	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;
		}
875 876 877

		if (card->dai_link[i].cpu_dai->ac97_control)
			ac97 = 1;
M
Mark Brown 已提交
878 879
	}

880 881 882 883 884 885 886 887 888 889 890 891 892 893 894 895 896 897 898 899 900
	/* 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;
			}
		}

901 902 903 904 905
	/* Note that we do not current check for codec components */

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

	/* Found everything, bring it up */
906 907
	if (card->probe) {
		ret = card->probe(pdev);
908
		if (ret < 0)
909
			return;
F
Frank Mandarino 已提交
910 911
	}

912 913
	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
914
		if (cpu_dai->probe) {
915
			ret = cpu_dai->probe(pdev, cpu_dai);
916
			if (ret < 0)
F
Frank Mandarino 已提交
917 918 919 920 921 922
				goto cpu_dai_err;
		}
	}

	if (codec_dev->probe) {
		ret = codec_dev->probe(pdev);
923
		if (ret < 0)
F
Frank Mandarino 已提交
924 925 926 927 928
			goto cpu_dai_err;
	}

	if (platform->probe) {
		ret = platform->probe(pdev);
929
		if (ret < 0)
F
Frank Mandarino 已提交
930 931 932 933
			goto platform_err;
	}

	/* DAPM stream work */
934
	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
R
Randy Dunlap 已提交
935
#ifdef CONFIG_PM
936
	/* deferred resume work */
937
	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
R
Randy Dunlap 已提交
938
#endif
939

940 941 942
	card->instantiated = 1;

	return;
F
Frank Mandarino 已提交
943 944 945 946 947 948

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

cpu_dai_err:
949
	for (i--; i >= 0; i--) {
950
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
951
		if (cpu_dai->remove)
952
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
953 954
	}

955 956
	if (card->remove)
		card->remove(pdev);
957
}
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958

959 960 961 962 963 964 965 966 967 968 969 970 971 972 973 974 975 976 977 978 979 980 981 982 983 984 985 986 987 988
/*
 * 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
989 990 991 992 993 994 995
}

/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
	int i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
996
	struct snd_soc_card *card = socdev->card;
997
	struct snd_soc_platform *platform = card->platform;
F
Frank Mandarino 已提交
998 999
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

1000 1001 1002
	if (!card->instantiated)
		return 0;

1003
	run_delayed_work(&card->delayed_work);
1004

F
Frank Mandarino 已提交
1005 1006 1007 1008 1009 1010
	if (platform->remove)
		platform->remove(pdev);

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

1011 1012
	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
1013
		if (cpu_dai->remove)
1014
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
1015 1016
	}

1017 1018
	if (card->remove)
		card->remove(pdev);
F
Frank Mandarino 已提交
1019

M
Mark Brown 已提交
1020 1021
	snd_soc_unregister_card(card);

F
Frank Mandarino 已提交
1022 1023 1024
	return 0;
}

M
Mark Brown 已提交
1025
static int soc_poweroff(struct device *dev)
M
Mark Brown 已提交
1026
{
M
Mark Brown 已提交
1027
	struct platform_device *pdev = to_platform_device(dev);
M
Mark Brown 已提交
1028 1029 1030 1031
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_card *card = socdev->card;

	if (!card->instantiated)
M
Mark Brown 已提交
1032
		return 0;
M
Mark Brown 已提交
1033 1034 1035 1036 1037 1038

	/* Flush out pmdown_time work - we actually do want to run it
	 * now, we're shutting down so no imminent restart. */
	run_delayed_work(&card->delayed_work);

	snd_soc_dapm_shutdown(socdev);
M
Mark Brown 已提交
1039 1040

	return 0;
M
Mark Brown 已提交
1041 1042
}

M
Mark Brown 已提交
1043 1044 1045 1046 1047 1048
static struct dev_pm_ops soc_pm_ops = {
	.suspend = soc_suspend,
	.resume = soc_resume,
	.poweroff = soc_poweroff,
};

F
Frank Mandarino 已提交
1049 1050 1051 1052
/* ASoC platform driver */
static struct platform_driver soc_driver = {
	.driver		= {
		.name		= "soc-audio",
1053
		.owner		= THIS_MODULE,
M
Mark Brown 已提交
1054
		.pm		= &soc_pm_ops,
F
Frank Mandarino 已提交
1055 1056 1057 1058 1059 1060 1061 1062 1063
	},
	.probe		= soc_probe,
	.remove		= soc_remove,
};

/* create a new pcm */
static int soc_new_pcm(struct snd_soc_device *socdev,
	struct snd_soc_dai_link *dai_link, int num)
{
1064
	struct snd_soc_card *card = socdev->card;
1065
	struct snd_soc_codec *codec = card->codec;
1066
	struct snd_soc_platform *platform = card->platform;
1067 1068
	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
1069 1070 1071 1072 1073 1074 1075 1076
	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;
1077 1078

	rtd->dai = dai_link;
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1079
	rtd->socdev = socdev;
1080
	codec_dai->codec = card->codec;
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1081 1082

	/* check client and interface hw capabilities */
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	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
		num);
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1085 1086 1087 1088 1089 1090 1091 1092 1093

	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) {
1094 1095
		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
			codec->name);
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1096 1097 1098 1099
		kfree(rtd);
		return ret;
	}

1100
	dai_link->pcm = pcm;
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	pcm->private_data = rtd;
1102 1103 1104 1105 1106 1107 1108
	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;
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	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);

1116
	ret = platform->pcm_new(codec->card, codec_dai, pcm);
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1117 1118 1119 1120 1121 1122
	if (ret < 0) {
		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
		kfree(rtd);
		return ret;
	}

1123
	pcm->private_free = platform->pcm_free;
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	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
		cpu_dai->name);
	return ret;
}

1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145
/**
 * snd_soc_codec_volatile_register: Report if a register is volatile.
 *
 * @codec: CODEC to query.
 * @reg: Register to query.
 *
 * Boolean function indiciating if a CODEC register is volatile.
 */
int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg)
{
	if (codec->volatile_register)
		return codec->volatile_register(reg);
	else
		return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register);

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1146
/* codec register dump */
1147
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
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{
	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);
1158
	for (i = 0; i < codec->reg_cache_size; i += step) {
1159 1160 1161
		if (codec->readable_register && !codec->readable_register(i))
			continue;

1162 1163 1164 1165 1166 1167 1168 1169 1170 1171 1172 1173 1174 1175 1176 1177 1178 1179 1180 1181 1182 1183
		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;
}
1187 1188 1189 1190
static ssize_t codec_reg_show(struct device *dev,
	struct device_attribute *attr, char *buf)
{
	struct snd_soc_device *devdata = dev_get_drvdata(dev);
1191
	return soc_codec_reg_show(devdata->card->codec, buf);
1192 1193
}

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

1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206
#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;
1207
	struct snd_soc_codec *codec = file->private_data;
1208 1209 1210
	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
	if (!buf)
		return -ENOMEM;
1211
	ret = soc_codec_reg_show(codec, buf);
1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222 1223 1224 1225
	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;
1226
	struct snd_soc_codec *codec = file->private_data;
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

	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,
};

1255
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1256
{
1257 1258 1259 1260 1261 1262 1263 1264 1265 1266 1267 1268 1269
	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");
1270 1271 1272 1273 1274 1275 1276

	codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root);
	if (!codec->debugfs_dapm)
		printk(KERN_WARNING
		       "Failed to create DAPM debugfs directory\n");

	snd_soc_dapm_debugfs_init(codec);
1277 1278
}

1279
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1280
{
1281
	debugfs_remove_recursive(codec->debugfs_dapm);
1282 1283
	debugfs_remove(codec->debugfs_pop_time);
	debugfs_remove(codec->debugfs_reg);
1284 1285 1286 1287
}

#else

1288
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1289 1290 1291
{
}

1292
static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1293 1294 1295 1296
{
}
#endif

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1297 1298 1299 1300 1301 1302 1303 1304 1305 1306 1307 1308 1309 1310 1311 1312 1313 1314 1315 1316 1317 1318 1319 1320 1321 1322 1323 1324 1325 1326 1327 1328 1329 1330 1331 1332 1333 1334 1335 1336 1337 1338 1339 1340 1341 1342 1343 1344 1345 1346 1347 1348 1349 1350 1351 1352 1353 1354 1355 1356 1357 1358
/**
 * 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,
1359
				unsigned int mask, unsigned int value)
F
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1360 1361
{
	int change;
1362
	unsigned int old, new;
F
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1363 1364 1365 1366 1367 1368 1369 1370 1371 1372 1373 1374 1375 1376 1377 1378 1379 1380 1381 1382 1383 1384 1385 1386 1387 1388

	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,
1389
				unsigned int mask, unsigned int value)
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1390 1391
{
	int change;
1392
	unsigned int old, new;
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1393 1394 1395 1396 1397 1398 1399 1400 1401 1402 1403 1404 1405 1406

	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
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1407 1408
 * @idx: ALSA card index
 * @xid: card identification
F
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1409 1410 1411 1412 1413
 *
 * Create a new sound card based upon the codec and interface pcms.
 *
 * Returns 0 for success, else error.
 */
1414
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
F
Frank Mandarino 已提交
1415
{
1416
	struct snd_soc_card *card = socdev->card;
1417
	struct snd_soc_codec *codec = card->codec;
1418
	int ret, i;
F
Frank Mandarino 已提交
1419 1420 1421 1422

	mutex_lock(&codec->mutex);

	/* register a sound card */
1423 1424
	ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
	if (ret < 0) {
F
Frank Mandarino 已提交
1425 1426 1427
		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
			codec->name);
		mutex_unlock(&codec->mutex);
1428
		return ret;
F
Frank Mandarino 已提交
1429 1430
	}

1431
	codec->socdev = socdev;
F
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1432 1433 1434 1435 1436
	codec->card->dev = socdev->dev;
	codec->card->private_data = codec;
	strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));

	/* create the pcms */
1437 1438
	for (i = 0; i < card->num_links; i++) {
		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
F
Frank Mandarino 已提交
1439 1440
		if (ret < 0) {
			printk(KERN_ERR "asoc: can't create pcm %s\n",
1441
				card->dai_link[i].stream_name);
F
Frank Mandarino 已提交
1442 1443 1444 1445 1446 1447 1448 1449 1450 1451 1452
			mutex_unlock(&codec->mutex);
			return ret;
		}
	}

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

/**
1453
 * snd_soc_init_card - register sound card
F
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1454 1455 1456 1457 1458 1459 1460
 * @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.
 */
1461
int snd_soc_init_card(struct snd_soc_device *socdev)
F
Frank Mandarino 已提交
1462
{
1463
	struct snd_soc_card *card = socdev->card;
1464
	struct snd_soc_codec *codec = card->codec;
1465
	int ret = 0, i, ac97 = 0, err = 0;
F
Frank Mandarino 已提交
1466

1467 1468 1469
	for (i = 0; i < card->num_links; i++) {
		if (card->dai_link[i].init) {
			err = card->dai_link[i].init(codec);
1470 1471
			if (err < 0) {
				printk(KERN_ERR "asoc: failed to init %s\n",
1472
					card->dai_link[i].stream_name);
1473 1474 1475
				continue;
			}
		}
1476
		if (card->dai_link[i].codec_dai->ac97_control) {
F
Frank Mandarino 已提交
1477
			ac97 = 1;
1478 1479 1480
			snd_ac97_dev_add_pdata(codec->ac97,
				card->dai_link[i].cpu_dai->ac97_pdata);
		}
F
Frank Mandarino 已提交
1481 1482
	}
	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
1483
		 "%s",  card->name);
F
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1484
	snprintf(codec->card->longname, sizeof(codec->card->longname),
1485
		 "%s (%s)", card->name, codec->name);
F
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1486

1487 1488 1489
	/* Make sure all DAPM widgets are instantiated */
	snd_soc_dapm_new_widgets(codec);

F
Frank Mandarino 已提交
1490 1491
	ret = snd_card_register(codec->card);
	if (ret < 0) {
1492
		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
F
Frank Mandarino 已提交
1493
				codec->name);
1494
		goto out;
F
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1495 1496
	}

1497
	mutex_lock(&codec->mutex);
F
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1498
#ifdef CONFIG_SND_SOC_AC97_BUS
1499 1500 1501 1502
	/* Only instantiate AC97 if not already done by the adaptor
	 * for the generic AC97 subsystem.
	 */
	if (ac97 && strcmp(codec->name, "AC97") != 0) {
1503 1504 1505 1506
		ret = soc_ac97_dev_register(codec);
		if (ret < 0) {
			printk(KERN_ERR "asoc: AC97 device register failed\n");
			snd_card_free(codec->card);
1507
			mutex_unlock(&codec->mutex);
1508 1509 1510
			goto out;
		}
	}
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1511 1512
#endif

1513 1514 1515 1516 1517 1518
	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)
1519
		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
1520

1521
	soc_init_codec_debugfs(codec);
F
Frank Mandarino 已提交
1522
	mutex_unlock(&codec->mutex);
1523 1524

out:
F
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1525 1526
	return ret;
}
1527
EXPORT_SYMBOL_GPL(snd_soc_init_card);
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1528 1529 1530 1531 1532 1533 1534 1535 1536 1537

/**
 * 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)
{
1538
	struct snd_soc_codec *codec = socdev->card->codec;
1539
#ifdef CONFIG_SND_SOC_AC97_BUS
1540
	struct snd_soc_dai *codec_dai;
1541 1542
	int i;
#endif
F
Frank Mandarino 已提交
1543 1544

	mutex_lock(&codec->mutex);
1545
	soc_cleanup_codec_debugfs(codec);
F
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1546
#ifdef CONFIG_SND_SOC_AC97_BUS
1547
	for (i = 0; i < codec->num_dai; i++) {
1548
		codec_dai = &codec->dai[i];
1549 1550
		if (codec_dai->ac97_control && codec->ac97 &&
		    strcmp(codec->name, "AC97") != 0) {
1551 1552 1553 1554 1555
			soc_ac97_dev_unregister(codec);
			goto free_card;
		}
	}
free_card:
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1556 1557 1558 1559 1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571 1572 1573 1574 1575 1576 1577 1578 1579 1580 1581 1582 1583 1584 1585 1586 1587 1588 1589 1590 1591
#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
1592
 * @long_name: control long name
F
Frank Mandarino 已提交
1593 1594 1595 1596 1597 1598 1599 1600 1601 1602 1603 1604 1605 1606 1607 1608 1609 1610 1611
 *
 * 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
1612 1613 1614 1615 1616 1617 1618 1619 1620 1621 1622 1623 1624 1625 1626 1627 1628 1629 1630 1631 1632 1633 1634 1635 1636 1637 1638 1639 1640 1641 1642
/**
 * 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
1643 1644 1645 1646 1647 1648 1649 1650 1651 1652 1653 1654 1655 1656 1657 1658 1659
/**
 * 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;
1660
	uinfo->value.enumerated.items = e->max;
F
Frank Mandarino 已提交
1661

1662 1663
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
F
Frank Mandarino 已提交
1664 1665 1666 1667 1668 1669 1670 1671 1672
	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
1673
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1674 1675 1676 1677 1678 1679 1680 1681 1682 1683
 *
 * 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;
1684
	unsigned int val, bitmask;
F
Frank Mandarino 已提交
1685

1686
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1687 1688
		;
	val = snd_soc_read(codec, e->reg);
1689 1690
	ucontrol->value.enumerated.item[0]
		= (val >> e->shift_l) & (bitmask - 1);
F
Frank Mandarino 已提交
1691 1692 1693 1694 1695 1696 1697 1698 1699 1700 1701
	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
1702
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1703 1704 1705 1706 1707 1708 1709 1710 1711 1712
 *
 * 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;
1713 1714
	unsigned int val;
	unsigned int mask, bitmask;
F
Frank Mandarino 已提交
1715

1716
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1717
		;
1718
	if (ucontrol->value.enumerated.item[0] > e->max - 1)
F
Frank Mandarino 已提交
1719 1720 1721 1722
		return -EINVAL;
	val = ucontrol->value.enumerated.item[0] << e->shift_l;
	mask = (bitmask - 1) << e->shift_l;
	if (e->shift_l != e->shift_r) {
1723
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
F
Frank Mandarino 已提交
1724 1725 1726 1727 1728 1729 1730 1731 1732
			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
1733 1734 1735 1736 1737 1738 1739 1740 1741 1742 1743 1744 1745 1746 1747 1748
/**
 * 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);
1749
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
1750
	unsigned int reg_val, val, mux;
P
Peter Ujfalusi 已提交
1751 1752 1753 1754 1755 1756 1757 1758 1759 1760 1761 1762 1763 1764 1765 1766 1767 1768 1769 1770 1771 1772 1773 1774 1775 1776 1777 1778 1779 1780 1781 1782 1783 1784 1785 1786 1787

	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);
1788
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
1789 1790
	unsigned int val;
	unsigned int mask;
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Peter Ujfalusi 已提交
1791 1792 1793 1794 1795 1796 1797 1798 1799 1800 1801 1802 1803 1804 1805 1806

	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
1807 1808 1809 1810 1811 1812 1813 1814 1815 1816 1817 1818 1819 1820 1821 1822 1823
/**
 * 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;
1824
	uinfo->value.enumerated.items = e->max;
F
Frank Mandarino 已提交
1825

1826 1827
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
F
Frank Mandarino 已提交
1828 1829 1830 1831 1832 1833 1834 1835 1836 1837 1838 1839 1840 1841 1842 1843 1844 1845
	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
1846 1847
	int max = kcontrol->private_value;

1848
	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
P
Philipp Zabel 已提交
1849 1850 1851
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
F
Frank Mandarino 已提交
1852 1853 1854

	uinfo->count = 1;
	uinfo->value.integer.min = 0;
P
Philipp Zabel 已提交
1855
	uinfo->value.integer.max = max;
F
Frank Mandarino 已提交
1856 1857 1858 1859 1860 1861 1862 1863 1864 1865 1866 1867 1868 1869 1870 1871
	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)
{
1872 1873 1874
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
1875
	unsigned int shift = mc->shift;
1876
	unsigned int rshift = mc->rshift;
F
Frank Mandarino 已提交
1877

1878
	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
P
Philipp Zabel 已提交
1879 1880 1881 1882
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;

F
Frank Mandarino 已提交
1883 1884
	uinfo->count = shift == rshift ? 1 : 2;
	uinfo->value.integer.min = 0;
P
Philipp Zabel 已提交
1885
	uinfo->value.integer.max = max;
F
Frank Mandarino 已提交
1886 1887 1888 1889 1890 1891 1892
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);

/**
 * snd_soc_get_volsw - single mixer get callback
 * @kcontrol: mixer control
M
Mark Brown 已提交
1893
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1894 1895 1896 1897 1898 1899 1900 1901
 *
 * 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)
{
1902 1903
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
1904
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1905 1906 1907
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1908
	int max = mc->max;
1909 1910
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
F
Frank Mandarino 已提交
1911 1912 1913 1914 1915 1916 1917 1918

	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
Philipp Zabel 已提交
1919
			max - ucontrol->value.integer.value[0];
F
Frank Mandarino 已提交
1920 1921
		if (shift != rshift)
			ucontrol->value.integer.value[1] =
P
Philipp Zabel 已提交
1922
				max - ucontrol->value.integer.value[1];
F
Frank Mandarino 已提交
1923 1924 1925 1926 1927 1928 1929 1930 1931
	}

	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
1932
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1933 1934 1935 1936 1937 1938 1939 1940
 *
 * 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)
{
1941 1942
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
1943
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1944 1945 1946
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1947
	int max = mc->max;
1948 1949
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
1950
	unsigned int val, val2, val_mask;
F
Frank Mandarino 已提交
1951 1952 1953

	val = (ucontrol->value.integer.value[0] & mask);
	if (invert)
P
Philipp Zabel 已提交
1954
		val = max - val;
F
Frank Mandarino 已提交
1955 1956 1957 1958 1959
	val_mask = mask << shift;
	val = val << shift;
	if (shift != rshift) {
		val2 = (ucontrol->value.integer.value[1] & mask);
		if (invert)
P
Philipp Zabel 已提交
1960
			val2 = max - val2;
F
Frank Mandarino 已提交
1961 1962 1963
		val_mask |= mask << rshift;
		val |= val2 << rshift;
	}
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Philipp Zabel 已提交
1964
	return snd_soc_update_bits(codec, reg, val_mask, val);
F
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)
{
1981 1982 1983
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
P
Philipp Zabel 已提交
1984

1985
	if (max == 1 && !strstr(kcontrol->id.name, " Volume"))
P
Philipp Zabel 已提交
1986 1987 1988
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
F
Frank Mandarino 已提交
1989 1990 1991

	uinfo->count = 2;
	uinfo->value.integer.min = 0;
P
Philipp Zabel 已提交
1992
	uinfo->value.integer.max = max;
F
Frank Mandarino 已提交
1993 1994 1995 1996 1997 1998 1999
	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
2000
 * @ucontrol: control element information
F
Frank Mandarino 已提交
2001 2002 2003 2004 2005 2006 2007 2008
 *
 * 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)
{
2009 2010
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
2011
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2012 2013 2014
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
2015
	int max = mc->max;
2016
	unsigned int mask = (1 << fls(max)) - 1;
2017
	unsigned int invert = mc->invert;
F
Frank Mandarino 已提交
2018 2019 2020 2021 2022 2023 2024

	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
2025
			max - ucontrol->value.integer.value[0];
F
Frank Mandarino 已提交
2026
		ucontrol->value.integer.value[1] =
P
Philipp Zabel 已提交
2027
			max - ucontrol->value.integer.value[1];
F
Frank Mandarino 已提交
2028 2029 2030 2031 2032 2033 2034 2035 2036
	}

	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
2037
 * @ucontrol: control element information
F
Frank Mandarino 已提交
2038 2039 2040 2041 2042 2043 2044 2045
 *
 * 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)
{
2046 2047
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
F
Frank Mandarino 已提交
2048
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2049 2050 2051
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
2052
	int max = mc->max;
2053 2054
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
F
Frank Mandarino 已提交
2055
	int err;
2056
	unsigned int val, val2, val_mask;
F
Frank Mandarino 已提交
2057 2058 2059 2060 2061 2062

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

	if (invert) {
P
Philipp Zabel 已提交
2063 2064
		val = max - val;
		val2 = max - val2;
F
Frank Mandarino 已提交
2065 2066 2067 2068 2069
	}

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

2070 2071
	err = snd_soc_update_bits(codec, reg, val_mask, val);
	if (err < 0)
F
Frank Mandarino 已提交
2072 2073 2074 2075 2076 2077 2078
		return err;

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

2079 2080 2081 2082 2083 2084 2085 2086 2087 2088 2089 2090
/**
 * 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)
{
2091 2092 2093 2094
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
	int min = mc->min;
2095 2096 2097 2098 2099 2100 2101 2102 2103 2104 2105 2106

	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
2107
 * @ucontrol: control element information
2108 2109 2110 2111 2112 2113 2114 2115
 *
 * 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)
{
2116 2117
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
2118
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2119
	unsigned int reg = mc->reg;
2120
	int min = mc->min;
2121 2122 2123 2124 2125 2126 2127 2128 2129 2130 2131 2132 2133
	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
2134
 * @ucontrol: control element information
2135 2136 2137 2138 2139 2140 2141 2142
 *
 * 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)
{
2143 2144
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
2145
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2146
	unsigned int reg = mc->reg;
2147
	int min = mc->min;
2148
	unsigned int val;
2149 2150 2151 2152 2153 2154 2155 2156

	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);

2157 2158 2159 2160 2161 2162 2163 2164 2165 2166 2167 2168
/**
 * 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)
{
2169
	if (dai->ops && dai->ops->set_sysclk)
2170
		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
2171 2172 2173 2174 2175 2176 2177 2178
	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
2179
 * @div_id: DAI specific clock divider ID
2180 2181 2182 2183 2184 2185 2186 2187 2188
 * @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)
{
2189
	if (dai->ops && dai->ops->set_clkdiv)
2190
		return dai->ops->set_clkdiv(dai, div_id, div);
2191 2192 2193 2194 2195 2196 2197 2198 2199 2200 2201 2202 2203 2204 2205 2206 2207
	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)
{
2208
	if (dai->ops && dai->ops->set_pll)
2209
		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
2210 2211 2212 2213 2214 2215 2216 2217 2218 2219 2220 2221 2222 2223
	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)
{
2224
	if (dai->ops && dai->ops->set_fmt)
2225
		return dai->ops->set_fmt(dai, fmt);
2226 2227 2228 2229 2230 2231 2232 2233
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);

/**
 * snd_soc_dai_set_tdm_slot - configure DAI TDM.
 * @dai: DAI
2234 2235
 * @tx_mask: bitmask representing active TX slots.
 * @rx_mask: bitmask representing active RX slots.
2236
 * @slots: Number of slots in use.
2237
 * @slot_width: Width in bits for each slot.
2238 2239 2240 2241 2242
 *
 * 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,
2243
	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
2244
{
2245
	if (dai->ops && dai->ops->set_tdm_slot)
2246 2247
		return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask,
				slots, slot_width);
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	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)
{
2262
	if (dai->ops && dai->ops->set_tristate)
2263
		return dai->ops->set_tristate(dai, tristate);
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	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)
{
2278
	if (dai->ops && dai->ops->digital_mute)
2279
		return dai->ops->digital_mute(dai, mute);
2280 2281 2282 2283 2284
	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
 *
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 * @card: Card to register
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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_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);
2304
	snd_soc_instantiate_cards();
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	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;
}

2332 2333 2334
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);

2349 2350 2351
	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);
2356
	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);
2435
	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);

2459 2460 2461 2462 2463 2464 2465 2466 2467 2468 2469 2470 2471 2472 2473 2474 2475 2476 2477 2478 2479 2480 2481 2482 2483 2484 2485 2486 2487 2488 2489 2490 2491
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
 *
M
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 * @codec: codec to register
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 */
int snd_soc_register_codec(struct snd_soc_codec *codec)
{
2499 2500
	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);

2510 2511 2512 2513 2514
	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);

2541
static int __init snd_soc_init(void)
F
Frank Mandarino 已提交
2542
{
2543 2544 2545 2546 2547 2548 2549 2550 2551
#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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2555
static void __exit snd_soc_exit(void)
F
Frank Mandarino 已提交
2556
{
2557 2558 2559
#ifdef CONFIG_DEBUG_FS
	debugfs_remove_recursive(debugfs_root);
#endif
2560
	platform_driver_unregister(&soc_driver);
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}

module_init(snd_soc_init);
module_exit(snd_soc_exit);

/* Module information */
2567
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
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MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
2570
MODULE_ALIAS("platform:soc-audio");