Merge branch 'master' into 85xx

1 files changed, 1587 insertions, 0 deletions

diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
new file mode 100644
index 00000000000..36519aef55d
--- /dev/null
+++ b/sound/soc/soc-core.c
@@ -0,0 +1,1587 @@
+/*
+ * soc-core.c  --  ALSA SoC Audio Layer
+ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Liam Girdwood
+ *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *         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.
+ *
+ *  Revision history
+ *    12th Aug 2005   Initial version.
+ *    25th Oct 2005   Working Codec, Interface and Platform registration.
+ *
+ *  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/driver.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;
+
+	snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s",
+		 codec->card->number, 0, codec->name);
+	err = device_register(&codec->ac97->dev);
+	if (err < 0) {
+		snd_printk(KERN_ERR "Can't register ac97 bus\n");
+		codec->ac97->dev.bus = NULL;
+		return err;
+	}
+	return 0;
+}
+#endif
+
+static inline const char* get_dai_name(int type)
+{
+	switch(type) {
+	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_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_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 subsytem 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_codec_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) {
+
+			codec_dai->pop_wait = 0;
+			snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name,
+				SND_SOC_DAPM_STREAM_STOP);
+
+			/* power down the codec power domain if no longer active */
+			if (codec->active == 0) {
+				dbg("pop wq D3 %s %s\n", codec->name,
+					codec_dai->playback.stream_name);
+		 		if (codec->dapm_event)
+					codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+			}
+		}
+	}
+	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_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_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--;
+
+	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){
+			if (codec->dapm_event)
+				codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot);
+		}
+	}
+
+	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_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_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);
+			if (codec_dai->dai_ops.digital_mute)
+				codec_dai->dai_ops.digital_mute(codec_dai, 0);
+		}
+	} else {
+		/* no delayed work - do we need to power up codec */
+		if (codec->dapm_state != SNDRV_CTL_POWER_D0) {
+
+			if (codec->dapm_event)
+				codec->dapm_event(codec, SNDRV_CTL_POWER_D1);
+
+			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);
+
+			if (codec->dapm_event)
+				codec->dapm_event(codec, SNDRV_CTL_POWER_D0);
+			if (codec_dai->dai_ops.digital_mute)
+				codec_dai->dai_ops.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);
+			if (codec_dai->dai_ops.digital_mute)
+				codec_dai->dai_ops.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_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_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: can't set interface %s hw params\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: can't set platform %s hw params\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_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_dai *codec_dai = machine->codec_dai;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	mutex_lock(&pcm_mutex);
+
+	/* apply codec digital mute */
+	if (!codec->active && codec_dai->dai_ops.digital_mute)
+		codec_dai->dai_ops.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_cpu_dai *cpu_dai = machine->cpu_dai;
+	struct snd_soc_codec_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;
+
+	/* mute any active DAC's */
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai;
+		if (dai->dai_ops.digital_mute && dai->playback.active)
+			dai->dai_ops.digital_mute(dai, 1);
+	}
+
+	if (machine->suspend_pre)
+		machine->suspend_pre(pdev, state);
+
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_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_dapm_state = codec->dapm_state;
+
+	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_cpu_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;
+}
+
+/* powers up audio subsystem after a suspend */
+static int soc_resume(struct platform_device *pdev)
+{
+ 	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;
+
+	if (machine->resume_pre)
+		machine->resume_pre(pdev);
+
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_cpu_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 DAC's */
+	for(i = 0; i < machine->num_links; i++) {
+		struct snd_soc_codec_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_cpu_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);
+
+	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_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->probe) {
+			ret = cpu_dai->probe(pdev);
+			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);
+	return 0;
+
+platform_err:
+	if (codec_dev->remove)
+		codec_dev->remove(pdev);
+
+cpu_dai_err:
+	for (i--; i >= 0; i--) {
+		struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->remove)
+			cpu_dai->remove(pdev);
+	}
+
+	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_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai;
+		if (cpu_dai->remove)
+			cpu_dai->remove(pdev);
+	}
+
+	if (machine->remove)
+		machine->remove(pdev);
+
+	return 0;
+}
+
+/* ASoC platform driver */
+static struct platform_driver soc_driver = {
+	.driver		= {
+		.name		= "soc-audio",
+	},
+	.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_codec_dai *codec_dai = dai_link->codec_dai;
+	struct snd_soc_cpu_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;
+	}
+
+	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: %4x\n", i, codec->read(codec, i));
+
+	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;
+
+	mutex_lock(&codec->mutex);
+	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].cpu_dai->type == SND_SOC_DAI_AC97)
+			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 codec %s\n",
+				codec->name);
+		goto out;
+	}
+
+#ifdef CONFIG_SND_SOC_AC97_BUS