soc-core.c 57.5 KB
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/*
 * soc-core.c  --  ALSA SoC Audio Layer
 *
 * Copyright 2005 Wolfson Microelectronics PLC.
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 * Copyright 2005 Openedhand Ltd.
 *
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 * Author: Liam Girdwood <lrg@slimlogic.co.uk>
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 *         with code, comments and ideas from :-
 *         Richard Purdie <richard@openedhand.com>
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 *
 *  This program is free software; you can redistribute  it and/or modify it
 *  under  the terms of  the GNU General  Public License as published by the
 *  Free Software Foundation;  either version 2 of the  License, or (at your
 *  option) any later version.
 *
 *  TODO:
 *   o Add hw rules to enforce rates, etc.
 *   o More testing with other codecs/machines.
 *   o Add more codecs and platforms to ensure good API coverage.
 *   o Support TDM on PCM and I2S
 */

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

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

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#ifdef CONFIG_DEBUG_FS
static struct dentry *debugfs_root;
#endif

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static DEFINE_MUTEX(client_mutex);
static LIST_HEAD(card_list);
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static LIST_HEAD(dai_list);
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static LIST_HEAD(platform_list);
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static int snd_soc_register_card(struct snd_soc_card *card);
static int snd_soc_unregister_card(struct snd_soc_card *card);

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

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

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

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

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

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

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

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

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

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

	mutex_lock(&pcm_mutex);

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	mutex_lock(&pcm_mutex);

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

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

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

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

	mutex_unlock(&pcm_mutex);
	return 0;
}

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

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

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

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

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

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

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

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

	mutex_lock(&pcm_mutex);

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

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

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

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

out:
	mutex_unlock(&pcm_mutex);
	return ret;

platform_err:
528
	if (cpu_dai->ops.hw_free)
529
		cpu_dai->ops.hw_free(substream, cpu_dai);
F
Frank Mandarino 已提交
530 531

interface_err:
532
	if (codec_dai->ops.hw_free)
533
		codec_dai->ops.hw_free(substream, codec_dai);
534 535

codec_err:
536
	if (machine->ops && machine->ops->hw_free)
537
		machine->ops->hw_free(substream);
F
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538 539 540 541 542 543 544 545 546 547 548 549

	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;
550
	struct snd_soc_dai_link *machine = rtd->dai;
551 552
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
553 554
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
F
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555 556 557 558 559
	struct snd_soc_codec *codec = socdev->codec;

	mutex_lock(&pcm_mutex);

	/* apply codec digital mute */
560 561
	if (!codec->active)
		snd_soc_dai_digital_mute(codec_dai, 1);
F
Frank Mandarino 已提交
562 563 564 565 566 567 568 569 570 571

	/* 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  */
572
	if (codec_dai->ops.hw_free)
573
		codec_dai->ops.hw_free(substream, codec_dai);
F
Frank Mandarino 已提交
574

575
	if (cpu_dai->ops.hw_free)
576
		cpu_dai->ops.hw_free(substream, cpu_dai);
F
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577 578 579 580 581 582 583 584 585

	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;
586
	struct snd_soc_card *card= socdev->card;
587
	struct snd_soc_dai_link *machine = rtd->dai;
588
	struct snd_soc_platform *platform = card->platform;
589 590
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
F
Frank Mandarino 已提交
591 592
	int ret;

593
	if (codec_dai->ops.trigger) {
594
		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
F
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595 596 597 598 599 600 601 602 603 604
		if (ret < 0)
			return ret;
	}

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

605
	if (cpu_dai->ops.trigger) {
606
		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
F
Frank Mandarino 已提交
607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624 625 626
		if (ret < 0)
			return ret;
	}
	return 0;
}

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

#ifdef CONFIG_PM
/* powers down audio subsystem for suspend */
static int soc_suspend(struct platform_device *pdev, pm_message_t state)
{
627
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
628
	struct snd_soc_card *card = socdev->card;
629
	struct snd_soc_platform *platform = card->platform;
630
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
F
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631 632 633
	struct snd_soc_codec *codec = socdev->codec;
	int i;

634 635 636 637 638 639 640 641 642 643
	/* Due to the resume being scheduled into a workqueue we could
	* suspend before that's finished - wait for it to complete.
	 */
	snd_power_lock(codec->card);
	snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
	snd_power_unlock(codec->card);

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

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

651
	/* suspend all pcms */
652 653
	for (i = 0; i < card->num_links; i++)
		snd_pcm_suspend_all(card->dai_link[i].pcm);
654

655 656
	if (card->suspend_pre)
		card->suspend_pre(pdev, state);
F
Frank Mandarino 已提交
657

658 659
	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
660
		if (cpu_dai->suspend && !cpu_dai->ac97_control)
661
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
662
		if (platform->suspend)
663
			platform->suspend(cpu_dai);
F
Frank Mandarino 已提交
664 665 666
	}

	/* close any waiting streams and save state */
667
	run_delayed_work(&card->delayed_work);
668
	codec->suspend_bias_level = codec->bias_level;
F
Frank Mandarino 已提交
669

670
	for (i = 0; i < codec->num_dai; i++) {
F
Frank Mandarino 已提交
671 672 673 674 675 676 677 678 679 680 681 682 683
		char *stream = codec->dai[i].playback.stream_name;
		if (stream != NULL)
			snd_soc_dapm_stream_event(codec, stream,
				SND_SOC_DAPM_STREAM_SUSPEND);
		stream = codec->dai[i].capture.stream_name;
		if (stream != NULL)
			snd_soc_dapm_stream_event(codec, stream,
				SND_SOC_DAPM_STREAM_SUSPEND);
	}

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

684 685
	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
686
		if (cpu_dai->suspend && cpu_dai->ac97_control)
687
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
688 689
	}

690 691
	if (card->suspend_post)
		card->suspend_post(pdev, state);
F
Frank Mandarino 已提交
692 693 694 695

	return 0;
}

696 697 698 699
/* 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
700
{
701 702 703 704
	struct snd_soc_card *card = container_of(work,
						 struct snd_soc_card,
						 deferred_resume_work);
	struct snd_soc_device *socdev = card->socdev;
705
	struct snd_soc_platform *platform = card->platform;
706
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
F
Frank Mandarino 已提交
707
	struct snd_soc_codec *codec = socdev->codec;
708
	struct platform_device *pdev = to_platform_device(socdev->dev);
F
Frank Mandarino 已提交
709 710
	int i;

711 712 713 714
	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
	 * so userspace apps are blocked from touching us
	 */

715
	dev_dbg(socdev->dev, "starting resume work\n");
716

717 718
	if (card->resume_pre)
		card->resume_pre(pdev);
F
Frank Mandarino 已提交
719

720 721
	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
722
		if (cpu_dai->resume && cpu_dai->ac97_control)
723
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
724 725 726 727 728
	}

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

729 730
	for (i = 0; i < codec->num_dai; i++) {
		char *stream = codec->dai[i].playback.stream_name;
F
Frank Mandarino 已提交
731 732 733 734 735 736 737 738 739
		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);
	}

740
	/* unmute any active DACs */
741 742 743 744
	for (i = 0; i < card->num_links; i++) {
		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
		if (dai->ops.digital_mute && dai->playback.active)
			dai->ops.digital_mute(dai, 0);
F
Frank Mandarino 已提交
745 746
	}

747 748
	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
749
		if (cpu_dai->resume && !cpu_dai->ac97_control)
750
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
751
		if (platform->resume)
752
			platform->resume(cpu_dai);
F
Frank Mandarino 已提交
753 754
	}

755 756
	if (card->resume_post)
		card->resume_post(pdev);
F
Frank Mandarino 已提交
757

758
	dev_dbg(socdev->dev, "resume work completed\n");
759 760 761 762 763 764 765 766 767

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

/* powers up audio subsystem after a suspend */
static int soc_resume(struct platform_device *pdev)
{
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
768
	struct snd_soc_card *card = socdev->card;
769

770
	dev_dbg(socdev->dev, "scheduling resume work\n");
771

772
	if (!schedule_work(&card->deferred_resume_work))
773
		dev_err(socdev->dev, "resume work item may be lost\n");
774

F
Frank Mandarino 已提交
775 776 777 778 779 780 781 782
	return 0;
}

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

783
static void snd_soc_instantiate_card(struct snd_soc_card *card)
F
Frank Mandarino 已提交
784
{
785 786 787 788 789 790 791 792 793 794 795 796 797 798 799 800 801 802 803 804 805 806
	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;
	int i, found, ret;

	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;
	}
807

808 809 810 811 812 813 814 815 816 817 818 819
	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;
		}
M
Mark Brown 已提交
820 821
	}

822 823 824 825 826
	/* Note that we do not current check for codec components */

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

	/* Found everything, bring it up */
827 828
	if (card->probe) {
		ret = card->probe(pdev);
829
		if (ret < 0)
830
			return;
F
Frank Mandarino 已提交
831 832
	}

833 834
	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
835
		if (cpu_dai->probe) {
836
			ret = cpu_dai->probe(pdev, cpu_dai);
837
			if (ret < 0)
F
Frank Mandarino 已提交
838 839 840 841 842 843
				goto cpu_dai_err;
		}
	}

	if (codec_dev->probe) {
		ret = codec_dev->probe(pdev);
844
		if (ret < 0)
F
Frank Mandarino 已提交
845 846 847 848 849
			goto cpu_dai_err;
	}

	if (platform->probe) {
		ret = platform->probe(pdev);
850
		if (ret < 0)
F
Frank Mandarino 已提交
851 852 853 854
			goto platform_err;
	}

	/* DAPM stream work */
855
	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
R
Randy Dunlap 已提交
856
#ifdef CONFIG_PM
857
	/* deferred resume work */
858
	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
R
Randy Dunlap 已提交
859
#endif
860

861 862 863
	card->instantiated = 1;

	return;
F
Frank Mandarino 已提交
864 865 866 867 868 869

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

cpu_dai_err:
870
	for (i--; i >= 0; i--) {
871
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
872
		if (cpu_dai->remove)
873
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
874 875
	}

876 877
	if (card->remove)
		card->remove(pdev);
878
}
F
Frank Mandarino 已提交
879

880 881 882 883 884 885 886 887 888 889 890 891 892 893 894 895 896 897 898 899 900 901 902 903 904 905 906 907 908 909
/*
 * 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
910 911 912 913 914 915 916
}

/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
	int i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
917
	struct snd_soc_card *card = socdev->card;
918
	struct snd_soc_platform *platform = card->platform;
F
Frank Mandarino 已提交
919 920
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

921
	run_delayed_work(&card->delayed_work);
922

F
Frank Mandarino 已提交
923 924 925 926 927 928
	if (platform->remove)
		platform->remove(pdev);

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

929 930
	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
931
		if (cpu_dai->remove)
932
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
933 934
	}

935 936
	if (card->remove)
		card->remove(pdev);
F
Frank Mandarino 已提交
937

M
Mark Brown 已提交
938 939
	snd_soc_unregister_card(card);

F
Frank Mandarino 已提交
940 941 942 943 944 945 946
	return 0;
}

/* ASoC platform driver */
static struct platform_driver soc_driver = {
	.driver		= {
		.name		= "soc-audio",
947
		.owner		= THIS_MODULE,
F
Frank Mandarino 已提交
948 949 950 951 952 953 954 955 956 957 958 959
	},
	.probe		= soc_probe,
	.remove		= soc_remove,
	.suspend	= soc_suspend,
	.resume		= soc_resume,
};

/* create a new pcm */
static int soc_new_pcm(struct snd_soc_device *socdev,
	struct snd_soc_dai_link *dai_link, int num)
{
	struct snd_soc_codec *codec = socdev->codec;
960 961
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
962 963
	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
964 965 966 967 968 969 970 971
	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;
972 973

	rtd->dai = dai_link;
F
Frank Mandarino 已提交
974
	rtd->socdev = socdev;
975
	codec_dai->codec = socdev->codec;
F
Frank Mandarino 已提交
976 977

	/* check client and interface hw capabilities */
M
Mark Brown 已提交
978 979
	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
		num);
F
Frank Mandarino 已提交
980 981 982 983 984 985 986 987 988

	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) {
989 990
		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
			codec->name);
F
Frank Mandarino 已提交
991 992 993 994
		kfree(rtd);
		return ret;
	}

995
	dai_link->pcm = pcm;
F
Frank Mandarino 已提交
996
	pcm->private_data = rtd;
997 998 999 1000 1001 1002 1003
	soc_pcm_ops.mmap = platform->pcm_ops->mmap;
	soc_pcm_ops.pointer = platform->pcm_ops->pointer;
	soc_pcm_ops.ioctl = platform->pcm_ops->ioctl;
	soc_pcm_ops.copy = platform->pcm_ops->copy;
	soc_pcm_ops.silence = platform->pcm_ops->silence;
	soc_pcm_ops.ack = platform->pcm_ops->ack;
	soc_pcm_ops.page = platform->pcm_ops->page;
F
Frank Mandarino 已提交
1004 1005 1006 1007 1008 1009 1010

	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);

1011
	ret = platform->pcm_new(codec->card, codec_dai, pcm);
F
Frank Mandarino 已提交
1012 1013 1014 1015 1016 1017
	if (ret < 0) {
		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
		kfree(rtd);
		return ret;
	}

1018
	pcm->private_free = platform->pcm_free;
F
Frank Mandarino 已提交
1019 1020 1021 1022 1023 1024
	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
		cpu_dai->name);
	return ret;
}

/* codec register dump */
1025
static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
F
Frank Mandarino 已提交
1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036
{
	struct snd_soc_codec *codec = devdata->codec;
	int i, step = 1, count = 0;

	if (!codec->reg_cache_size)
		return 0;

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

	count += sprintf(buf, "%s registers\n", codec->name);
1037 1038 1039 1040 1041 1042 1043 1044 1045 1046 1047 1048 1049 1050 1051 1052 1053 1054 1055 1056 1057 1058 1059
	for (i = 0; i < codec->reg_cache_size; i += step) {
		count += sprintf(buf + count, "%2x: ", i);
		if (count >= PAGE_SIZE - 1)
			break;

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

		if (count >= PAGE_SIZE - 1)
			break;

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

	/* Truncate count; min() would cause a warning */
	if (count >= PAGE_SIZE)
		count = PAGE_SIZE - 1;
F
Frank Mandarino 已提交
1060 1061 1062

	return count;
}
1063 1064 1065 1066 1067 1068 1069
static ssize_t codec_reg_show(struct device *dev,
	struct device_attribute *attr, char *buf)
{
	struct snd_soc_device *devdata = dev_get_drvdata(dev);
	return soc_codec_reg_show(devdata, buf);
}

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

1072 1073 1074 1075 1076 1077 1078 1079 1080 1081 1082
#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;
1083 1084 1085
	struct snd_soc_codec *codec = file->private_data;
	struct device *card_dev = codec->card->dev;
	struct snd_soc_device *devdata = card_dev->driver_data;
1086 1087 1088 1089 1090 1091 1092 1093 1094 1095 1096 1097 1098 1099 1100 1101 1102 1103
	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
	if (!buf)
		return -ENOMEM;
	ret = soc_codec_reg_show(devdata, buf);
	if (ret >= 0)
		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
	kfree(buf);
	return ret;
}

static ssize_t codec_reg_write_file(struct file *file,
		const char __user *user_buf, size_t count, loff_t *ppos)
{
	char buf[32];
	int buf_size;
	char *start = buf;
	unsigned long reg, value;
	int step = 1;
1104
	struct snd_soc_codec *codec = file->private_data;
1105 1106 1107 1108 1109 1110 1111 1112 1113 1114 1115 1116 1117 1118 1119 1120 1121 1122 1123 1124 1125 1126 1127 1128 1129 1130 1131 1132

	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,
};

1133
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1134
{
1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146 1147
	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");
1148 1149
}

1150
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1151
{
1152 1153
	debugfs_remove(codec->debugfs_pop_time);
	debugfs_remove(codec->debugfs_reg);
1154 1155 1156 1157
}

#else

1158
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1159 1160 1161
{
}

1162
static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1163 1164 1165 1166
{
}
#endif

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/**
 * snd_soc_new_ac97_codec - initailise AC97 device
 * @codec: audio codec
 * @ops: AC97 bus operations
 * @num: AC97 codec number
 *
 * Initialises AC97 codec resources for use by ad-hoc devices only.
 */
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
	struct snd_ac97_bus_ops *ops, int num)
{
	mutex_lock(&codec->mutex);

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

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

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

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

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

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

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

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

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

	return change;
}
EXPORT_SYMBOL_GPL(snd_soc_test_bits);

/**
 * snd_soc_new_pcms - create new sound card and pcms
 * @socdev: the SoC audio device
 *
 * Create a new sound card based upon the codec and interface pcms.
 *
 * Returns 0 for success, else error.
 */
1282
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
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1283 1284
{
	struct snd_soc_codec *codec = socdev->codec;
1285
	struct snd_soc_card *card = socdev->card;
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1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298 1299 1300 1301 1302 1303
	int ret = 0, i;

	mutex_lock(&codec->mutex);

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

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

	/* create the pcms */
1304 1305
	for (i = 0; i < card->num_links; i++) {
		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
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		if (ret < 0) {
			printk(KERN_ERR "asoc: can't create pcm %s\n",
1308
				card->dai_link[i].stream_name);
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1309 1310 1311 1312 1313 1314 1315 1316 1317 1318 1319
			mutex_unlock(&codec->mutex);
			return ret;
		}
	}

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

/**
1320
 * snd_soc_init_card - register sound card
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 * @socdev: the SoC audio device
 *
 * Register a SoC sound card. Also registers an AC97 device if the
 * codec is AC97 for ad hoc devices.
 *
 * Returns 0 for success, else error.
 */
1328
int snd_soc_init_card(struct snd_soc_device *socdev)
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1329 1330
{
	struct snd_soc_codec *codec = socdev->codec;
1331
	struct snd_soc_card *card = socdev->card;
1332
	int ret = 0, i, ac97 = 0, err = 0;
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1334 1335 1336
	for (i = 0; i < card->num_links; i++) {
		if (card->dai_link[i].init) {
			err = card->dai_link[i].init(codec);
1337 1338
			if (err < 0) {
				printk(KERN_ERR "asoc: failed to init %s\n",
1339
					card->dai_link[i].stream_name);
1340 1341 1342
				continue;
			}
		}
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		if (card->dai_link[i].codec_dai->ac97_control)
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			ac97 = 1;
	}
	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
1347
		 "%s",  card->name);
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	snprintf(codec->card->longname, sizeof(codec->card->longname),
1349
		 "%s (%s)", card->name, codec->name);
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	ret = snd_card_register(codec->card);
	if (ret < 0) {
1353
		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
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				codec->name);
1355
		goto out;
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	}

1358
	mutex_lock(&codec->mutex);
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#ifdef CONFIG_SND_SOC_AC97_BUS
1360 1361 1362 1363 1364
	if (ac97) {
		ret = soc_ac97_dev_register(codec);
		if (ret < 0) {
			printk(KERN_ERR "asoc: AC97 device register failed\n");
			snd_card_free(codec->card);
1365
			mutex_unlock(&codec->mutex);
1366 1367 1368
			goto out;
		}
	}
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#endif

1371 1372 1373 1374 1375 1376
	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)
1377
		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
1378

1379
	soc_init_codec_debugfs(socdev->codec);
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	mutex_unlock(&codec->mutex);
1381 1382

out:
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1383 1384
	return ret;
}
1385
EXPORT_SYMBOL_GPL(snd_soc_init_card);
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/**
 * snd_soc_free_pcms - free sound card and pcms
 * @socdev: the SoC audio device
 *
 * Frees sound card and pcms associated with the socdev.
 * Also unregister the codec if it is an AC97 device.
 */
void snd_soc_free_pcms(struct snd_soc_device *socdev)
{
	struct snd_soc_codec *codec = socdev->codec;
1397
#ifdef CONFIG_SND_SOC_AC97_BUS
1398
	struct snd_soc_dai *codec_dai;
1399 1400
	int i;
#endif
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	mutex_lock(&codec->mutex);
1403
	soc_cleanup_codec_debugfs(socdev->codec);
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#ifdef CONFIG_SND_SOC_AC97_BUS
1405
	for (i = 0; i < codec->num_dai; i++) {
1406
		codec_dai = &codec->dai[i];
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1407
		if (codec_dai->ac97_control && codec->ac97) {
1408 1409 1410 1411 1412
			soc_ac97_dev_unregister(codec);
			goto free_card;
		}
	}
free_card:
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#endif

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

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

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

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

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

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

	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
	uinfo->count = e->shift_l == e->shift_r ? 1 : 2;
1486
	uinfo->value.enumerated.items = e->max;
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1488 1489
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
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1490 1491 1492 1493 1494 1495 1496 1497 1498 1499 1500 1501 1502 1503 1504 1505 1506 1507 1508 1509 1510 1511
	strcpy(uinfo->value.enumerated.name,
		e->texts[uinfo->value.enumerated.item]);
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);

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

1512
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1513 1514
		;
	val = snd_soc_read(codec, e->reg);
1515 1516
	ucontrol->value.enumerated.item[0]
		= (val >> e->shift_l) & (bitmask - 1);
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1517 1518 1519 1520 1521 1522 1523 1524 1525 1526 1527 1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541
	if (e->shift_l != e->shift_r)
		ucontrol->value.enumerated.item[1] =
			(val >> e->shift_r) & (bitmask - 1);

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);

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

1542
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1543
		;
1544
	if (ucontrol->value.enumerated.item[0] > e->max - 1)
F
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1545 1546 1547 1548
		return -EINVAL;
	val = ucontrol->value.enumerated.item[0] << e->shift_l;
	mask = (bitmask - 1) << e->shift_l;
	if (e->shift_l != e->shift_r) {
1549
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
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1550 1551 1552 1553 1554 1555 1556 1557 1558 1559 1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570 1571 1572 1573 1574 1575
			return -EINVAL;
		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
		mask |= (bitmask - 1) << e->shift_r;
	}

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

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

	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
	uinfo->count = 1;
1576
	uinfo->value.enumerated.items = e->max;
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1577

1578 1579
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
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1580 1581 1582 1583 1584 1585 1586 1587 1588 1589 1590 1591 1592 1593 1594 1595 1596 1597
	strcpy(uinfo->value.enumerated.name,
		e->texts[uinfo->value.enumerated.item]);
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext);

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

	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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1604 1605 1606

	uinfo->count = 1;
	uinfo->value.integer.min = 0;
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1607
	uinfo->value.integer.max = max;
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	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext);

/**
 * snd_soc_info_volsw - single mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
1624 1625 1626
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
1627
	unsigned int shift = mc->shift;
1628
	unsigned int rshift = mc->rshift;
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1629

P
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1630 1631 1632 1633 1634
	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;

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1635 1636
	uinfo->count = shift == rshift ? 1 : 2;
	uinfo->value.integer.min = 0;
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1637
	uinfo->value.integer.max = max;
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1638 1639 1640 1641 1642 1643 1644 1645 1646 1647 1648 1649 1650 1651 1652 1653
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);

/**
 * snd_soc_get_volsw - single mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1654 1655
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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1656
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1657 1658 1659
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1660
	int max = mc->max;
1661 1662
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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	ucontrol->value.integer.value[0] =
		(snd_soc_read(codec, reg) >> shift) & mask;
	if (shift != rshift)
		ucontrol->value.integer.value[1] =
			(snd_soc_read(codec, reg) >> rshift) & mask;
	if (invert) {
		ucontrol->value.integer.value[0] =
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			max - ucontrol->value.integer.value[0];
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		if (shift != rshift)
			ucontrol->value.integer.value[1] =
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				max - ucontrol->value.integer.value[1];
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	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw);

/**
 * snd_soc_put_volsw - single mixer put callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1693 1694
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1696 1697 1698
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1699
	int max = mc->max;
1700 1701
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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	unsigned short val, val2, val_mask;

	val = (ucontrol->value.integer.value[0] & mask);
	if (invert)
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		val = max - val;
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1707 1708 1709 1710 1711
	val_mask = mask << shift;
	val = val << shift;
	if (shift != rshift) {
		val2 = (ucontrol->value.integer.value[1] & mask);
		if (invert)
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			val2 = max - val2;
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1713 1714 1715
		val_mask |= mask << rshift;
		val |= val2 << rshift;
	}
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	return snd_soc_update_bits(codec, reg, val_mask, val);
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}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);

/**
 * snd_soc_info_volsw_2r - double mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a double mixer control that
 * spans 2 codec registers.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
1733 1734 1735
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
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1736 1737 1738 1739 1740

	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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	uinfo->count = 2;
	uinfo->value.integer.min = 0;
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	uinfo->value.integer.max = max;
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	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);

/**
 * snd_soc_get_volsw_2r - double mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a double mixer control that spans 2 registers.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1761 1762
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1764 1765 1766
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1767
	int max = mc->max;
1768 1769
	unsigned int mask = (1<<fls(max))-1;
	unsigned int invert = mc->invert;
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	ucontrol->value.integer.value[0] =
		(snd_soc_read(codec, reg) >> shift) & mask;
	ucontrol->value.integer.value[1] =
		(snd_soc_read(codec, reg2) >> shift) & mask;
	if (invert) {
		ucontrol->value.integer.value[0] =
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			max - ucontrol->value.integer.value[0];
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		ucontrol->value.integer.value[1] =
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			max - ucontrol->value.integer.value[1];
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	}

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);

/**
 * snd_soc_put_volsw_2r - double mixer set callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a double mixer control that spans 2 registers.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1798 1799
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1801 1802 1803
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1804
	int max = mc->max;
1805 1806
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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	int err;
	unsigned short val, val2, val_mask;

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

	if (invert) {
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1815 1816
		val = max - val;
		val2 = max - val2;
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	}

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

1822 1823
	err = snd_soc_update_bits(codec, reg, val_mask, val);
	if (err < 0)
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		return err;

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

1831 1832 1833 1834 1835 1836 1837 1838 1839 1840 1841 1842
/**
 * 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)
{
1843 1844 1845 1846
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
	int min = mc->min;
1847 1848 1849 1850 1851 1852 1853 1854 1855 1856 1857 1858 1859 1860 1861 1862 1863 1864 1865 1866 1867

	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
	uinfo->count = 2;
	uinfo->value.integer.min = 0;
	uinfo->value.integer.max = max-min;
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_s8);

/**
 * snd_soc_get_volsw_s8 - signed mixer get callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to get the value of a signed mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1868 1869
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
1870
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1871
	unsigned int reg = mc->reg;
1872
	int min = mc->min;
1873 1874 1875 1876 1877 1878 1879 1880 1881 1882 1883 1884 1885 1886 1887 1888 1889 1890 1891 1892 1893 1894
	int val = snd_soc_read(codec, reg);

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

/**
 * snd_soc_put_volsw_sgn - signed mixer put callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a signed mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1895 1896
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
1897
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1898
	unsigned int reg = mc->reg;
1899
	int min = mc->min;
1900 1901 1902 1903 1904 1905 1906 1907 1908
	unsigned short val;

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

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

1909 1910 1911 1912 1913 1914 1915 1916 1917 1918 1919 1920
/**
 * 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)
{
1921 1922
	if (dai->ops.set_sysclk)
		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
1923 1924 1925 1926 1927 1928 1929 1930 1931 1932 1933 1934 1935 1936 1937 1938 1939 1940
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);

/**
 * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
 * @dai: DAI
 * @clk_id: DAI specific clock divider ID
 * @div: new clock divisor.
 *
 * Configures the clock dividers. This is used to derive the best DAI bit and
 * frame clocks from the system or master clock. It's best to set the DAI bit
 * and frame clocks as low as possible to save system power.
 */
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
	int div_id, int div)
{
1941 1942
	if (dai->ops.set_clkdiv)
		return dai->ops.set_clkdiv(dai, div_id, div);
1943 1944 1945 1946 1947 1948 1949 1950 1951 1952 1953 1954 1955 1956 1957 1958 1959
	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)
{
1960 1961
	if (dai->ops.set_pll)
		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
1962 1963 1964 1965 1966 1967 1968 1969 1970 1971 1972 1973 1974 1975
	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)
{
1976 1977
	if (dai->ops.set_fmt)
		return dai->ops.set_fmt(dai, fmt);
1978 1979 1980 1981 1982 1983 1984 1985 1986 1987 1988 1989 1990 1991 1992 1993 1994
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);

/**
 * snd_soc_dai_set_tdm_slot - configure DAI TDM.
 * @dai: DAI
 * @mask: DAI specific mask representing used slots.
 * @slots: Number of slots in use.
 *
 * Configures a DAI for TDM operation. Both mask and slots are codec and DAI
 * specific.
 */
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
	unsigned int mask, int slots)
{
1995 1996
	if (dai->ops.set_sysclk)
		return dai->ops.set_tdm_slot(dai, mask, slots);
1997 1998 1999 2000 2001 2002 2003 2004 2005 2006 2007 2008 2009 2010
	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)
{
2011 2012
	if (dai->ops.set_sysclk)
		return dai->ops.set_tristate(dai, tristate);
2013 2014 2015 2016 2017 2018 2019 2020 2021 2022 2023 2024 2025 2026
	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)
{
2027 2028
	if (dai->ops.digital_mute)
		return dai->ops.digital_mute(dai, mute);
2029 2030 2031 2032 2033
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);

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2034 2035 2036 2037 2038 2039 2040 2041 2042 2043 2044 2045 2046 2047 2048 2049 2050 2051 2052
/**
 * snd_soc_register_card - Register a card with the ASoC core
 *
 * @param card Card to register
 *
 * Note that currently this is an internal only function: it will be
 * exposed to machine drivers after further backporting of ASoC v2
 * registration APIs.
 */
static int snd_soc_register_card(struct snd_soc_card *card)
{
	if (!card->name || !card->dev)
		return -EINVAL;

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

	mutex_lock(&client_mutex);
	list_add(&card->list, &card_list);
2053
	snd_soc_instantiate_cards();
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Mark Brown 已提交
2054 2055 2056 2057 2058 2059 2060 2061 2062 2063 2064 2065 2066 2067 2068 2069 2070 2071 2072 2073 2074 2075 2076 2077 2078 2079 2080
	mutex_unlock(&client_mutex);

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

	return 0;
}

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

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

	return 0;
}

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2081 2082 2083 2084 2085 2086 2087 2088 2089 2090 2091 2092 2093 2094 2095 2096 2097 2098
/**
 * snd_soc_register_dai - Register a DAI with the ASoC core
 *
 * @param dai DAI to register
 */
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);

	INIT_LIST_HEAD(&dai->list);

	mutex_lock(&client_mutex);
	list_add(&dai->list, &dai_list);
2099
	snd_soc_instantiate_cards();
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2100 2101 2102 2103 2104 2105 2106 2107 2108 2109 2110 2111 2112 2113 2114 2115 2116 2117 2118 2119 2120 2121 2122 2123 2124 2125 2126 2127 2128 2129 2130 2131 2132 2133 2134 2135 2136 2137 2138 2139 2140 2141 2142 2143 2144 2145 2146 2147 2148 2149 2150 2151 2152 2153 2154 2155 2156 2157 2158 2159 2160 2161 2162 2163
	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
 *
 * @param dai DAI to unregister
 */
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
 *
 * @param dai Array of DAIs to register
 * @param count Number of DAIs
 */
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
 *
 * @param dai Array of DAIs to unregister
 * @param count Number of DAIs
 */
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);

M
Mark Brown 已提交
2164 2165 2166 2167 2168 2169 2170 2171 2172 2173 2174 2175 2176 2177
/**
 * snd_soc_register_platform - Register a platform with the ASoC core
 *
 * @param platform platform to register
 */
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);
2178
	snd_soc_instantiate_cards();
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2179 2180 2181 2182 2183 2184 2185 2186 2187 2188 2189 2190 2191 2192 2193 2194 2195 2196 2197 2198 2199 2200 2201
	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
 *
 * @param platform platform to unregister
 */
void snd_soc_unregister_platform(struct snd_soc_platform *platform)
{
	mutex_lock(&client_mutex);
	list_del(&platform->list);
	mutex_unlock(&client_mutex);

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

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Frank Mandarino 已提交
2202 2203
static int __devinit snd_soc_init(void)
{
2204 2205 2206 2207 2208 2209 2210 2211 2212
#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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Frank Mandarino 已提交
2213 2214 2215
	return platform_driver_register(&soc_driver);
}

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Mark Brown 已提交
2216
static void __exit snd_soc_exit(void)
F
Frank Mandarino 已提交
2217
{
2218 2219 2220
#ifdef CONFIG_DEBUG_FS
	debugfs_remove_recursive(debugfs_root);
#endif
2221
	platform_driver_unregister(&soc_driver);
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Frank Mandarino 已提交
2222 2223 2224 2225 2226 2227
}

module_init(snd_soc_init);
module_exit(snd_soc_exit);

/* Module information */
2228
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
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Frank Mandarino 已提交
2229 2230
MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
2231
MODULE_ALIAS("platform:soc-audio");