Merge branch 'alarm' into release

[linux-2.6/mini2440.git] / sound / soc / soc-core.c

/*
 * soc-core.c  --  ALSA SoC Audio Layer
 *
 * Copyright 2005 Wolfson Microelectronics PLC.
 * Copyright 2005 Openedhand Ltd.
 *
 * Author: Liam Girdwood <lrg@slimlogic.co.uk>
 *         with code, comments and ideas from :-
 *         Richard Purdie <richard@openedhand.com>
 *
 *  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>
#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>

/* debug */
#define SOC_DEBUG 0
#if SOC_DEBUG
#define dbg(format, arg...) printk(format, ## arg)
#else
#define dbg(format, arg...)
#endif

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)");

/*
 * 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;
}

#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;

        dev_set_name(&codec->ac97->dev, "%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

static inline const char *get_dai_name(int type)
{
        switch (type) {
        case SND_SOC_DAI_AC97_BUS:
        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;
        struct snd_soc_dai_link *machine = rtd->dai;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
        struct snd_soc_dai *codec_dai = machine->codec_dai;
        int ret = 0;

        mutex_lock(&pcm_mutex);

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

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

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

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

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

        dbg("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name);
        dbg("asoc: rate mask 0x%x\n", runtime->hw.rates);
        dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min,
                runtime->hw.channels_max);
        dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min,
                runtime->hw.rate_max);

        if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
                cpu_dai->playback.active = codec_dai->playback.active = 1;
        else
                cpu_dai->capture.active = codec_dai->capture.active = 1;
        cpu_dai->active = codec_dai->active = 1;
        cpu_dai->runtime = runtime;
        socdev->codec->active++;
        mutex_unlock(&pcm_mutex);
        return 0;

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

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

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

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

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

                dbg("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");

                /* are we waiting on this codec DAI stream */
                if (codec_dai->pop_wait == 1) {

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

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

                        /* Fall into standby if no longer active */
                        if (codec->active == 0) {
                                dbg("pop wq D3 %s %s\n", codec->name,
                                        codec_dai->playback.stream_name);
                                snd_soc_dapm_set_bias_level(socdev,
                                        SND_SOC_BIAS_STANDBY);
                        }
                }
        }
        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;
        struct snd_soc_dai_link *machine = rtd->dai;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
        struct snd_soc_dai *codec_dai = machine->codec_dai;
        struct snd_soc_codec *codec = socdev->codec;

        mutex_lock(&pcm_mutex);

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

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

        /* 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);

        if (cpu_dai->ops.shutdown)
                cpu_dai->ops.shutdown(substream);

        if (codec_dai->ops.shutdown)
                codec_dai->ops.shutdown(substream);

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

        if (platform->pcm_ops->close)
                platform->pcm_ops->close(substream);
        cpu_dai->runtime = NULL;

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

                if (codec->active == 0 && codec_dai->pop_wait == 0)
                        snd_soc_dapm_set_bias_level(socdev,
                                                SND_SOC_BIAS_STANDBY);
        }

        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;
        struct snd_soc_dai_link *machine = rtd->dai;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
        struct snd_soc_dai *codec_dai = machine->codec_dai;
        struct snd_soc_codec *codec = socdev->codec;
        int ret = 0;

        mutex_lock(&pcm_mutex);

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

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

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

        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;
                }
        }

        /* we only want to start a DAPM playback stream if we are not waiting
         * on an existing one stopping */
        if (codec_dai->pop_wait) {
                /* we are waiting for the delayed work to start */
                if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
                                snd_soc_dapm_stream_event(socdev->codec,
                                        codec_dai->capture.stream_name,
                                        SND_SOC_DAPM_STREAM_START);
                else {
                        codec_dai->pop_wait = 0;
                        cancel_delayed_work(&socdev->delayed_work);
                        snd_soc_dai_digital_mute(codec_dai, 0);
                }
        } else {
                /* no delayed work - 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);

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

                        snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_ON);
                        snd_soc_dai_digital_mute(codec_dai, 0);

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

                        snd_soc_dai_digital_mute(codec_dai, 0);
                }
        }

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;
        struct snd_soc_dai_link *machine = rtd->dai;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
        struct snd_soc_dai *codec_dai = machine->codec_dai;
        int ret = 0;

        mutex_lock(&pcm_mutex);

        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");
                        goto out;
                }
        }

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

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

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

out:
        mutex_unlock(&pcm_mutex);
        return ret;

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

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

codec_err:
        if (machine->ops && machine->ops->hw_free)
                machine->ops->hw_free(substream);

        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;
        struct snd_soc_dai_link *machine = rtd->dai;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
        struct snd_soc_dai *codec_dai = machine->codec_dai;
        struct snd_soc_codec *codec = socdev->codec;

        mutex_lock(&pcm_mutex);

        /* apply codec digital mute */
        if (!codec->active)
                snd_soc_dai_digital_mute(codec_dai, 1);

        /* 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  */
        if (codec_dai->ops.hw_free)
                codec_dai->ops.hw_free(substream);

        if (cpu_dai->ops.hw_free)
                cpu_dai->ops.hw_free(substream);

        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;
        struct snd_soc_dai_link *machine = rtd->dai;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_dai *cpu_dai = machine->cpu_dai;
        struct snd_soc_dai *codec_dai = machine->codec_dai;
        int ret;

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

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

        if (cpu_dai->ops.trigger) {
                ret = cpu_dai->ops.trigger(substream, cmd);
                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)
{
        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;
        struct snd_soc_codec *codec = socdev->codec;
        int i;

        /* 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);

        /* mute any active DAC's */
        for (i = 0; i < machine->num_links; i++) {
                struct snd_soc_dai *dai = machine->dai_link[i].codec_dai;
                if (dai->dai_ops.digital_mute && dai->playback.active)
                        dai->dai_ops.digital_mute(dai, 1);
        }

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

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

        for (i = 0; i < machine->num_links; i++) {
                struct snd_soc_dai  *cpu_dai = machine->dai_link[i].cpu_dai;
                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 */
        run_delayed_work(&socdev->delayed_work);
        codec->suspend_bias_level = codec->bias_level;

        for (i = 0; i < codec->num_dai; i++) {
                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);

        for (i = 0; i < machine->num_links; i++) {
                struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
                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;
}

/* 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)
{
        struct snd_soc_device *socdev = container_of(work,
                                                     struct snd_soc_device,
                                                     deferred_resume_work);
        struct snd_soc_machine *machine = socdev->machine;
        struct snd_soc_platform *platform = socdev->platform;
        struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
        struct snd_soc_codec *codec = socdev->codec;
        struct platform_device *pdev = to_platform_device(socdev->dev);
        int i;

        /* 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");

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

        for (i = 0; i < machine->num_links; i++) {
                struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
                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);

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

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

        for (i = 0; i < machine->num_links; i++) {
                struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
                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);

        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");

        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);
                if (ret < 0)
                        return ret;
        }

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

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

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

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

        return 0;

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

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

        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;

        run_delayed_work(&socdev->delayed_work);

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

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

        for (i = 0; i < machine->num_links; i++) {
                struct snd_soc_dai *cpu_dai = machine->dai_link[i].cpu_dai;
                if (cpu_dai->remove)
                        cpu_dai->remove(pdev, cpu_dai);
        }

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

        return 0;
}

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

        rtd->dai = dai_link;
        rtd->socdev = socdev;
        codec_dai->codec = socdev->codec;

        /* check client and interface hw capabilities */
        sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name,
                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) {
                printk(KERN_ERR "asoc: can't create pcm for codec %s\n",
                        codec->name);
                kfree(rtd);
                return ret;
        }

        dai_link->pcm = pcm;
        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 */
static ssize_t codec_reg_show(struct device *dev,
        struct device_attribute *attr, char *buf)
{
        struct snd_soc_device *devdata = dev_get_drvdata(dev);
        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);
        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;

        return count;
}
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);

/**
 * 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.
 */
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
{
        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 */
        for (i = 0; i < machine->num_links; i++) {
                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;
        int ret = 0, i, ac97 = 0, err = 0;

        for (i = 0; i < machine->num_links; i++) {
                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;
                        }
                }
                if (socdev->machine->dai_link[i].codec_dai->type ==
                        SND_SOC_DAI_AC97_BUS)
                        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) {
                printk(KERN_ERR "asoc: failed to register soundcard for %s\n",
                                codec->name);
                goto out;
        }

        mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
        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);
                        mutex_unlock(&codec->mutex);
                        goto out;
                }
        }
#endif

        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)
                printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");

        mutex_unlock(&codec->mutex);

out:
        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;
#ifdef CONFIG_SND_SOC_AC97_BUS
        struct snd_soc_dai *codec_dai;
        int i;
#endif

        mutex_lock(&codec->mutex);
#ifdef CONFIG_SND_SOC_AC97_BUS
        for (i = 0; i < codec->num_dai; i++) {
                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:
#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;
        uinfo->value.enumerated.items = e->max;

        if (uinfo->value.enumerated.item > e->max - 1)
                uinfo->value.enumerated.item = e->max - 1;
        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;

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

        for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
                ;
        if (ucontrol->value.enumerated.item[0] > e->max - 1)
                return -EINVAL;
        val = ucontrol->value.enumerated.item[0] << e->shift_l;
        mask = (bitmask - 1) << e->shift_l;
        if (e->shift_l != e->shift_r) {
                if (ucontrol->value.enumerated.item[1] > e->max - 1)
                        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;
        uinfo->value.enumerated.items = e->max;

        if (uinfo->value.enumerated.item > e->max - 1)
                uinfo->value.enumerated.item = e->max - 1;
        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)
{
        int max = kcontrol->private_value;

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

        uinfo->count = 1;
        uinfo->value.integer.min = 0;
        uinfo->value.integer.max = max;
        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        int max = mc->max;
        unsigned int shift = mc->shift;
        unsigned int rshift = mc->rshift;

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

        uinfo->count = shift == rshift ? 1 : 2;
        uinfo->value.integer.min = 0;
        uinfo->value.integer.max = max;
        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        unsigned int reg = mc->reg;
        unsigned int shift = mc->shift;
        unsigned int rshift = mc->rshift;
        int max = mc->max;
        unsigned int mask = (1 << fls(max)) - 1;
        unsigned int invert = mc->invert;

        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] =
                        max - ucontrol->value.integer.value[0];
                if (shift != rshift)
                        ucontrol->value.integer.value[1] =
                                max - ucontrol->value.integer.value[1];
        }

        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        unsigned int reg = mc->reg;
        unsigned int shift = mc->shift;
        unsigned int rshift = mc->rshift;
        int max = mc->max;
        unsigned int mask = (1 << fls(max)) - 1;
        unsigned int invert = mc->invert;
        unsigned short val, val2, val_mask;

        val = (ucontrol->value.integer.value[0] & mask);
        if (invert)
                val = max - val;
        val_mask = mask << shift;
        val = val << shift;
        if (shift != rshift) {
                val2 = (ucontrol->value.integer.value[1] & mask);
                if (invert)
                        val2 = max - val2;
                val_mask |= mask << rshift;
                val |= val2 << rshift;
        }
        return snd_soc_update_bits(codec, reg, val_mask, val);
}
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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        int max = mc->max;

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

        uinfo->count = 2;
        uinfo->value.integer.min = 0;
        uinfo->value.integer.max = max;
        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        unsigned int reg = mc->reg;
        unsigned int reg2 = mc->rreg;
        unsigned int shift = mc->shift;
        int max = mc->max;
        unsigned int mask = (1<<fls(max))-1;
        unsigned int invert = mc->invert;

        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] =
                        max - ucontrol->value.integer.value[0];
                ucontrol->value.integer.value[1] =
                        max - ucontrol->value.integer.value[1];
        }

        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        unsigned int reg = mc->reg;
        unsigned int reg2 = mc->rreg;
        unsigned int shift = mc->shift;
        int max = mc->max;
        unsigned int mask = (1 << fls(max)) - 1;
        unsigned int invert = mc->invert;
        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) {
                val = max - val;
                val2 = max - val2;
        }

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

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

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

/**
 * 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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        int max = mc->max;
        int min = mc->min;

        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        unsigned int reg = mc->reg;
        int min = mc->min;
        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)
{
        struct soc_mixer_control *mc =
                (struct soc_mixer_control *)kcontrol->private_value;
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        unsigned int reg = mc->reg;
        int min = mc->min;
        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);

/**
 * 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);

static int __devinit snd_soc_init(void)
{
        printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION);
        return platform_driver_register(&soc_driver);
}

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

module_init(snd_soc_init);
module_exit(snd_soc_exit);

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