soc-core.c 51.9 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);

/*
 * 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;

	snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
		 codec->card->number, 0, codec->name);
	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 inline const char *get_dai_name(int type)
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{
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	switch (type) {
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	case SND_SOC_DAI_AC97_BUS:
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	case SND_SOC_DAI_AC97:
		return "AC97";
	case SND_SOC_DAI_I2S:
		return "I2S";
	case SND_SOC_DAI_PCM:
		return "PCM";
	}
	return NULL;
}

/*
 * 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;
	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 = socdev->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);
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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);
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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)
		cpu_dai->ops.shutdown(substream);
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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_device *socdev =
		container_of(work, struct snd_soc_device, delayed_work.work);
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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_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = socdev->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)
		cpu_dai->ops.shutdown(substream);
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	if (codec_dai->ops.shutdown)
		codec_dai->ops.shutdown(substream);
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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(&socdev->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_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = socdev->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) {
		ret = codec_dai->ops.prepare(substream);
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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);
		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;
		cancel_delayed_work(&socdev->delayed_work);
	}
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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_platform *platform = socdev->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);
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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);
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		if (ret < 0) {
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			printk(KERN_ERR "asoc: interface %s hw params failed\n",
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				cpu_dai->name);
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			goto interface_err;
		}
	}

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

out:
	mutex_unlock(&pcm_mutex);
	return ret;

platform_err:
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	if (cpu_dai->ops.hw_free)
		cpu_dai->ops.hw_free(substream);
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interface_err:
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	if (codec_dai->ops.hw_free)
		codec_dai->ops.hw_free(substream);

codec_err:
533
	if (machine->ops && machine->ops->hw_free)
534
		machine->ops->hw_free(substream);
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	mutex_unlock(&pcm_mutex);
	return ret;
}

/*
 * Free's resources allocated by hw_params, can be called multiple times
 */
static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
{
	struct snd_soc_pcm_runtime *rtd = substream->private_data;
	struct snd_soc_device *socdev = rtd->socdev;
547
	struct snd_soc_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = socdev->platform;
549 550
	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);

	/* apply codec digital mute */
556 557
	if (!codec->active)
		snd_soc_dai_digital_mute(codec_dai, 1);
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	/* 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  */
568 569
	if (codec_dai->ops.hw_free)
		codec_dai->ops.hw_free(substream);
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571 572
	if (cpu_dai->ops.hw_free)
		cpu_dai->ops.hw_free(substream);
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	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;
582
	struct snd_soc_dai_link *machine = rtd->dai;
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	struct snd_soc_platform *platform = socdev->platform;
584 585
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
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	int ret;

588 589
	if (codec_dai->ops.trigger) {
		ret = codec_dai->ops.trigger(substream, cmd);
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		if (ret < 0)
			return ret;
	}

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

600 601
	if (cpu_dai->ops.trigger) {
		ret = cpu_dai->ops.trigger(substream, cmd);
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		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)
{
622 623 624 625
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_machine *machine = socdev->machine;
	struct snd_soc_platform *platform = socdev->platform;
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
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	struct snd_soc_codec *codec = socdev->codec;
	int i;

629 630 631 632 633 634 635 636 637 638
	/* 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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	/* mute any active DAC's */
640
	for (i = 0; i < machine->num_links; i++) {
641
		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
642 643
		if (dai->dai_ops.digital_mute && dai->playback.active)
			dai->dai_ops.digital_mute(dai, 1);
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	}

646 647 648 649
	/* suspend all pcms */
	for (i = 0; i < machine->num_links; i++)
		snd_pcm_suspend_all(machine->dai_link[i].pcm);

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	if (machine->suspend_pre)
		machine->suspend_pre(pdev, state);

653
	for (i = 0; i < machine->num_links; i++) {
654
		struct snd_soc_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
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		if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97)
			cpu_dai->suspend(pdev, cpu_dai);
		if (platform->suspend)
			platform->suspend(pdev, cpu_dai);
	}

	/* close any waiting streams and save state */
662
	run_delayed_work(&socdev->delayed_work);
663
	codec->suspend_bias_level = codec->bias_level;
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665
	for (i = 0; i < codec->num_dai; i++) {
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		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);

679
	for (i = 0; i < machine->num_links; i++) {
680
		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
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		if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97)
			cpu_dai->suspend(pdev, cpu_dai);
	}

	if (machine->suspend_post)
		machine->suspend_post(pdev, state);

	return 0;
}

691 692 693 694
/* 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)
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{
696 697 698
	struct snd_soc_device *socdev = container_of(work,
						     struct snd_soc_device,
						     deferred_resume_work);
699 700 701
	struct snd_soc_machine *machine = socdev->machine;
	struct snd_soc_platform *platform = socdev->platform;
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
F
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	struct snd_soc_codec *codec = socdev->codec;
703
	struct platform_device *pdev = to_platform_device(socdev->dev);
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	int i;

706 707 708 709 710 711
	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
	 * so userspace apps are blocked from touching us
	 */

	dev_info(socdev->dev, "starting resume work\n");

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	if (machine->resume_pre)
		machine->resume_pre(pdev);

715
	for (i = 0; i < machine->num_links; i++) {
716
		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
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		if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97)
			cpu_dai->resume(pdev, cpu_dai);
	}

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

724 725
	for (i = 0; i < codec->num_dai; i++) {
		char *stream = codec->dai[i].playback.stream_name;
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		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);
	}

735 736
	/* unmute any active DACs */
	for (i = 0; i < machine->num_links; i++) {
737
		struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
738 739
		if (dai->dai_ops.digital_mute && dai->playback.active)
			dai->dai_ops.digital_mute(dai, 0);
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	}

742
	for (i = 0; i < machine->num_links; i++) {
743
		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
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		if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97)
			cpu_dai->resume(pdev, cpu_dai);
		if (platform->resume)
			platform->resume(pdev, cpu_dai);
	}

	if (machine->resume_post)
		machine->resume_post(pdev);

753 754 755 756 757 758 759 760 761 762 763 764 765 766 767 768
	dev_info(socdev->dev, "resume work completed\n");

	/* 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);

	dev_info(socdev->dev, "scheduling resume work\n");

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

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	return 0;
}

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

/* probes a new socdev */
static int soc_probe(struct platform_device *pdev)
{
	int ret = 0, i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_machine *machine = socdev->machine;
	struct snd_soc_platform *platform = socdev->platform;
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

	if (machine->probe) {
		ret = machine->probe(pdev);
788
		if (ret < 0)
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			return ret;
	}

	for (i = 0; i < machine->num_links; i++) {
793
		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
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		if (cpu_dai->probe) {
795
			ret = cpu_dai->probe(pdev, cpu_dai);
796
			if (ret < 0)
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				goto cpu_dai_err;
		}
	}

	if (codec_dev->probe) {
		ret = codec_dev->probe(pdev);
803
		if (ret < 0)
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			goto cpu_dai_err;
	}

	if (platform->probe) {
		ret = platform->probe(pdev);
809
		if (ret < 0)
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			goto platform_err;
	}

	/* DAPM stream work */
814
	INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work);
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815
#ifdef CONFIG_PM
816 817
	/* deferred resume work */
	INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred);
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#endif
819

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	return 0;

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

cpu_dai_err:
827
	for (i--; i >= 0; i--) {
828
		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
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		if (cpu_dai->remove)
830
			cpu_dai->remove(pdev, cpu_dai);
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	}

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

	return ret;
}

/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
	int i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
	struct snd_soc_machine *machine = socdev->machine;
	struct snd_soc_platform *platform = socdev->platform;
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

848 849
	run_delayed_work(&socdev->delayed_work);

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

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

	for (i = 0; i < machine->num_links; i++) {
857
		struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
858
		if (cpu_dai->remove)
859
			cpu_dai->remove(pdev, cpu_dai);
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	}

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

	return 0;
}

/* ASoC platform driver */
static struct platform_driver soc_driver = {
	.driver		= {
		.name		= "soc-audio",
872
		.owner		= THIS_MODULE,
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	},
	.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;
885 886
	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
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	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;
895 896

	rtd->dai = dai_link;
F
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897
	rtd->socdev = socdev;
898
	codec_dai->codec = socdev->codec;
F
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899 900

	/* check client and interface hw capabilities */
901
	sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
F
Frank Mandarino 已提交
902 903 904 905 906 907 908 909 910 911
		get_dai_name(cpu_dai->type), num);

	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) {
912 913
		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
			codec->name);
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Frank Mandarino 已提交
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		kfree(rtd);
		return ret;
	}

918
	dai_link->pcm = pcm;
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	pcm->private_data = rtd;
	soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
	soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
	soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl;
	soc_pcm_ops.copy = socdev->platform->pcm_ops->copy;
	soc_pcm_ops.silence = socdev->platform->pcm_ops->silence;
	soc_pcm_ops.ack = socdev->platform->pcm_ops->ack;
	soc_pcm_ops.page = socdev->platform->pcm_ops->page;

	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);

	ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm);
	if (ret < 0) {
		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
		kfree(rtd);
		return ret;
	}

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

/* codec register dump */
948
static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
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Frank Mandarino 已提交
949 950 951 952 953 954 955 956 957 958 959
{
	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);
960 961 962 963 964 965 966 967 968 969 970 971 972 973 974 975 976 977 978 979 980 981 982
	for (i = 0; i < codec->reg_cache_size; i += step) {
		count += sprintf(buf + count, "%2x: ", i);
		if (count >= PAGE_SIZE - 1)
			break;

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

		if (count >= PAGE_SIZE - 1)
			break;

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

	/* Truncate count; min() would cause a warning */
	if (count >= PAGE_SIZE)
		count = PAGE_SIZE - 1;
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	return count;
}
986 987 988 989 990 991 992
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);

995 996 997 998 999 1000 1001 1002 1003 1004 1005 1006 1007 1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019 1020 1021 1022 1023 1024 1025 1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045 1046 1047 1048 1049 1050 1051 1052 1053 1054 1055 1056 1057 1058 1059 1060 1061 1062 1063 1064 1065 1066 1067 1068 1069 1070 1071 1072 1073 1074 1075 1076 1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088 1089 1090 1091 1092 1093 1094 1095 1096
#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;
	struct snd_soc_device *devdata = file->private_data;
	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;
	struct snd_soc_device *devdata = file->private_data;
	struct snd_soc_codec *codec = devdata->codec;

	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,
};

static void soc_init_debugfs(struct snd_soc_device *socdev)
{
	struct dentry *root, *file;
	struct snd_soc_codec *codec = socdev->codec;
	root = debugfs_create_dir(dev_name(socdev->dev), NULL);
	if (IS_ERR(root) || !root)
		goto exit1;

	file = debugfs_create_file("codec_reg", 0644,
			root, socdev, &codec_reg_fops);
	if (!file)
		goto exit2;

	file = debugfs_create_u32("dapm_pop_time", 0744,
			root, &codec->pop_time);
	if (!file)
		goto exit2;
	socdev->debugfs_root = root;
	return;
exit2:
	debugfs_remove_recursive(root);
exit1:
	dev_err(socdev->dev, "debugfs is not available\n");
}

static void soc_cleanup_debugfs(struct snd_soc_device *socdev)
{
	debugfs_remove_recursive(socdev->debugfs_root);
	socdev->debugfs_root = NULL;
}

#else

static inline void soc_init_debugfs(struct snd_soc_device *socdev)
{
}

static inline void soc_cleanup_debugfs(struct snd_soc_device *socdev)
{
}
#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.
 */
1212
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
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{
	struct snd_soc_codec *codec = socdev->codec;
	struct snd_soc_machine *machine = socdev->machine;
	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 */
1234
	for (i = 0; i < machine->num_links; i++) {
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		ret = soc_new_pcm(socdev, &machine->dai_link[i], i);
		if (ret < 0) {
			printk(KERN_ERR "asoc: can't create pcm %s\n",
				machine->dai_link[i].stream_name);
			mutex_unlock(&codec->mutex);
			return ret;
		}
	}

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

/**
 * snd_soc_register_card - register sound card
 * @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.
 */
int snd_soc_register_card(struct snd_soc_device *socdev)
{
	struct snd_soc_codec *codec = socdev->codec;
	struct snd_soc_machine *machine = socdev->machine;
1262
	int ret = 0, i, ac97 = 0, err = 0;
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1264
	for (i = 0; i < machine->num_links; i++) {
1265 1266 1267 1268 1269 1270 1271 1272
		if (socdev->machine->dai_link[i].init) {
			err = socdev->machine->dai_link[i].init(codec);
			if (err < 0) {
				printk(KERN_ERR "asoc: failed to init %s\n",
					socdev->machine->dai_link[i].stream_name);
				continue;
			}
		}
1273
		if (socdev->machine->dai_link[i].codec_dai->type ==
1274
			SND_SOC_DAI_AC97_BUS)
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			ac97 = 1;
	}
	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
		 "%s", machine->name);
	snprintf(codec->card->longname, sizeof(codec->card->longname),
		 "%s (%s)", machine->name, codec->name);

	ret = snd_card_register(codec->card);
	if (ret < 0) {
1284
		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
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				codec->name);
1286
		goto out;
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	}

1289
	mutex_lock(&codec->mutex);
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#ifdef CONFIG_SND_SOC_AC97_BUS
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	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);
1296
			mutex_unlock(&codec->mutex);
1297 1298 1299
			goto out;
		}
	}
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#endif

1302 1303 1304 1305 1306 1307
	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)
1308
		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
1309

1310
	soc_init_debugfs(socdev);
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	mutex_unlock(&codec->mutex);
1312 1313

out:
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	return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_register_card);

/**
 * 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;
1328
#ifdef CONFIG_SND_SOC_AC97_BUS
1329
	struct snd_soc_dai *codec_dai;
1330 1331
	int i;
#endif
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	mutex_lock(&codec->mutex);
1334
	soc_cleanup_debugfs(socdev);
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#ifdef CONFIG_SND_SOC_AC97_BUS
1336
	for (i = 0; i < codec->num_dai; i++) {
1337 1338 1339 1340 1341 1342 1343
		codec_dai = &codec->dai[i];
		if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) {
			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;
1417
	uinfo->value.enumerated.items = e->max;
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1419 1420
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
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	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;

1443
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
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		;
	val = snd_soc_read(codec, e->reg);
1446 1447
	ucontrol->value.enumerated.item[0]
		= (val >> e->shift_l) & (bitmask - 1);
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	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;

1473
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
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		;
1475
	if (ucontrol->value.enumerated.item[0] > e->max - 1)
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		return -EINVAL;
	val = ucontrol->value.enumerated.item[0] << e->shift_l;
	mask = (bitmask - 1) << e->shift_l;
	if (e->shift_l != e->shift_r) {
1480
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
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			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;
1507
	uinfo->value.enumerated.items = e->max;
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1509 1510
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
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	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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	uinfo->count = 1;
	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_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)
{
1555 1556 1557
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
1558
	unsigned int shift = mc->shift;
1559
	unsigned int rshift = mc->rshift;
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	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;

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	uinfo->count = shift == rshift ? 1 : 2;
	uinfo->value.integer.min = 0;
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	uinfo->value.integer.max = max;
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	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw);

/**
 * snd_soc_get_volsw - single mixer get callback
 * @kcontrol: mixer control
 * @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)
{
1585 1586
	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);
1588 1589 1590
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1591
	int max = mc->max;
1592 1593
	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)
{
1624 1625
	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);
1627 1628 1629
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1630
	int max = mc->max;
1631 1632
	unsigned int mask = (1 << fls(max)) - 1;
	unsigned int invert = mc->invert;
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	unsigned short val, val2, val_mask;

	val = (ucontrol->value.integer.value[0] & mask);
	if (invert)
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		val = max - val;
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	val_mask = mask << shift;
	val = val << shift;
	if (shift != rshift) {
		val2 = (ucontrol->value.integer.value[1] & mask);
		if (invert)
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1643
			val2 = max - val2;
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		val_mask |= mask << rshift;
		val |= val2 << rshift;
	}
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	return snd_soc_update_bits(codec, reg, val_mask, val);
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}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);

/**
 * snd_soc_info_volsw_2r - double mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a double mixer control that
 * spans 2 codec registers.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
1664 1665 1666
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
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	if (max == 1)
		uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
	else
		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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	uinfo->count = 2;
	uinfo->value.integer.min = 0;
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	uinfo->value.integer.max = max;
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	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r);

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

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r);

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

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

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

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

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

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

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

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

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

/**
 * snd_soc_put_volsw_sgn - signed mixer put callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to set the value of a signed mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
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	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
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	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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	unsigned int reg = mc->reg;
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	int min = mc->min;
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	unsigned short val;

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

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

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

/**
 * snd_soc_dai_set_fmt - configure DAI hardware audio format.
 * @dai: DAI
 * @clk_id: DAI specific clock ID
 * @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)
{
	if (dai->dai_ops.set_fmt)
		return dai->dai_ops.set_fmt(dai, fmt);
	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)
{
	if (dai->dai_ops.set_sysclk)
		return dai->dai_ops.set_tdm_slot(dai, mask, slots);
	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)
{
	if (dai->dai_ops.set_sysclk)
		return dai->dai_ops.set_tristate(dai, tristate);
	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)
{
	if (dai->dai_ops.digital_mute)
		return dai->dai_ops.digital_mute(dai, mute);
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);

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static int __devinit snd_soc_init(void)
{
	return platform_driver_register(&soc_driver);
}

static void snd_soc_exit(void)
{
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	platform_driver_unregister(&soc_driver);
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}

module_init(snd_soc_init);
module_exit(snd_soc_exit);

/* Module information */
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MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
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MODULE_DESCRIPTION("ALSA SoC Core");
MODULE_LICENSE("GPL");
1983
MODULE_ALIAS("platform:soc-audio");