soc-core.c 62.1 KB
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/*
 * soc-core.c  --  ALSA SoC Audio Layer
 *
 * Copyright 2005 Wolfson Microelectronics PLC.
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 * Copyright 2005 Openedhand Ltd.
 *
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 * Author: Liam Girdwood <lrg@slimlogic.co.uk>
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 *         with code, comments and ideas from :-
 *         Richard Purdie <richard@openedhand.com>
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 *
 *  This program is free software; you can redistribute  it and/or modify it
 *  under  the terms of  the GNU General  Public License as published by the
 *  Free Software Foundation;  either version 2 of the  License, or (at your
 *  option) any later version.
 *
 *  TODO:
 *   o Add hw rules to enforce rates, etc.
 *   o More testing with other codecs/machines.
 *   o Add more codecs and platforms to ensure good API coverage.
 *   o Support TDM on PCM and I2S
 */

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

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

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#ifdef CONFIG_DEBUG_FS
static struct dentry *debugfs_root;
#endif

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

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

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

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

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

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

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

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

	codec->ac97->dev.bus = &ac97_bus_type;
	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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	card->codec->active++;
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	mutex_unlock(&pcm_mutex);
	return 0;

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

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

platform_err:
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	if (cpu_dai->ops.shutdown)
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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 = card->codec;
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	struct snd_soc_dai *codec_dai;
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	int i;

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

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

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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 = card->codec;
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	mutex_lock(&pcm_mutex);

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

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

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	if (cpu_dai->ops.shutdown)
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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 = card->codec;
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	int ret = 0;

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

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

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	if (codec_dai->ops.prepare) {
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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:
529
	if (cpu_dai->ops.hw_free)
530
		cpu_dai->ops.hw_free(substream, cpu_dai);
F
Frank Mandarino 已提交
531 532

interface_err:
533
	if (codec_dai->ops.hw_free)
534
		codec_dai->ops.hw_free(substream, codec_dai);
535 536

codec_err:
537
	if (machine->ops && machine->ops->hw_free)
538
		machine->ops->hw_free(substream);
F
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539 540 541 542 543 544 545 546 547 548 549 550

	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;
551
	struct snd_soc_dai_link *machine = rtd->dai;
552 553
	struct snd_soc_card *card = socdev->card;
	struct snd_soc_platform *platform = card->platform;
554 555
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
556
	struct snd_soc_codec *codec = card->codec;
F
Frank Mandarino 已提交
557 558 559 560

	mutex_lock(&pcm_mutex);

	/* apply codec digital mute */
561 562
	if (!codec->active)
		snd_soc_dai_digital_mute(codec_dai, 1);
F
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563 564 565 566 567 568 569 570 571 572

	/* 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  */
573
	if (codec_dai->ops.hw_free)
574
		codec_dai->ops.hw_free(substream, codec_dai);
F
Frank Mandarino 已提交
575

576
	if (cpu_dai->ops.hw_free)
577
		cpu_dai->ops.hw_free(substream, cpu_dai);
F
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578 579 580 581 582 583 584 585 586

	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;
587
	struct snd_soc_card *card= socdev->card;
588
	struct snd_soc_dai_link *machine = rtd->dai;
589
	struct snd_soc_platform *platform = card->platform;
590 591
	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
	struct snd_soc_dai *codec_dai = machine->codec_dai;
F
Frank Mandarino 已提交
592 593
	int ret;

594
	if (codec_dai->ops.trigger) {
595
		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
F
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596 597 598 599 600 601 602 603 604 605
		if (ret < 0)
			return ret;
	}

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

606
	if (cpu_dai->ops.trigger) {
607
		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
F
Frank Mandarino 已提交
608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624 625 626 627
		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)
{
628
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
629
	struct snd_soc_card *card = socdev->card;
630
	struct snd_soc_platform *platform = card->platform;
631
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
632
	struct snd_soc_codec *codec = card->codec;
F
Frank Mandarino 已提交
633 634
	int i;

635 636 637 638 639 640 641 642 643 644
	/* Due to the resume being scheduled into a workqueue we could
	* suspend before that's finished - wait for it to complete.
	 */
	snd_power_lock(codec->card);
	snd_power_wait(codec->card, SNDRV_CTL_POWER_D0);
	snd_power_unlock(codec->card);

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

F
Frank Mandarino 已提交
645
	/* mute any active DAC's */
646 647 648 649
	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
650 651
	}

652
	/* suspend all pcms */
653 654
	for (i = 0; i < card->num_links; i++)
		snd_pcm_suspend_all(card->dai_link[i].pcm);
655

656 657
	if (card->suspend_pre)
		card->suspend_pre(pdev, state);
F
Frank Mandarino 已提交
658

659 660
	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
661
		if (cpu_dai->suspend && !cpu_dai->ac97_control)
662
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
663
		if (platform->suspend)
664
			platform->suspend(cpu_dai);
F
Frank Mandarino 已提交
665 666 667
	}

	/* close any waiting streams and save state */
668
	run_delayed_work(&card->delayed_work);
669
	codec->suspend_bias_level = codec->bias_level;
F
Frank Mandarino 已提交
670

671
	for (i = 0; i < codec->num_dai; i++) {
F
Frank Mandarino 已提交
672 673 674 675 676 677 678 679 680 681 682 683 684
		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);

685 686
	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
687
		if (cpu_dai->suspend && cpu_dai->ac97_control)
688
			cpu_dai->suspend(cpu_dai);
F
Frank Mandarino 已提交
689 690
	}

691 692
	if (card->suspend_post)
		card->suspend_post(pdev, state);
F
Frank Mandarino 已提交
693 694 695 696

	return 0;
}

697 698 699 700
/* 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
701
{
702 703 704 705
	struct snd_soc_card *card = container_of(work,
						 struct snd_soc_card,
						 deferred_resume_work);
	struct snd_soc_device *socdev = card->socdev;
706
	struct snd_soc_platform *platform = card->platform;
707
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
708
	struct snd_soc_codec *codec = card->codec;
709
	struct platform_device *pdev = to_platform_device(socdev->dev);
F
Frank Mandarino 已提交
710 711
	int i;

712 713 714 715
	/* our power state is still SNDRV_CTL_POWER_D3hot from suspend time,
	 * so userspace apps are blocked from touching us
	 */

716
	dev_dbg(socdev->dev, "starting resume work\n");
717

718 719
	if (card->resume_pre)
		card->resume_pre(pdev);
F
Frank Mandarino 已提交
720

721 722
	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
723
		if (cpu_dai->resume && cpu_dai->ac97_control)
724
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
725 726 727 728 729
	}

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

730 731
	for (i = 0; i < codec->num_dai; i++) {
		char *stream = codec->dai[i].playback.stream_name;
F
Frank Mandarino 已提交
732 733 734 735 736 737 738 739 740
		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);
	}

741
	/* unmute any active DACs */
742 743 744 745
	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
746 747
	}

748 749
	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
750
		if (cpu_dai->resume && !cpu_dai->ac97_control)
751
			cpu_dai->resume(cpu_dai);
F
Frank Mandarino 已提交
752
		if (platform->resume)
753
			platform->resume(cpu_dai);
F
Frank Mandarino 已提交
754 755
	}

756 757
	if (card->resume_post)
		card->resume_post(pdev);
F
Frank Mandarino 已提交
758

759
	dev_dbg(socdev->dev, "resume work completed\n");
760 761 762 763 764 765 766 767 768

	/* 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);
769
	struct snd_soc_card *card = socdev->card;
770

771
	dev_dbg(socdev->dev, "scheduling resume work\n");
772

773
	if (!schedule_work(&card->deferred_resume_work))
774
		dev_err(socdev->dev, "resume work item may be lost\n");
775

F
Frank Mandarino 已提交
776 777 778 779 780 781 782 783
	return 0;
}

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

784
static void snd_soc_instantiate_card(struct snd_soc_card *card)
F
Frank Mandarino 已提交
785
{
786 787 788 789 790 791
	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;
792
	int i, found, ret, ac97;
793 794 795 796 797 798 799 800 801 802 803 804 805 806 807

	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;
	}
808

809
	ac97 = 0;
810 811 812 813 814 815 816 817 818 819 820 821
	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;
		}
822 823 824

		if (card->dai_link[i].cpu_dai->ac97_control)
			ac97 = 1;
M
Mark Brown 已提交
825 826
	}

827 828 829 830 831 832 833 834 835 836 837 838 839 840 841 842 843 844 845 846 847
	/* If we have AC97 in the system then don't wait for the
	 * codec.  This will need revisiting if we have to handle
	 * systems with mixed AC97 and non-AC97 parts.  Only check for
	 * DAIs currently; we can't do this per link since some AC97
	 * codecs have non-AC97 DAIs.
	 */
	if (!ac97)
		for (i = 0; i < card->num_links; i++) {
			found = 0;
			list_for_each_entry(dai, &dai_list, list)
				if (card->dai_link[i].codec_dai == dai) {
					found = 1;
					break;
				}
			if (!found) {
				dev_dbg(card->dev, "DAI %s not registered\n",
					card->dai_link[i].codec_dai->name);
				return;
			}
		}

848 849 850 851 852
	/* Note that we do not current check for codec components */

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

	/* Found everything, bring it up */
853 854
	if (card->probe) {
		ret = card->probe(pdev);
855
		if (ret < 0)
856
			return;
F
Frank Mandarino 已提交
857 858
	}

859 860
	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
861
		if (cpu_dai->probe) {
862
			ret = cpu_dai->probe(pdev, cpu_dai);
863
			if (ret < 0)
F
Frank Mandarino 已提交
864 865 866 867 868 869
				goto cpu_dai_err;
		}
	}

	if (codec_dev->probe) {
		ret = codec_dev->probe(pdev);
870
		if (ret < 0)
F
Frank Mandarino 已提交
871 872 873 874 875
			goto cpu_dai_err;
	}

	if (platform->probe) {
		ret = platform->probe(pdev);
876
		if (ret < 0)
F
Frank Mandarino 已提交
877 878 879 880
			goto platform_err;
	}

	/* DAPM stream work */
881
	INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work);
R
Randy Dunlap 已提交
882
#ifdef CONFIG_PM
883
	/* deferred resume work */
884
	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
R
Randy Dunlap 已提交
885
#endif
886

887 888 889
	card->instantiated = 1;

	return;
F
Frank Mandarino 已提交
890 891 892 893 894 895

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

cpu_dai_err:
896
	for (i--; i >= 0; i--) {
897
		struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai;
F
Frank Mandarino 已提交
898
		if (cpu_dai->remove)
899
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
900 901
	}

902 903
	if (card->remove)
		card->remove(pdev);
904
}
F
Frank Mandarino 已提交
905

906 907 908 909 910 911 912 913 914 915 916 917 918 919 920 921 922 923 924 925 926 927 928 929 930 931 932 933 934 935
/*
 * 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
936 937 938 939 940 941 942
}

/* removes a socdev */
static int soc_remove(struct platform_device *pdev)
{
	int i;
	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
943
	struct snd_soc_card *card = socdev->card;
944
	struct snd_soc_platform *platform = card->platform;
F
Frank Mandarino 已提交
945 946
	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;

947
	run_delayed_work(&card->delayed_work);
948

F
Frank Mandarino 已提交
949 950 951 952 953 954
	if (platform->remove)
		platform->remove(pdev);

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

955 956
	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
957
		if (cpu_dai->remove)
958
			cpu_dai->remove(pdev, cpu_dai);
F
Frank Mandarino 已提交
959 960
	}

961 962
	if (card->remove)
		card->remove(pdev);
F
Frank Mandarino 已提交
963

M
Mark Brown 已提交
964 965
	snd_soc_unregister_card(card);

F
Frank Mandarino 已提交
966 967 968 969 970 971 972
	return 0;
}

/* ASoC platform driver */
static struct platform_driver soc_driver = {
	.driver		= {
		.name		= "soc-audio",
973
		.owner		= THIS_MODULE,
F
Frank Mandarino 已提交
974 975 976 977 978 979 980 981 982 983 984
	},
	.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)
{
985
	struct snd_soc_card *card = socdev->card;
986
	struct snd_soc_codec *codec = card->codec;
987
	struct snd_soc_platform *platform = card->platform;
988 989
	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
990 991 992 993 994 995 996 997
	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;
998 999

	rtd->dai = dai_link;
F
Frank Mandarino 已提交
1000
	rtd->socdev = socdev;
1001
	codec_dai->codec = card->codec;
F
Frank Mandarino 已提交
1002 1003

	/* check client and interface hw capabilities */
M
Mark Brown 已提交
1004 1005
	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
		num);
F
Frank Mandarino 已提交
1006 1007 1008 1009 1010 1011 1012 1013 1014

	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) {
1015 1016
		printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
			codec->name);
F
Frank Mandarino 已提交
1017 1018 1019 1020
		kfree(rtd);
		return ret;
	}

1021
	dai_link->pcm = pcm;
F
Frank Mandarino 已提交
1022
	pcm->private_data = rtd;
1023 1024 1025 1026 1027 1028 1029
	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
1030 1031 1032 1033 1034 1035 1036

	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);

1037
	ret = platform->pcm_new(codec->card, codec_dai, pcm);
F
Frank Mandarino 已提交
1038 1039 1040 1041 1042 1043
	if (ret < 0) {
		printk(KERN_ERR "asoc: platform pcm constructor failed\n");
		kfree(rtd);
		return ret;
	}

1044
	pcm->private_free = platform->pcm_free;
F
Frank Mandarino 已提交
1045 1046 1047 1048 1049 1050
	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
		cpu_dai->name);
	return ret;
}

/* codec register dump */
1051
static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
F
Frank Mandarino 已提交
1052 1053 1054 1055 1056 1057 1058 1059 1060 1061
{
	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);
1062 1063 1064 1065 1066 1067 1068 1069 1070 1071 1072 1073 1074 1075 1076 1077 1078 1079 1080 1081 1082 1083 1084
	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;
}
1088 1089 1090 1091
static ssize_t codec_reg_show(struct device *dev,
	struct device_attribute *attr, char *buf)
{
	struct snd_soc_device *devdata = dev_get_drvdata(dev);
1092
	return soc_codec_reg_show(devdata->card->codec, buf);
1093 1094
}

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

1097 1098 1099 1100 1101 1102 1103 1104 1105 1106 1107
#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;
1108
	struct snd_soc_codec *codec = file->private_data;
1109 1110 1111
	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
	if (!buf)
		return -ENOMEM;
1112
	ret = soc_codec_reg_show(codec, buf);
1113 1114 1115 1116 1117 1118 1119 1120 1121 1122 1123 1124 1125 1126
	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;
1127
	struct snd_soc_codec *codec = file->private_data;
1128 1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146 1147 1148 1149 1150 1151 1152 1153 1154 1155

	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,
};

1156
static void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1157
{
1158 1159 1160 1161 1162 1163 1164 1165 1166 1167 1168 1169 1170
	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");
1171 1172
}

1173
static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1174
{
1175 1176
	debugfs_remove(codec->debugfs_pop_time);
	debugfs_remove(codec->debugfs_reg);
1177 1178 1179 1180
}

#else

1181
static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec)
1182 1183 1184
{
}

1185
static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec)
1186 1187 1188 1189
{
}
#endif

F
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1190 1191 1192 1193 1194 1195 1196 1197 1198 1199 1200 1201 1202 1203 1204 1205 1206 1207 1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219 1220 1221 1222 1223 1224 1225 1226 1227 1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243 1244 1245 1246 1247 1248 1249 1250 1251 1252 1253 1254 1255 1256 1257 1258 1259 1260 1261 1262 1263 1264 1265 1266 1267 1268 1269 1270 1271 1272 1273 1274 1275 1276 1277 1278 1279 1280 1281 1282 1283 1284 1285 1286 1287 1288 1289 1290 1291 1292 1293 1294 1295 1296 1297 1298 1299
/**
 * snd_soc_new_ac97_codec - initailise AC97 device
 * @codec: audio codec
 * @ops: AC97 bus operations
 * @num: AC97 codec number
 *
 * Initialises AC97 codec resources for use by ad-hoc devices only.
 */
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
	struct snd_ac97_bus_ops *ops, int num)
{
	mutex_lock(&codec->mutex);

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

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

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

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

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

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

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

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

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

	return change;
}
EXPORT_SYMBOL_GPL(snd_soc_test_bits);

/**
 * snd_soc_new_pcms - create new sound card and pcms
 * @socdev: the SoC audio device
M
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1300 1301
 * @idx: ALSA card index
 * @xid: card identification
F
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1302 1303 1304 1305 1306
 *
 * Create a new sound card based upon the codec and interface pcms.
 *
 * Returns 0 for success, else error.
 */
1307
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
F
Frank Mandarino 已提交
1308
{
1309
	struct snd_soc_card *card = socdev->card;
1310
	struct snd_soc_codec *codec = card->codec;
1311
	int ret, i;
F
Frank Mandarino 已提交
1312 1313 1314 1315

	mutex_lock(&codec->mutex);

	/* register a sound card */
1316 1317
	ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
	if (ret < 0) {
F
Frank Mandarino 已提交
1318 1319 1320
		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
			codec->name);
		mutex_unlock(&codec->mutex);
1321
		return ret;
F
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1322 1323 1324 1325 1326 1327 1328
	}

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

	/* create the pcms */
1329 1330
	for (i = 0; i < card->num_links; i++) {
		ret = soc_new_pcm(socdev, &card->dai_link[i], i);
F
Frank Mandarino 已提交
1331 1332
		if (ret < 0) {
			printk(KERN_ERR "asoc: can't create pcm %s\n",
1333
				card->dai_link[i].stream_name);
F
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1334 1335 1336 1337 1338 1339 1340 1341 1342 1343 1344
			mutex_unlock(&codec->mutex);
			return ret;
		}
	}

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

/**
1345
 * snd_soc_init_card - register sound card
F
Frank Mandarino 已提交
1346 1347 1348 1349 1350 1351 1352
 * @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.
 */
1353
int snd_soc_init_card(struct snd_soc_device *socdev)
F
Frank Mandarino 已提交
1354
{
1355
	struct snd_soc_card *card = socdev->card;
1356
	struct snd_soc_codec *codec = card->codec;
1357
	int ret = 0, i, ac97 = 0, err = 0;
F
Frank Mandarino 已提交
1358

1359 1360 1361
	for (i = 0; i < card->num_links; i++) {
		if (card->dai_link[i].init) {
			err = card->dai_link[i].init(codec);
1362 1363
			if (err < 0) {
				printk(KERN_ERR "asoc: failed to init %s\n",
1364
					card->dai_link[i].stream_name);
1365 1366 1367
				continue;
			}
		}
M
Mark Brown 已提交
1368
		if (card->dai_link[i].codec_dai->ac97_control)
F
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			ac97 = 1;
	}
	snprintf(codec->card->shortname, sizeof(codec->card->shortname),
1372
		 "%s",  card->name);
F
Frank Mandarino 已提交
1373
	snprintf(codec->card->longname, sizeof(codec->card->longname),
1374
		 "%s (%s)", card->name, codec->name);
F
Frank Mandarino 已提交
1375 1376 1377

	ret = snd_card_register(codec->card);
	if (ret < 0) {
1378
		printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
F
Frank Mandarino 已提交
1379
				codec->name);
1380
		goto out;
F
Frank Mandarino 已提交
1381 1382
	}

1383
	mutex_lock(&codec->mutex);
F
Frank Mandarino 已提交
1384
#ifdef CONFIG_SND_SOC_AC97_BUS
1385 1386 1387 1388
	/* Only instantiate AC97 if not already done by the adaptor
	 * for the generic AC97 subsystem.
	 */
	if (ac97 && strcmp(codec->name, "AC97") != 0) {
1389 1390 1391 1392
		ret = soc_ac97_dev_register(codec);
		if (ret < 0) {
			printk(KERN_ERR "asoc: AC97 device register failed\n");
			snd_card_free(codec->card);
1393
			mutex_unlock(&codec->mutex);
1394 1395 1396
			goto out;
		}
	}
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1397 1398
#endif

1399 1400 1401 1402 1403 1404
	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)
1405
		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
1406

1407
	soc_init_codec_debugfs(codec);
F
Frank Mandarino 已提交
1408
	mutex_unlock(&codec->mutex);
1409 1410

out:
F
Frank Mandarino 已提交
1411 1412
	return ret;
}
1413
EXPORT_SYMBOL_GPL(snd_soc_init_card);
F
Frank Mandarino 已提交
1414 1415 1416 1417 1418 1419 1420 1421 1422 1423

/**
 * 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)
{
1424
	struct snd_soc_codec *codec = socdev->card->codec;
1425
#ifdef CONFIG_SND_SOC_AC97_BUS
1426
	struct snd_soc_dai *codec_dai;
1427 1428
	int i;
#endif
F
Frank Mandarino 已提交
1429 1430

	mutex_lock(&codec->mutex);
1431
	soc_cleanup_codec_debugfs(codec);
F
Frank Mandarino 已提交
1432
#ifdef CONFIG_SND_SOC_AC97_BUS
1433
	for (i = 0; i < codec->num_dai; i++) {
1434
		codec_dai = &codec->dai[i];
M
Mark Brown 已提交
1435
		if (codec_dai->ac97_control && codec->ac97) {
1436 1437 1438 1439 1440
			soc_ac97_dev_unregister(codec);
			goto free_card;
		}
	}
free_card:
F
Frank Mandarino 已提交
1441 1442 1443 1444 1445 1446 1447 1448 1449 1450 1451 1452 1453 1454 1455 1456 1457 1458 1459 1460 1461 1462 1463 1464 1465 1466 1467 1468 1469 1470 1471 1472 1473 1474 1475 1476
#endif

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

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

/**
 * snd_soc_cnew - create new control
 * @_template: control template
 * @data: control private data
M
Mark Brown 已提交
1477
 * @long_name: control long name
F
Frank Mandarino 已提交
1478 1479 1480 1481 1482 1483 1484 1485 1486 1487 1488 1489 1490 1491 1492 1493 1494 1495 1496
 *
 * Create a new mixer control from a template control.
 *
 * Returns 0 for success, else error.
 */
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
	void *data, char *long_name)
{
	struct snd_kcontrol_new template;

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

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

I
Ian Molton 已提交
1497 1498 1499 1500 1501 1502 1503 1504 1505 1506 1507 1508 1509 1510 1511 1512 1513 1514 1515 1516 1517 1518 1519 1520 1521 1522 1523 1524 1525 1526 1527
/**
 * snd_soc_add_controls - add an array of controls to a codec.
 * Convienience function to add a list of controls. Many codecs were
 * duplicating this code.
 *
 * @codec: codec to add controls to
 * @controls: array of controls to add
 * @num_controls: number of elements in the array
 *
 * Return 0 for success, else error.
 */
int snd_soc_add_controls(struct snd_soc_codec *codec,
	const struct snd_kcontrol_new *controls, int num_controls)
{
	struct snd_card *card = codec->card;
	int err, i;

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

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_add_controls);

F
Frank Mandarino 已提交
1528 1529 1530 1531 1532 1533 1534 1535 1536 1537 1538 1539 1540 1541 1542 1543 1544
/**
 * 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;
1545
	uinfo->value.enumerated.items = e->max;
F
Frank Mandarino 已提交
1546

1547 1548
	if (uinfo->value.enumerated.item > e->max - 1)
		uinfo->value.enumerated.item = e->max - 1;
F
Frank Mandarino 已提交
1549 1550 1551 1552 1553 1554 1555 1556 1557
	strcpy(uinfo->value.enumerated.name,
		e->texts[uinfo->value.enumerated.item]);
	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_info_enum_double);

/**
 * snd_soc_get_enum_double - enumerated double mixer get callback
 * @kcontrol: mixer control
M
Mark Brown 已提交
1558
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1559 1560 1561 1562 1563 1564 1565 1566 1567 1568 1569 1570
 *
 * 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;

1571
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1572 1573
		;
	val = snd_soc_read(codec, e->reg);
1574 1575
	ucontrol->value.enumerated.item[0]
		= (val >> e->shift_l) & (bitmask - 1);
F
Frank Mandarino 已提交
1576 1577 1578 1579 1580 1581 1582 1583 1584 1585 1586
	if (e->shift_l != e->shift_r)
		ucontrol->value.enumerated.item[1] =
			(val >> e->shift_r) & (bitmask - 1);

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_enum_double);

/**
 * snd_soc_put_enum_double - enumerated double mixer put callback
 * @kcontrol: mixer control
M
Mark Brown 已提交
1587
 * @ucontrol: control element information
F
Frank Mandarino 已提交
1588 1589 1590 1591 1592 1593 1594 1595 1596 1597 1598 1599 1600
 *
 * 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;

1601
	for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
F
Frank Mandarino 已提交
1602
		;
1603
	if (ucontrol->value.enumerated.item[0] > e->max - 1)
F
Frank Mandarino 已提交
1604 1605 1606 1607
		return -EINVAL;
	val = ucontrol->value.enumerated.item[0] << e->shift_l;
	mask = (bitmask - 1) << e->shift_l;
	if (e->shift_l != e->shift_r) {
1608
		if (ucontrol->value.enumerated.item[1] > e->max - 1)
F
Frank Mandarino 已提交
1609 1610 1611 1612 1613 1614 1615 1616 1617
			return -EINVAL;
		val |= ucontrol->value.enumerated.item[1] << e->shift_r;
		mask |= (bitmask - 1) << e->shift_r;
	}

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

P
Peter Ujfalusi 已提交
1618 1619 1620 1621 1622 1623 1624 1625 1626 1627 1628 1629 1630 1631 1632 1633
/**
 * snd_soc_get_value_enum_double - semi enumerated double mixer get callback
 * @kcontrol: mixer control
 * @ucontrol: control element information
 *
 * Callback to get the value of a double semi enumerated mixer.
 *
 * Semi enumerated mixer: the enumerated items are referred as values. Can be
 * used for handling bitfield coded enumeration for example.
 *
 * Returns 0 for success.
 */
int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1634
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
P
Peter Ujfalusi 已提交
1635 1636 1637 1638 1639 1640 1641 1642 1643 1644 1645 1646 1647 1648 1649 1650 1651 1652 1653 1654 1655 1656 1657 1658 1659 1660 1661 1662 1663 1664 1665 1666 1667 1668 1669 1670 1671 1672
	unsigned short reg_val, val, mux;

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

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double);

/**
 * snd_soc_put_value_enum_double - semi enumerated double mixer put callback
 * @kcontrol: mixer control
 * @ucontrol: control element information
 *
 * Callback to set the value of a double semi enumerated mixer.
 *
 * Semi enumerated mixer: the enumerated items are referred as values. Can be
 * used for handling bitfield coded enumeration for example.
 *
 * Returns 0 for success.
 */
int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
1673
	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
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Peter Ujfalusi 已提交
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	unsigned short val;
	unsigned short mask;

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

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

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/**
 * 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;
1709
	uinfo->value.enumerated.items = e->max;
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1711 1712
	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)
{
1757 1758 1759
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
1760
	unsigned int shift = mc->shift;
1761
	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
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 * @ucontrol: control element information
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 *
 * Callback to get the value of a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1787 1788
	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);
1790 1791 1792
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1793
	int max = mc->max;
1794 1795
	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
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 * @ucontrol: control element information
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 *
 * Callback to set the value of a single mixer control.
 *
 * Returns 0 for success.
 */
int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_value *ucontrol)
{
1826 1827
	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);
1829 1830 1831
	unsigned int reg = mc->reg;
	unsigned int shift = mc->shift;
	unsigned int rshift = mc->rshift;
1832
	int max = mc->max;
1833 1834
	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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Philipp Zabel 已提交
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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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Philipp Zabel 已提交
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			val2 = max - val2;
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		val_mask |= mask << rshift;
		val |= val2 << rshift;
	}
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	return snd_soc_update_bits(codec, reg, val_mask, val);
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}
EXPORT_SYMBOL_GPL(snd_soc_put_volsw);

/**
 * snd_soc_info_volsw_2r - double mixer info callback
 * @kcontrol: mixer control
 * @uinfo: control element information
 *
 * Callback to provide information about a double mixer control that
 * spans 2 codec registers.
 *
 * Returns 0 for success.
 */
int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
	struct snd_ctl_elem_info *uinfo)
{
1866 1867 1868
	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
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 * @ucontrol: control element information
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 *
 * 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)
{
1894 1895
	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);
1897 1898 1899
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1900
	int max = mc->max;
1901 1902
	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
M
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1922
 * @ucontrol: control element information
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 *
 * 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)
{
1931 1932
	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);
1934 1935 1936
	unsigned int reg = mc->reg;
	unsigned int reg2 = mc->rreg;
	unsigned int shift = mc->shift;
1937
	int max = mc->max;
1938 1939
	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;

1955 1956
	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);

1964 1965 1966 1967 1968 1969 1970 1971 1972 1973 1974 1975
/**
 * 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)
{
1976 1977 1978 1979
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
	int max = mc->max;
	int min = mc->min;
1980 1981 1982 1983 1984 1985 1986 1987 1988 1989 1990 1991

	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
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Mark Brown 已提交
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 * @ucontrol: control element information
1993 1994 1995 1996 1997 1998 1999 2000
 *
 * 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)
{
2001 2002
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
2003
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2004
	unsigned int reg = mc->reg;
2005
	int min = mc->min;
2006 2007 2008 2009 2010 2011 2012 2013 2014 2015 2016 2017 2018
	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
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Mark Brown 已提交
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 * @ucontrol: control element information
2020 2021 2022 2023 2024 2025 2026 2027
 *
 * 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)
{
2028 2029
	struct soc_mixer_control *mc =
		(struct soc_mixer_control *)kcontrol->private_value;
2030
	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
2031
	unsigned int reg = mc->reg;
2032
	int min = mc->min;
2033 2034 2035 2036 2037 2038 2039 2040 2041
	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);

2042 2043 2044 2045 2046 2047 2048 2049 2050 2051 2052 2053
/**
 * 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)
{
2054 2055
	if (dai->ops.set_sysclk)
		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
2056 2057 2058 2059 2060 2061 2062 2063
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);

/**
 * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
 * @dai: DAI
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 * @div_id: DAI specific clock divider ID
2065 2066 2067 2068 2069 2070 2071 2072 2073
 * @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)
{
2074 2075
	if (dai->ops.set_clkdiv)
		return dai->ops.set_clkdiv(dai, div_id, div);
2076 2077 2078 2079 2080 2081 2082 2083 2084 2085 2086 2087 2088 2089 2090 2091 2092
	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)
{
2093 2094
	if (dai->ops.set_pll)
		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
2095 2096 2097 2098 2099 2100 2101 2102 2103 2104 2105 2106 2107 2108
	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)
{
2109 2110
	if (dai->ops.set_fmt)
		return dai->ops.set_fmt(dai, fmt);
2111 2112 2113 2114 2115 2116 2117 2118 2119 2120 2121 2122 2123 2124 2125 2126 2127
	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)
{
2128 2129
	if (dai->ops.set_sysclk)
		return dai->ops.set_tdm_slot(dai, mask, slots);
2130 2131 2132 2133 2134 2135 2136 2137 2138 2139 2140 2141 2142 2143
	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)
{
2144 2145
	if (dai->ops.set_sysclk)
		return dai->ops.set_tristate(dai, tristate);
2146 2147 2148 2149 2150 2151 2152 2153 2154 2155 2156 2157 2158 2159
	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)
{
2160 2161
	if (dai->ops.digital_mute)
		return dai->ops.digital_mute(dai, mute);
2162 2163 2164 2165 2166
	else
		return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);

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

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

	mutex_lock(&client_mutex);
	list_add(&card->list, &card_list);
2186
	snd_soc_instantiate_cards();
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	mutex_unlock(&client_mutex);

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

	return 0;
}

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

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

	return 0;
}

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

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

	INIT_LIST_HEAD(&dai->list);

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

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

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_dai);

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

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

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

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

	return 0;

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

	return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_register_dais);

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

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

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

	INIT_LIST_HEAD(&platform->list);

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

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

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_platform);

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

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

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

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

	INIT_LIST_HEAD(&codec->list);

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

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

	return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_register_codec);

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

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

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static int __init snd_soc_init(void)
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{
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#ifdef CONFIG_DEBUG_FS
	debugfs_root = debugfs_create_dir("asoc", NULL);
	if (IS_ERR(debugfs_root) || !debugfs_root) {
		printk(KERN_WARNING
		       "ASoC: Failed to create debugfs directory\n");
		debugfs_root = NULL;
	}
#endif

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	return platform_driver_register(&soc_driver);
}

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

module_init(snd_soc_init);
module_exit(snd_soc_exit);

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