diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
new file mode 100644
index 000000000000..0cd590970883
--- /dev/null
+++ b/sound/soc/codecs/wl1273.c
@@ -0,0 +1,525 @@
+/*
+ * ALSA SoC WL1273 codec driver
+ *
+ * Author:      Matti Aaltonen, <matti.j.aaltonen@nokia.com>
+ *
+ * Copyright:   (C) 2010 Nokia Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/mfd/wl1273-core.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wl1273.h"
+
+enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX };
+
+/* codec private data */
+struct wl1273_priv {
+	enum wl1273_mode mode;
+	struct wl1273_core *core;
+	unsigned int channels;
+};
+
+static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core,
+				      int rate, int width)
+{
+	struct device *dev = &core->i2c_dev->dev;
+	int r = 0;
+	u16 mode;
+
+	dev_dbg(dev, "rate: %d\n", rate);
+	dev_dbg(dev, "width: %d\n", width);
+
+	mutex_lock(&core->lock);
+
+	mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE;
+
+	switch (rate) {
+	case 48000:
+		mode |= WL1273_IS2_RATE_48K;
+		break;
+	case 44100:
+		mode |= WL1273_IS2_RATE_44_1K;
+		break;
+	case 32000:
+		mode |= WL1273_IS2_RATE_32K;
+		break;
+	case 22050:
+		mode |= WL1273_IS2_RATE_22_05K;
+		break;
+	case 16000:
+		mode |= WL1273_IS2_RATE_16K;
+		break;
+	case 12000:
+		mode |= WL1273_IS2_RATE_12K;
+		break;
+	case 11025:
+		mode |= WL1273_IS2_RATE_11_025;
+		break;
+	case 8000:
+		mode |= WL1273_IS2_RATE_8K;
+		break;
+	default:
+		dev_err(dev, "Sampling rate: %d not supported\n", rate);
+		r = -EINVAL;
+		goto out;
+	}
+
+	switch (width) {
+	case 16:
+		mode |= WL1273_IS2_WIDTH_32;
+		break;
+	case 20:
+		mode |= WL1273_IS2_WIDTH_40;
+		break;
+	case 24:
+		mode |= WL1273_IS2_WIDTH_48;
+		break;
+	case 25:
+		mode |= WL1273_IS2_WIDTH_50;
+		break;
+	case 30:
+		mode |= WL1273_IS2_WIDTH_60;
+		break;
+	case 32:
+		mode |= WL1273_IS2_WIDTH_64;
+		break;
+	case 40:
+		mode |= WL1273_IS2_WIDTH_80;
+		break;
+	case 48:
+		mode |= WL1273_IS2_WIDTH_96;
+		break;
+	case 64:
+		mode |= WL1273_IS2_WIDTH_128;
+		break;
+	default:
+		dev_err(dev, "Data width: %d not supported\n", width);
+		r = -EINVAL;
+		goto out;
+	}
+
+	dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n",  WL1273_I2S_DEF_MODE);
+	dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode);
+	dev_dbg(dev, "mode: 0x%04x\n", mode);
+
+	if (core->i2s_mode != mode) {
+		r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode);
+		if (r)
+			goto out;
+
+		core->i2s_mode = mode;
+		r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE,
+					WL1273_AUDIO_ENABLE_I2S);
+		if (r)
+			goto out;
+	}
+out:
+	mutex_unlock(&core->lock);
+
+	return r;
+}
+
+static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core,
+					    int channel_number)
+{
+	struct i2c_client *client = core->i2c_dev;
+	struct device *dev = &client->dev;
+	int r = 0;
+
+	dev_dbg(dev, "%s\n", __func__);
+
+	mutex_lock(&core->lock);
+
+	if (core->channel_number == channel_number)
+		goto out;
+
+	if (channel_number == 1 && core->mode == WL1273_MODE_RX)
+		r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
+					WL1273_RX_MONO);
+	else if (channel_number == 1 && core->mode == WL1273_MODE_TX)
+		r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
+					WL1273_TX_MONO);
+	else if (channel_number == 2 && core->mode == WL1273_MODE_RX)
+		r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
+					WL1273_RX_STEREO);
+	else if (channel_number == 2 && core->mode == WL1273_MODE_TX)
+		r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
+					WL1273_TX_STEREO);
+	else
+		r = -EINVAL;
+out:
+	mutex_unlock(&core->lock);
+
+	return r;
+}
+
+static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = wl1273->mode;
+
+	return 0;
+}
+
+static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
+
+static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	/* Do not allow changes while stream is running */
+	if (codec->active)
+		return -EPERM;
+
+	if (ucontrol->value.integer.value[0] < 0 ||
+	    ucontrol->value.integer.value[0] >=  ARRAY_SIZE(wl1273_audio_route))
+		return -EINVAL;
+
+	wl1273->mode = ucontrol->value.integer.value[0];
+
+	return 1;
+}
+
+static const struct soc_enum wl1273_enum =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route);
+
+static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+	ucontrol->value.integer.value[0] = wl1273->core->audio_mode;
+
+	return 0;
+}
+
+static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+	int val, r = 0;
+
+	dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+	val = ucontrol->value.integer.value[0];
+	if (wl1273->core->audio_mode == val)
+		return 0;
+
+	r = wl1273_fm_set_audio(wl1273->core, val);
+	if (r < 0)
+		return r;
+
+	return 1;
+}
+
+static const char *wl1273_audio_strings[] = { "Digital", "Analog" };
+
+static const struct soc_enum wl1273_audio_enum =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
+			    wl1273_audio_strings);
+
+static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+	ucontrol->value.integer.value[0] = wl1273->core->volume;
+
+	return 0;
+}
+
+static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+	int r;
+
+	dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+	r = wl1273_fm_set_volume(wl1273->core,
+				 ucontrol->value.integer.value[0]);
+	if (r)
+		return r;
+
+	return 1;
+}
+
+static const struct snd_kcontrol_new wl1273_controls[] = {
+	SOC_ENUM_EXT("Codec Mode", wl1273_enum,
+		     snd_wl1273_get_audio_route, snd_wl1273_set_audio_route),
+	SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum,
+		     snd_wl1273_fm_audio_get,  snd_wl1273_fm_audio_put),
+	SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0,
+		       snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put),
+};
+
+static int wl1273_startup(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	switch (wl1273->mode) {
+	case WL1273_MODE_BT:
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_RATE,
+					     8000, 8000);
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1);
+		break;
+	case WL1273_MODE_FM_RX:
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			pr_err("Cannot play in RX mode.\n");
+			return -EINVAL;
+		}
+		break;
+	case WL1273_MODE_FM_TX:
+		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+			pr_err("Cannot capture in TX mode.\n");
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+		break;
+	}
+
+	return 0;
+}
+
+static int wl1273_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
+	struct wl1273_core *core = wl1273->core;
+	unsigned int rate, width, r;
+
+	if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) {
+		pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n");
+		return -EINVAL;
+	}
+
+	rate = params_rate(params);
+	width =  hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+
+	if (wl1273->mode == WL1273_MODE_BT) {
+		if (rate != 8000) {
+			pr_err("Rate %d not supported.\n", params_rate(params));
+			return -EINVAL;
+		}
+
+		if (params_channels(params) != 1) {
+			pr_err("Only mono supported.\n");
+			return -EINVAL;
+		}
+
+		return 0;
+	}
+
+	if (wl1273->mode == WL1273_MODE_FM_TX &&
+	    substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		pr_err("Only playback supported with TX.\n");
+		return -EINVAL;
+	}
+
+	if (wl1273->mode == WL1273_MODE_FM_RX  &&
+	    substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		pr_err("Only capture supported with RX.\n");
+		return -EINVAL;
+	}
+
+	if (wl1273->mode != WL1273_MODE_FM_RX  &&
+	    wl1273->mode != WL1273_MODE_FM_TX) {
+		pr_err("Unexpected mode: %d.\n", wl1273->mode);
+		return -EINVAL;
+	}
+
+	r = snd_wl1273_fm_set_i2s_mode(core, rate, width);
+	if (r)
+		return r;
+
+	wl1273->channels = params_channels(params);
+	r = snd_wl1273_fm_set_channel_number(core, wl1273->channels);
+	if (r)
+		return r;
+
+	return 0;
+}
+
+static struct snd_soc_dai_ops wl1273_dai_ops = {
+	.startup	= wl1273_startup,
+	.hw_params	= wl1273_hw_params,
+};
+
+static struct snd_soc_dai_driver wl1273_dai = {
+	.name = "wl1273-fm",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE},
+	.ops = &wl1273_dai_ops,
+};
+
+/* Audio interface format for the soc_card driver */
+int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt)
+{
+	struct wl1273_priv *wl1273;
+
+	if (codec == NULL || fmt == NULL)
+		return -EINVAL;
+
+	wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	switch (wl1273->mode) {
+	case WL1273_MODE_FM_RX:
+	case WL1273_MODE_FM_TX:
+		*fmt =	SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+
+		break;
+	case WL1273_MODE_BT:
+		*fmt =	SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_IB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wl1273_get_format);
+
+static int wl1273_probe(struct snd_soc_codec *codec)
+{
+	struct wl1273_core **core = codec->dev->platform_data;
+	struct wl1273_priv *wl1273;
+	int r;
+
+	dev_dbg(codec->dev, "%s.\n", __func__);
+
+	if (!core) {
+		dev_err(codec->dev, "Platform data is missing.\n");
+		return -EINVAL;
+	}
+
+	wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL);
+	if (wl1273 == NULL) {
+		dev_err(codec->dev, "Cannot allocate memory.\n");
+		return -ENOMEM;
+	}
+
+	wl1273->mode = WL1273_MODE_BT;
+	wl1273->core = *core;
+
+	snd_soc_codec_set_drvdata(codec, wl1273);
+	mutex_init(&codec->mutex);
+
+	r = snd_soc_add_controls(codec, wl1273_controls,
+				 ARRAY_SIZE(wl1273_controls));
+	if (r)
+		kfree(wl1273);
+
+	return r;
+}
+
+static int wl1273_remove(struct snd_soc_codec *codec)
+{
+	struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+	dev_dbg(codec->dev, "%s\n", __func__);
+	kfree(wl1273);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
+	.probe = wl1273_probe,
+	.remove = wl1273_remove,
+};
+
+static int __devinit wl1273_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wl1273,
+				      &wl1273_dai, 1);
+}
+
+static int __devexit wl1273_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+MODULE_ALIAS("platform:wl1273-codec");
+
+static struct platform_driver wl1273_platform_driver = {
+	.driver		= {
+		.name	= "wl1273-codec",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wl1273_platform_probe,
+	.remove		= __devexit_p(wl1273_platform_remove),
+};
+
+static int __init wl1273_init(void)
+{
+	return platform_driver_register(&wl1273_platform_driver);
+}
+module_init(wl1273_init);
+
+static void __exit wl1273_exit(void)
+{
+	platform_driver_unregister(&wl1273_platform_driver);
+}
+module_exit(wl1273_exit);
+
+MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen@nokia.com>");
+MODULE_DESCRIPTION("ASoC WL1273 codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h
new file mode 100644
index 000000000000..14ed027fdcfc
--- /dev/null
+++ b/sound/soc/codecs/wl1273.h
@@ -0,0 +1,101 @@
+/*
+ * sound/soc/codec/wl1273.h
+ *
+ * ALSA SoC WL1273 codec driver
+ *
+ * Copyright (C) Nokia Corporation
+ * Author: Matti Aaltonen <matti.j.aaltonen@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __WL1273_CODEC_H__
+#define __WL1273_CODEC_H__
+
+/* I2S protocol, left channel first, data width 16 bits */
+#define WL1273_PCM_DEF_MODE		0x00
+
+/* Rx */
+#define WL1273_AUDIO_ENABLE_I2S		(1 << 0)
+#define WL1273_AUDIO_ENABLE_ANALOG	(1 << 1)
+
+/* Tx */
+#define WL1273_AUDIO_IO_SET_ANALOG	0
+#define WL1273_AUDIO_IO_SET_I2S		1
+
+#define WL1273_POWER_SET_OFF		0
+#define WL1273_POWER_SET_FM		(1 << 0)
+#define WL1273_POWER_SET_RDS		(1 << 1)
+#define WL1273_POWER_SET_RETENTION	(1 << 4)
+
+#define WL1273_PUPD_SET_OFF		0x00
+#define WL1273_PUPD_SET_ON		0x01
+#define WL1273_PUPD_SET_RETENTION	0x10
+
+/* I2S mode */
+#define WL1273_IS2_WIDTH_32	0x0
+#define WL1273_IS2_WIDTH_40	0x1
+#define WL1273_IS2_WIDTH_22_23	0x2
+#define WL1273_IS2_WIDTH_23_22	0x3
+#define WL1273_IS2_WIDTH_48	0x4
+#define WL1273_IS2_WIDTH_50	0x5
+#define WL1273_IS2_WIDTH_60	0x6
+#define WL1273_IS2_WIDTH_64	0x7
+#define WL1273_IS2_WIDTH_80	0x8
+#define WL1273_IS2_WIDTH_96	0x9
+#define WL1273_IS2_WIDTH_128	0xa
+#define WL1273_IS2_WIDTH	0xf
+
+#define WL1273_IS2_FORMAT_STD	(0x0 << 4)
+#define WL1273_IS2_FORMAT_LEFT	(0x1 << 4)
+#define WL1273_IS2_FORMAT_RIGHT	(0x2 << 4)
+#define WL1273_IS2_FORMAT_USER	(0x3 << 4)
+
+#define WL1273_IS2_MASTER	(0x0 << 6)
+#define WL1273_IS2_SLAVEW	(0x1 << 6)
+
+#define WL1273_IS2_TRI_AFTER_SENDING	(0x0 << 7)
+#define WL1273_IS2_TRI_ALWAYS_ACTIVE	(0x1 << 7)
+
+#define WL1273_IS2_SDOWS_RR	(0x0 << 8)
+#define WL1273_IS2_SDOWS_RF	(0x1 << 8)
+#define WL1273_IS2_SDOWS_FR	(0x2 << 8)
+#define WL1273_IS2_SDOWS_FF	(0x3 << 8)
+
+#define WL1273_IS2_TRI_OPT	(0x0 << 10)
+#define WL1273_IS2_TRI_ALWAYS	(0x1 << 10)
+
+#define WL1273_IS2_RATE_48K	(0x0 << 12)
+#define WL1273_IS2_RATE_44_1K	(0x1 << 12)
+#define WL1273_IS2_RATE_32K	(0x2 << 12)
+#define WL1273_IS2_RATE_22_05K	(0x4 << 12)
+#define WL1273_IS2_RATE_16K	(0x5 << 12)
+#define WL1273_IS2_RATE_12K	(0x8 << 12)
+#define WL1273_IS2_RATE_11_025	(0x9 << 12)
+#define WL1273_IS2_RATE_8K	(0xa << 12)
+#define WL1273_IS2_RATE		(0xf << 12)
+
+#define WL1273_I2S_DEF_MODE	(WL1273_IS2_WIDTH_32 | \
+				 WL1273_IS2_FORMAT_STD | \
+				 WL1273_IS2_MASTER | \
+				 WL1273_IS2_TRI_AFTER_SENDING | \
+				 WL1273_IS2_SDOWS_RR | \
+				 WL1273_IS2_TRI_OPT | \
+				 WL1273_IS2_RATE_48K)
+
+int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt);
+
+#endif	/* End of __WL1273_CODEC_H__ */
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 5a6f56d63756..f039e8db0765 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -60,6 +60,7 @@ struct dma_object {
 	struct snd_soc_platform_driver dai;
 	dma_addr_t ssi_stx_phys;
 	dma_addr_t ssi_srx_phys;
+	unsigned int ssi_fifo_depth;
 	struct ccsr_dma_channel __iomem *channel;
 	unsigned int irq;
 	bool assigned;
@@ -99,6 +100,7 @@ struct fsl_dma_private {
 	unsigned int irq;
 	struct snd_pcm_substream *substream;
 	dma_addr_t ssi_sxx_phys;
+	unsigned int ssi_fifo_depth;
 	dma_addr_t ld_buf_phys;
 	unsigned int current_link;
 	dma_addr_t dma_buf_phys;
@@ -439,6 +441,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 	else
 		dma_private->ssi_sxx_phys = dma->ssi_srx_phys;
 
+	dma_private->ssi_fifo_depth = dma->ssi_fifo_depth;
 	dma_private->dma_channel = dma->channel;
 	dma_private->irq = dma->irq;
 	dma_private->substream = substream;
@@ -552,11 +555,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 	struct device *dev = rtd->platform->dev;
 
 	/* Number of bits per sample */
-	unsigned int sample_size =
+	unsigned int sample_bits =
 		snd_pcm_format_physical_width(params_format(hw_params));
 
 	/* Number of bytes per frame */
-	unsigned int frame_size = 2 * (sample_size / 8);
+	unsigned int sample_bytes = sample_bits / 8;
 
 	/* Bus address of SSI STX register */
 	dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
@@ -596,7 +599,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 	 * that offset here.  While we're at it, also tell the DMA controller
 	 * how much data to transfer per sample.
 	 */
-	switch (sample_size) {
+	switch (sample_bits) {
 	case 8:
 		mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
 		ssi_sxx_phys += 3;
@@ -610,22 +613,42 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
 		break;
 	default:
 		/* We should never get here */
-		dev_err(dev, "unsupported sample size %u\n", sample_size);
+		dev_err(dev, "unsupported sample size %u\n", sample_bits);
 		return -EINVAL;
 	}
 
 	/*
-	 * BWC should always be a multiple of the frame size.  BWC determines
-	 * how many bytes are sent/received before the DMA controller checks the
-	 * SSI to see if it needs to stop.  For playback, the transmit FIFO can
-	 * hold three frames, so we want to send two frames at a time. For
-	 * capture, the receive FIFO is triggered when it contains one frame, so
-	 * we want to receive one frame at a time.
+	 * BWC determines how many bytes are sent/received before the DMA
+	 * controller checks the SSI to see if it needs to stop. BWC should
+	 * always be a multiple of the frame size, so that we always transmit
+	 * whole frames.  Each frame occupies two slots in the FIFO.  The
+	 * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two
+	 * (MR[BWC] can only represent even powers of two).
+	 *
+	 * To simplify the process, we set BWC to the largest value that is
+	 * less than or equal to the FIFO watermark.  For playback, this ensures
+	 * that we transfer the maximum amount without overrunning the FIFO.
+	 * For capture, this ensures that we transfer the maximum amount without
+	 * underrunning the FIFO.
+	 *
+	 * f = SSI FIFO depth
+	 * w = SSI watermark value (which equals f - 2)
+	 * b = DMA bandwidth count (in bytes)
+	 * s = sample size (in bytes, which equals frame_size * 2)
+	 *
+	 * For playback, we never transmit more than the transmit FIFO
+	 * watermark, otherwise we might write more data than the FIFO can hold.
+	 * The watermark is equal to the FIFO depth minus two.
+	 *
+	 * For capture, two equations must hold:
+	 *	w > f - (b / s)
+	 *	w >= b / s
+	 *
+	 * So, b > 2 * s, but b must also be <= s * w.  To simplify, we set
+	 * b = s * w, which is equal to
+	 *      (dma_private->ssi_fifo_depth - 2) * sample_bytes.
 	 */
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		mr |= CCSR_DMA_MR_BWC(2 * frame_size);
-	else
-		mr |= CCSR_DMA_MR_BWC(frame_size);
+	mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes);
 
 	out_be32(&dma_channel->mr, mr);
 
@@ -879,6 +902,7 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	struct device_node *np = of_dev->dev.of_node;
 	struct device_node *ssi_np;
 	struct resource res;
+	const uint32_t *iprop;
 	int ret;
 
 	/* Find the SSI node that points to us. */
@@ -889,15 +913,17 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	}
 
 	ret = of_address_to_resource(ssi_np, 0, &res);
-	of_node_put(ssi_np);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not determine device resources\n");
+		dev_err(&of_dev->dev, "could not determine resources for %s\n",
+			ssi_np->full_name);
+		of_node_put(ssi_np);
 		return ret;
 	}
 
 	dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
 	if (!dma) {
 		dev_err(&of_dev->dev, "could not allocate dma object\n");
+		of_node_put(ssi_np);
 		return -ENOMEM;
 	}
 
@@ -910,6 +936,15 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
 	dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
 
+	iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
+	if (iprop)
+		dma->ssi_fifo_depth = *iprop;
+	else
+                /* Older 8610 DTs didn't have the fifo-depth property */
+		dma->ssi_fifo_depth = 8;
+
+	of_node_put(ssi_np);
+
 	ret = snd_soc_register_platform(&of_dev->dev, &dma->dai);
 	if (ret) {
 		dev_err(&of_dev->dev, "could not register platform\n");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 7939c337ed9d..d1c855ade8fb 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -93,6 +93,7 @@ struct fsl_ssi_private {
 	unsigned int playback;
 	unsigned int capture;
 	int asynchronous;
+	unsigned int fifo_depth;
 	struct snd_soc_dai_driver cpu_dai_drv;
 	struct device_attribute dev_attr;
 	struct platform_device *pdev;
@@ -337,11 +338,20 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 
 		/*
 		 * Set the watermark for transmit FIFI 0 and receive FIFO 0. We
-		 * don't use FIFO 1.  Since the SSI only supports stereo, the
-		 * watermark should never be an odd number.
+		 * don't use FIFO 1.  We program the transmit water to signal a
+		 * DMA transfer if there are only two (or fewer) elements left
+		 * in the FIFO.  Two elements equals one frame (left channel,
+		 * right channel).  This value, however, depends on the depth of
+		 * the transmit buffer.
+		 *
+		 * We program the receive FIFO to notify us if at least two
+		 * elements (one frame) have been written to the FIFO.  We could
+		 * make this value larger (and maybe we should), but this way
+		 * data will be written to memory as soon as it's available.
 		 */
 		out_be32(&ssi->sfcsr,
-			 CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2));
+			CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
+			CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2));
 
 		/*
 		 * We keep the SSI disabled because if we enable it, then the
@@ -622,6 +632,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	struct device_attribute *dev_attr = NULL;
 	struct device_node *np = of_dev->dev.of_node;
 	const char *p, *sprop;
+	const uint32_t *iprop;
 	struct resource res;
 	char name[64];
 
@@ -678,6 +689,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	else
 		ssi_private->cpu_dai_drv.symmetric_rates = 1;
 
+	/* Determine the FIFO depth. */
+	iprop = of_get_property(np, "fsl,fifo-depth", NULL);
+	if (iprop)
+		ssi_private->fifo_depth = *iprop;
+	else
+                /* Older 8610 DTs didn't have the fifo-depth property */
+		ssi_private->fifo_depth = 8;
+
 	/* Initialize the the device_attribute structure */
 	dev_attr = &ssi_private->dev_attr;
 	dev_attr->attr.name = "statistics";