Merge branch 'topic/workq-update' into topic/misc

73 files changed, 496 insertions, 367 deletions

diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c
index de8e03afa97..faa31749054 100644
--- a/sound/aoa/core/gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -287,10 +287,9 @@ static void ftr_gpio_exit(struct gpio_runtime *rt)
 		free_irq(linein_detect_irq, &rt->line_in_notify);
 	if (rt->line_out_notify.gpio_private)
 		free_irq(lineout_detect_irq, &rt->line_out_notify);
-	cancel_delayed_work(&rt->headphone_notify.work);
-	cancel_delayed_work(&rt->line_in_notify.work);
-	cancel_delayed_work(&rt->line_out_notify.work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
 	mutex_destroy(&rt->headphone_notify.mutex);
 	mutex_destroy(&rt->line_in_notify.mutex);
 	mutex_destroy(&rt->line_out_notify.mutex);
diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c
index 7e267c9379b..c8d8a1a6f96 100644
--- a/sound/aoa/core/gpio-pmf.c
+++ b/sound/aoa/core/gpio-pmf.c
@@ -107,10 +107,9 @@ static void pmf_gpio_exit(struct gpio_runtime *rt)
 
 	/* make sure no work is pending before freeing
 	 * all things */
-	cancel_delayed_work(&rt->headphone_notify.work);
-	cancel_delayed_work(&rt->line_in_notify.work);
-	cancel_delayed_work(&rt->line_out_notify.work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
 
 	mutex_destroy(&rt->headphone_notify.mutex);
 	mutex_destroy(&rt->line_in_notify.mutex);
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index 4e47baada66..6e240918189 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -422,7 +422,7 @@ static int __devinit atmel_abdac_probe(struct platform_device *pdev)
 	sample_clk = clk_get(&pdev->dev, "sample_clk");
 	if (IS_ERR(sample_clk)) {
 		dev_dbg(&pdev->dev, "no sample clock\n");
-		retval = PTR_ERR(pclk);
+		retval = PTR_ERR(sample_clk);
 		goto out_put_pclk;
 	}
 	clk_enable(pclk);
diff --git a/sound/core/info.c b/sound/core/info.c
index b70564ed8b3..7077f601da5 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -23,7 +23,6 @@
 #include <linux/time.h>
 #include <linux/mm.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/string.h>
 #include <sound/core.h>
 #include <sound/minors.h>
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 5c8c7dff8ed..b753ec661fc 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1510,16 +1510,19 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use
 static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file)
 {
 	struct snd_pcm_substream *substream;
+	struct snd_pcm_runtime *runtime;
+	int i;
 
-	substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
-	if (substream != NULL) {
-		snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-		substream->runtime->oss.prepare = 1;
-	}
-	substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE];
-	if (substream != NULL) {
+	for (i = 0; i < 2; i++) { 
+		substream = pcm_oss_file->streams[i];
+		if (!substream)
+			continue;
+		runtime = substream->runtime;
 		snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
-		substream->runtime->oss.prepare = 1;
+		runtime->oss.prepare = 1;
+		runtime->oss.buffer_used = 0;
+		runtime->oss.prev_hw_ptr_period = 0;
+		runtime->oss.period_ptr = 0;
 	}
 	return 0;
 }
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index f91a439f675..0db714e87a8 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -22,7 +22,6 @@
 #include <linux/mm.h>
 #include <linux/file.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/time.h>
 #include <linux/pm_qos_params.h>
 #include <linux/uio.h>
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 345caea2d74..1c7a3efe177 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -21,7 +21,6 @@
 
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/time.h>
 #include <linux/device.h>
 #include <linux/moduleparam.h>
diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c
index 971a84a4fa7..c424d329f80 100644
--- a/sound/i2c/other/ak4113.c
+++ b/sound/i2c/other/ak4113.c
@@ -57,8 +57,7 @@ static void snd_ak4113_free(struct ak4113 *chip)
 {
 	chip->init = 1;	/* don't schedule new work */
 	mb();
-	cancel_delayed_work(&chip->work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&chip->work);
 	kfree(chip);
 }
 
@@ -141,7 +140,7 @@ void snd_ak4113_reinit(struct ak4113 *chip)
 {
 	chip->init = 1;
 	mb();
-	flush_scheduled_work();
+	flush_delayed_work_sync(&chip->work);
 	ak4113_init_regs(chip);
 	/* bring up statistics / event queing */
 	chip->init = 0;
diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c
index 0341451f814..d9fb537b0b9 100644
--- a/sound/i2c/other/ak4114.c
+++ b/sound/i2c/other/ak4114.c
@@ -67,8 +67,7 @@ static void snd_ak4114_free(struct ak4114 *chip)
 {
 	chip->init = 1;	/* don't schedule new work */
 	mb();
-	cancel_delayed_work(&chip->work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&chip->work);
 	kfree(chip);
 }
 
@@ -154,7 +153,7 @@ void snd_ak4114_reinit(struct ak4114 *chip)
 {
 	chip->init = 1;
 	mb();
-	flush_scheduled_work();
+	flush_delayed_work_sync(&chip->work);
 	ak4114_init_regs(chip);
 	/* bring up statistics / event queing */
 	chip->init = 0;
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index a7630e9edf8..0fc614ce16c 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -1014,8 +1014,7 @@ static int snd_ac97_free(struct snd_ac97 *ac97)
 {
 	if (ac97) {
 #ifdef CONFIG_SND_AC97_POWER_SAVE
-		cancel_delayed_work(&ac97->power_work);
-		flush_scheduled_work();
+		cancel_delayed_work_sync(&ac97->power_work);
 #endif
 		snd_ac97_proc_done(ac97);
 		if (ac97->bus)
@@ -2456,8 +2455,7 @@ void snd_ac97_suspend(struct snd_ac97 *ac97)
 	if (ac97->build_ops->suspend)
 		ac97->build_ops->suspend(ac97);
 #ifdef CONFIG_SND_AC97_POWER_SAVE
-	cancel_delayed_work(&ac97->power_work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&ac97->power_work);
 #endif
 	snd_ac97_powerdown(ac97);
 }
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3cfb31e77b1..846d1ead47f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3111,6 +3111,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21c8, "Thinkpad Edge 11", CXT5066_IDEAPAD),
  	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
  	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G series", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x390a, "Lenovo S10-3t", CXT5066_IDEAPAD),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b7e234898fd..f17159d7dbe 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1614,6 +1614,7 @@ do_sku:
 		spec->init_amp = ALC_INIT_GPIO3;
 		break;
 	case 5:
+	default:
 		spec->init_amp = ALC_INIT_DEFAULT;
 		break;
 	}
@@ -2014,6 +2015,36 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 };
 
 /*
+ *ALC888 Acer Aspire 7730G model
+ */
+
+static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+/* Bias voltage on for external mic port */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Front Mic: set to PIN_IN (empty by default) */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+/* Unselect Front Mic by default in input mixer 3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)},
+/* Enable unsolicited event for HP jack */
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/* Enable headphone output */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+/*Enable internal subwoofer */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x17, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+/*
  * ALC889 Acer Aspire 8930G model
  */
 
@@ -2200,6 +2231,16 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x17;
 }
 
+static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->autocfg.speaker_pins[2] = 0x17;
+}
+
 static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -9524,13 +9565,6 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
 	{ }
 };
 
-static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	{ } /* end */
-};
-
 static void alc888_6st_dell_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -9831,7 +9865,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
 	SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY),
-	SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
 	SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
@@ -10328,7 +10361,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.const_channel_count = 6,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.setup = alc888_acer_aspire_6530g_setup,
+		.setup = alc888_acer_aspire_7730g_setup,
 		.init_hook = alc_automute_amp,
 	},
 	[ALC883_MEDION] = {
@@ -14629,7 +14662,10 @@ static int alc275_setup_dual_adc(struct hda_codec *codec)
 /* different alc269-variants */
 enum {
 	ALC269_TYPE_NORMAL,
+	ALC269_TYPE_ALC258,
 	ALC269_TYPE_ALC259,
+	ALC269_TYPE_ALC269VB,
+	ALC269_TYPE_ALC270,
 	ALC269_TYPE_ALC271X,
 };
 
@@ -15029,7 +15065,7 @@ static int alc269_fill_coef(struct hda_codec *codec)
 static int patch_alc269(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
-	int board_config;
+	int board_config, coef;
 	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -15040,14 +15076,23 @@ static int patch_alc269(struct hda_codec *codec)
 
 	alc_auto_parse_customize_define(codec);
 
-	if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){
+	coef = alc_read_coef_idx(codec, 0);
+	if ((coef & 0x00f0) == 0x0010) {
 		if (codec->bus->pci->subsystem_vendor == 0x1025 &&
 		    spec->cdefine.platform_type == 1) {
 			alc_codec_rename(codec, "ALC271X");
 			spec->codec_variant = ALC269_TYPE_ALC271X;
-		} else {
+		} else if ((coef & 0xf000) == 0x1000) {
+			spec->codec_variant = ALC269_TYPE_ALC270;
+		} else if ((coef & 0xf000) == 0x2000) {
 			alc_codec_rename(codec, "ALC259");
 			spec->codec_variant = ALC269_TYPE_ALC259;
+		} else if ((coef & 0xf000) == 0x3000) {
+			alc_codec_rename(codec, "ALC258");
+			spec->codec_variant = ALC269_TYPE_ALC258;
+		} else {
+			alc_codec_rename(codec, "ALC269VB");
+			spec->codec_variant = ALC269_TYPE_ALC269VB;
 		}
 	} else
 		alc_fix_pll_init(codec, 0x20, 0x04, 15);
@@ -15110,7 +15155,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
 
 	if (!spec->adc_nids) { /* wasn't filled automatically? use default */
-		if (spec->codec_variant != ALC269_TYPE_NORMAL) {
+		if (spec->codec_variant == ALC269_TYPE_NORMAL) {
 			spec->adc_nids = alc269_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
 			spec->capsrc_nids = alc269_capsrc_nids;
@@ -16904,7 +16949,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+	return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x22, 0);
 }
 
 
@@ -18958,6 +19003,8 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
 		return 0x02;
 	else if (nid >= 0x0c && nid <= 0x0e)
 		return nid - 0x0c + 0x02;
+	else if (nid == 0x26) /* ALC887-VD has this DAC too */
+		return 0x25;
 	else
 		return 0;
 }
@@ -18966,7 +19013,7 @@ static inline hda_nid_t alc662_mix_to_dac(hda_nid_t nid)
 static hda_nid_t alc662_dac_to_mix(struct hda_codec *codec, hda_nid_t pin,
 				   hda_nid_t dac)
 {
-	hda_nid_t mix[4];
+	hda_nid_t mix[5];
 	int i, num;
 
 	num = snd_hda_get_connections(codec, pin, mix, ARRAY_SIZE(mix));
@@ -19426,7 +19473,10 @@ static int patch_alc888(struct hda_codec *codec)
 {
 	if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){
 		kfree(codec->chip_name);
-		codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
+		if (codec->vendor_id == 0x10ec0887)
+			codec->chip_name = kstrdup("ALC887-VD", GFP_KERNEL);
+		else
+			codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL);
 		if (!codec->chip_name) {
 			alc_free(codec);
 			return -ENOMEM;
@@ -19916,7 +19966,7 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
+	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc888 },
 	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
 	  .patch = patch_alc882 },
 	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index cfd73afad88..efa4225f5fd 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1627,6 +1627,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
 		      "Alienware M17x", STAC_ALIENWARE_M17X),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
+		      "Alienware M17x", STAC_ALIENWARE_M17X),
 	{} /* terminator */
 };
 
@@ -3491,10 +3493,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 				return err;
 		}
 
-		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1)
 			snd_hda_add_imux_item(imux, label, index, NULL);
-			spec->num_analog_muxes++;
-		}
 	}
 
 	return 0;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index d1c3f8defc4..7f4852a478a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -263,8 +263,7 @@ static void vt1708_stop_hp_work(struct via_spec *spec)
 		return;
 	snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81,
 			    !spec->vt1708_jack_detectect);
-	cancel_delayed_work(&spec->vt1708_hp_work);
-	flush_scheduled_work();
+	cancel_delayed_work_sync(&spec->vt1708_hp_work);
 }
 
 
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 3078ed66ad6..c44c91e6fb1 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -565,7 +565,8 @@ static void oxygen_card_free(struct snd_card *card)
 	oxygen_shutdown(chip);
 	if (chip->irq >= 0)
 		free_irq(chip->irq, chip);
-	flush_scheduled_work();
+	flush_work_sync(&chip->spdif_input_bits_work);
+	flush_work_sync(&chip->gpio_work);
 	chip->model.cleanup(chip);
 	kfree(chip->model_data);
 	mutex_destroy(&chip->mutex);
@@ -741,7 +742,8 @@ int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
 	spin_unlock_irq(&chip->reg_lock);
 
 	synchronize_irq(chip->irq);
-	flush_scheduled_work();
+	flush_work_sync(&chip->spdif_input_bits_work);
+	flush_work_sync(&chip->gpio_work);
 	chip->interrupt_mask = saved_interrupt_mask;
 
 	pci_disable_device(pci);
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index e720d5e6f04..bee3c94f58b 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -16,7 +16,8 @@ config SND_ATMEL_SOC_SSC
 
 config SND_AT91_SOC_SAM9G20_WM8731
 	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
-	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC && \
+                   AT91_PROGRAMMABLE_CLOCKS
 	select SND_ATMEL_SOC_SSC
 	select SND_SOC_WM8731
 	help
@@ -25,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731
 
 config SND_AT32_SOC_PLAYPAQ
         tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ && AT91_PROGRAMMABLE_CLOCKS
         select SND_ATMEL_SOC_SSC
         select SND_SOC_WM8510
         help
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 293569dfd0e..e521ada8054 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -222,9 +222,9 @@ static int __init at91sam9g20ek_init(void)
 	}
 
 	pllb = clk_get(NULL, "pllb");
-	if (IS_ERR(mclk)) {
+	if (IS_ERR(pllb)) {
 		printk(KERN_ERR "ASoC: Failed to get PLLB\n");
-		ret = PTR_ERR(mclk);
+		ret = PTR_ERR(pllb);
 		goto err_mclk;
 	}
 	ret = clk_set_parent(mclk, pllb);
@@ -240,6 +240,7 @@ static int __init at91sam9g20ek_init(void)
 	if (!at91sam9g20ek_snd_device) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
 		ret = -ENOMEM;
+		goto err_mclk;
 	}
 
 	platform_set_drvdata(at91sam9g20ek_snd_device,
@@ -248,11 +249,13 @@ static int __init at91sam9g20ek_init(void)
 	ret = platform_device_add(at91sam9g20ek_snd_device);
 	if (ret) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
-		platform_device_put(at91sam9g20ek_snd_device);
+		goto err_device_add;
 	}
 
 	return ret;
 
+err_device_add:
+	platform_device_put(at91sam9g20ek_snd_device);
 err_mclk:
 	clk_put(mclk);
 	mclk = NULL;
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index e3d283561c1..86e0f8586dc 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -167,7 +167,6 @@ static int __init afeb9260_soc_init(void)
 
 	return 0;
 err1:
-	platform_device_del(afeb9260_snd_device);
 	platform_device_put(afeb9260_snd_device);
 	return err;
 }
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index bc22ee93a75..d63e28773eb 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -28,6 +28,11 @@
 #include <sound/max98088.h>
 #include "max98088.h"
 
+enum max98088_type {
+       MAX98088,
+       MAX98089,
+};
+
 struct max98088_cdata {
        unsigned int rate;
        unsigned int fmt;
@@ -36,6 +41,7 @@ struct max98088_cdata {
 
 struct max98088_priv {
        u8 reg_cache[M98088_REG_CNT];
+       enum max98088_type devtype;
        void *control_data;
        struct max98088_pdata *pdata;
        unsigned int sysclk;
@@ -2013,7 +2019,10 @@ err_access:
 
 static int max98088_remove(struct snd_soc_codec *codec)
 {
+       struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
+
        max98088_set_bias_level(codec, SND_SOC_BIAS_OFF);
+       kfree(max98088->eq_texts);
 
        return 0;
 }
@@ -2040,6 +2049,8 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        if (max98088 == NULL)
                return -ENOMEM;
 
+       max98088->devtype = id->driver_data;
+
        i2c_set_clientdata(i2c, max98088);
        max98088->control_data = i2c;
        max98088->pdata = i2c->dev.platform_data;
@@ -2059,7 +2070,8 @@ static int __devexit max98088_i2c_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id max98088_i2c_id[] = {
-       { "max98088", 0 },
+       { "max98088", MAX98088 },
+       { "max98089", MAX98089 },
        { }
 };
 MODULE_DEVICE_TABLE(i2c, max98088_i2c_id);
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index 00d67cc8e20..061f9e5a497 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -383,6 +383,7 @@ static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
 	.reg_cache_size = sizeof(stac9766_reg),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_step = 2,
+	.reg_cache_default = stac9766_reg,
 };
 
 static __devinit int stac9766_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index fc687790188..77b8f9ae29b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1176,7 +1176,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_gpio);
 int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
 {
 	u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
-	u8 val, bit = gpio ? 2: 1;
+	u8 val = 0, bit = gpio ? 2 : 1;
 
 	aic3x_read(codec, reg, &val);
 	return (val >> bit) & 1;
@@ -1204,7 +1204,7 @@ EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
 
 int aic3x_headset_detected(struct snd_soc_codec *codec)
 {
-	u8 val;
+	u8 val = 0;
 	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
 	return (val >> 4) & 1;
 }
@@ -1212,7 +1212,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 
 int aic3x_button_pressed(struct snd_soc_codec *codec)
 {
-	u8 val;
+	u8 val = 0;
 	aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
 	return (val >> 5) & 1;
 }
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index ee4fb201de6..d2c24309567 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -78,8 +78,10 @@ static int tpa6130a2_i2c_write(int reg, u8 value)
 
 	if (data->power_state) {
 		val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value);
-		if (val < 0)
+		if (val < 0) {
 			dev_err(&tpa6130a2_client->dev, "Write failed\n");
+			return val;
+		}
 	}
 
 	/* Either powered on or off, we save the context */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 7540a509a6f..464f0cfa4c7 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -597,6 +597,7 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = {
 	.resume =       uda134x_soc_resume,
 	.reg_cache_size = sizeof(uda134x_reg),
 	.reg_word_size = sizeof(u8),
+	.reg_cache_default = uda134x_reg,
 	.reg_cache_step = 1,
 	.read = uda134x_read_reg_cache,
 	.write = uda134x_write,
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index f4f1fba38eb..b3e9fac172e 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -831,7 +831,7 @@ static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	}
 
 	/* MCLK direction */
-	if (dir == WM8350_MCLK_DIR_OUT)
+	if (dir == SND_SOC_CLOCK_OUT)
 		wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
 				WM8350_MCLK_DIR);
 	else
@@ -1586,6 +1586,13 @@ static  int wm8350_codec_probe(struct snd_soc_codec *codec)
 	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
 			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
 
+	/* Make sure AIF tristating is disabled by default */
+	wm8350_clear_bits(wm8350, WM8350_AI_FORMATING, WM8350_AIF_TRI);
+
+	/* Make sure we've got a sane companding setup too */
+	wm8350_clear_bits(wm8350, WM8350_ADC_DAC_COMP,
+			  WM8350_DAC_COMP | WM8350_LOOPBACK);
+
 	/* Make sure jack detect is disabled to start off with */
 	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
@@ -1619,7 +1626,6 @@ static int  wm8350_codec_remove(struct snd_soc_codec *codec)
 {
 	struct wm8350_data *priv = snd_soc_codec_get_drvdata(codec);
 	struct wm8350 *wm8350 = dev_get_platdata(codec->dev);
-	int ret;
 
 	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
 			  WM8350_JDL_ENA | WM8350_JDR_ENA);
@@ -1634,15 +1640,9 @@ static int  wm8350_codec_remove(struct snd_soc_codec *codec)
 	priv->hpr.jack = NULL;
 	priv->mic.jack = NULL;
 
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(&codec->delayed_work);
-
 	/* if there was any work waiting then we run it now and
 	 * wait for its completion */
-	if (ret) {
-		schedule_delayed_work(&codec->delayed_work, 0);
-		flush_scheduled_work();
-	}
+	flush_delayed_work_sync(&codec->delayed_work);
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c
index 712ef7c76f9..9a433a5396c 100644
--- a/sound/soc/codecs/wm8523.c
+++ b/sound/soc/codecs/wm8523.c
@@ -146,7 +146,6 @@ static int wm8523_startup(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	return 0;
 	snd_pcm_hw_constraint_list(substream->runtime, 0,
 				   SNDRV_PCM_HW_PARAM_RATE,
 				   &wm8523->rate_constraint);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 631385802eb..e725c09a3e7 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -526,7 +526,7 @@ static int wm8731_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0);
 
 	/* Disable bypass path by default */
-	snd_soc_update_bits(codec, WM8731_APANA, 0x4, 0);
+	snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0);
 
 	snd_soc_add_controls(codec, wm8731_snd_controls,
 			     ARRAY_SIZE(wm8731_snd_controls));
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8f679a13f2b..84a23675cba 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1526,25 +1526,6 @@ static int wm8753_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
-/*
- * 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;
-}
-
 static int wm8753_probe(struct snd_soc_codec *codec)
 {
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
@@ -1604,7 +1585,7 @@ static int wm8753_probe(struct snd_soc_codec *codec)
 /* power down chip */
 static int wm8753_remove(struct snd_soc_codec *codec)
 {
-	run_delayed_work(&codec->delayed_work);
+	flush_delayed_work_sync(&codec->delayed_work);
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 04182c464e3..0132a27140a 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -34,7 +34,6 @@
 /* codec private data */
 struct wm8776_priv {
 	enum snd_soc_control_type control_type;
-	u16 reg_cache[WM8776_CACHEREGNUM];
 	int sysclk[2];
 };
 
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 33be84e506e..fca60a0b57b 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -2498,6 +2498,8 @@ static int wm8904_remove(struct snd_soc_codec *codec)
 
 	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+	kfree(wm8904->retune_mobile_texts);
+	kfree(wm8904->drc_texts);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 4f326f60410..8340485c985 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -711,7 +711,7 @@ static int wm8961_hw_params(struct snd_pcm_substream *substream,
 	if (fs <= 24000)
 		reg |= WM8961_DACSLOPE;
 	else
-		reg &= WM8961_DACSLOPE;
+		reg &= ~WM8961_DACSLOPE;
 	snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
 
 	return 0;
@@ -736,7 +736,7 @@ static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 		freq /= 2;
 	} else {
 		dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq);
-		reg &= WM8961_MCLKDIV;
+		reg &= ~WM8961_MCLKDIV;
 	}
 
 	snd_soc_write(codec, WM8961_CLOCKING1, reg);
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 894d0cd3aa9..e8092745a20 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3500,8 +3500,11 @@ static ssize_t wm8962_beep_set(struct device *dev,
 {
 	struct wm8962_priv *wm8962 = dev_get_drvdata(dev);
 	long int time;
+	int ret;
 
-	strict_strtol(buf, 10, &time);
+	ret = strict_strtol(buf, 10, &time);
+	if (ret != 0)
+		return ret;
 
 	input_event(wm8962->beep, EV_SND, SND_TONE, time);
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0db59c3aa5d..4d3e6f1ac58 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -3903,6 +3903,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		return -ENOMEM;
 	snd_soc_codec_set_drvdata(codec, wm8994);
 
+	codec->reg_cache = &wm8994->reg_cache;
+
 	wm8994->pdata = dev_get_platdata(codec->dev->parent);
 	wm8994->codec = codec;
 
@@ -4059,6 +4061,8 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
+	kfree(wm8994->retune_mobile_texts);
+	kfree(wm8994->drc_texts);
 	kfree(wm8994);
 
 	return 0;
@@ -4071,6 +4075,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = {
 	.resume =	wm8994_resume,
 	.read = wm8994_read,
 	.write = wm8994_write,
+	.readable_register = wm8994_readable,
+	.volatile_register = wm8994_volatile,
 	.set_bias_level = wm8994_set_bias_level,
 };
 
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 2b07b17a6b2..bc9e6b0b3f6 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -157,12 +157,23 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 }
 
 /* davinci-evm digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link evm_dai = {
+static struct snd_soc_dai_link dm6446_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-mcasp.0",
+	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "tlv320aic3x-hifi",
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-001b",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link dm355_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcbsp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-001b",
 	.platform_name = "davinci-pcm-audio",
 	.init = evm_aic3x_init,
 	.ops = &evm_ops,
@@ -172,10 +183,10 @@ static struct snd_soc_dai_link dm365_evm_dai = {
 #ifdef CONFIG_SND_DM365_AIC3X_CODEC
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
-	.cpu_dai_name = "davinci-i2s",
+	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "tlv320aic3x-hifi",
 	.init = evm_aic3x_init,
-	.codec_name = "tlv320aic3x-codec.0-001a",
+	.codec_name = "tlv320aic3x-codec.1-0018",
 	.ops = &evm_ops,
 #elif defined(CONFIG_SND_DM365_VOICE_CODEC)
 	.name = "Voice Codec - CQ93VC",
@@ -219,10 +230,17 @@ static struct snd_soc_dai_link da8xx_evm_dai = {
 	.ops = &evm_ops,
 };
 
-/* davinci dm6446, dm355 evm audio machine driver */
-static struct snd_soc_card snd_soc_card_evm = {
-	.name = "DaVinci EVM",
-	.dai_link = &evm_dai,
+/* davinci dm6446 evm audio machine driver */
+static struct snd_soc_card dm6446_snd_soc_card_evm = {
+	.name = "DaVinci DM6446 EVM",
+	.dai_link = &dm6446_evm_dai,
+	.num_links = 1,
+};
+
+/* davinci dm355 evm audio machine driver */
+static struct snd_soc_card dm355_snd_soc_card_evm = {
+	.name = "DaVinci DM355 EVM",
+	.dai_link = &dm355_evm_dai,
 	.num_links = 1,
 };
 
@@ -261,10 +279,10 @@ static int __init evm_init(void)
 	int ret;
 
 	if (machine_is_davinci_evm()) {
-		evm_snd_dev_data = &snd_soc_card_evm;
+		evm_snd_dev_data = &dm6446_snd_soc_card_evm;
 		index = 0;
 	} else if (machine_is_davinci_dm355_evm()) {
-		evm_snd_dev_data = &snd_soc_card_evm;
+		evm_snd_dev_data = &dm355_snd_soc_card_evm;
 		index = 1;
 	} else if (machine_is_davinci_dm365_evm()) {
 		evm_snd_dev_data = &dm365_snd_soc_card_evm;
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index d46b545d41f..9e0e565e6ed 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -426,9 +426,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
 	snd_pcm_format_t fmt;
 	unsigned element_cnt = 1;
 
-	dai->capture_dma_data = dev->dma_params;
-	dai->playback_dma_data = dev->dma_params;
-
 	/* general line settings */
 	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -601,6 +598,15 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int davinci_i2s_startup(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	return 0;
+}
+
 static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 		struct snd_soc_dai *dai)
 {
@@ -612,6 +618,7 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
 static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.startup	= davinci_i2s_startup,
 	.shutdown	= davinci_i2s_shutdown,
 	.prepare	= davinci_i2s_prepare,
 	.trigger	= davinci_i2s_trigger,
@@ -749,7 +756,7 @@ static struct platform_driver davinci_mcbsp_driver = {
 	.probe		= davinci_i2s_probe,
 	.remove		= davinci_i2s_remove,
 	.driver		= {
-		.name	= "davinci-i2s",
+		.name	= "davinci-mcbsp",
 		.owner	= THIS_MODULE,
 	},
 };
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 86918ee1241..fb55d2c5d70 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -715,9 +715,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
 	int word_length;
 	u8 fifo_level;
 
-	cpu_dai->capture_dma_data = dev->dma_params;
-	cpu_dai->playback_dma_data = dev->dma_params;
-
 	davinci_hw_common_param(dev, substream->stream);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		fifo_level = dev->txnumevt;
@@ -799,7 +796,17 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct davinci_audio_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	return 0;
+}
+
 static struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+	.startup	= davinci_mcasp_startup,
 	.trigger	= davinci_mcasp_trigger,
 	.hw_params	= davinci_mcasp_hw_params,
 	.set_fmt	= davinci_mcasp_set_dai_fmt,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 009b6521a1b..6c6666a1f94 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -84,7 +84,7 @@ static struct snd_soc_ops sffsdr_ops = {
 static struct snd_soc_dai_link sffsdr_dai = {
 	.name = "PCM3008", /* Codec name */
 	.stream_name = "PCM3008 HiFi",
-	.cpu_dai_name = "davinci-asp.0",
+	.cpu_dai_name = "davinci-mcbsp",
 	.codec_dai_name = "pcm3008-hifi",
 	.codec_name = "pcm3008-codec",
 	.platform_name = "davinci-pcm-audio",
diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c
index ea232f6a2c2..9d2afccc3a2 100644
--- a/sound/soc/davinci/davinci-vcif.c
+++ b/sound/soc/davinci/davinci-vcif.c
@@ -97,9 +97,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream,
 			&davinci_vcif_dev->dma_params[substream->stream];
 	u32 w;
 
-	dai->capture_dma_data = davinci_vcif_dev->dma_params;
-	dai->playback_dma_data = davinci_vcif_dev->dma_params;
-
 	/* Restart the codec before setup */
 	davinci_vcif_stop(substream);
 	davinci_vcif_start(substream);
@@ -174,9 +171,19 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd,
 	return ret;
 }
 
+static int davinci_vcif_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	snd_soc_dai_set_dma_data(dai, substream, dev->dma_params);
+	return 0;
+}
+
 #define DAVINCI_VCIF_RATES	SNDRV_PCM_RATE_8000_48000
 
 static struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+	.startup	= davinci_vcif_startup,
 	.trigger	= davinci_vcif_trigger,
 	.hw_params	= davinci_vcif_hw_params,
 };
@@ -240,7 +247,10 @@ fail:
 
 static int davinci_vcif_remove(struct platform_device *pdev)
 {
+	struct davinci_vcif_dev *davinci_vcif_dev = dev_get_drvdata(&pdev->dev);
+
 	snd_soc_unregister_dai(&pdev->dev);
+	kfree(davinci_vcif_dev);
 
 	return 0;
 }
diff --git a/sound/soc/ep93xx/simone.c b/sound/soc/ep93xx/simone.c
index 4b0d1991372..286817946c5 100644
--- a/sound/soc/ep93xx/simone.c
+++ b/sound/soc/ep93xx/simone.c
@@ -54,24 +54,26 @@ static int __init simone_init(void)
 
 	ret = platform_device_add(simone_snd_ac97_device);
 	if (ret)
-		goto fail;
+		goto fail1;
 
 	simone_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!simone_snd_device) {
 		ret = -ENOMEM;
-		goto fail;
+		goto fail2;
 	}
 
 	platform_set_drvdata(simone_snd_device, &snd_soc_simone);
 	ret = platform_device_add(simone_snd_device);
-	if (ret) {
-		platform_device_put(simone_snd_device);
-		goto fail;
-	}
+	if (ret)
+		goto fail3;
 
-	return ret;
+	return 0;
 
-fail:
+fail3:
+	platform_device_put(simone_snd_device);
+fail2:
+	platform_device_del(simone_snd_ac97_device);
+fail1:
 	platform_device_put(simone_snd_ac97_device);
 	return ret;
 }
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 53251e6b5bd..108b5d8bd0e 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -76,6 +76,7 @@ static __init int efika_fabric_init(void)
 	rc = platform_device_add(pdev);
 	if (rc) {
 		pr_err("efika_fabric_init: platform_device_add() failed\n");
+		platform_device_put(pdev);
 		return -ENODEV;
 	}
 	return 0;
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index dce6b551cd7..f92dca07cd3 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -9,7 +9,6 @@
 #include <linux/module.h>
 #include <linux/of_device.h>
 #include <linux/slab.h>
-#include <linux/of_device.h>
 #include <linux/of_platform.h>
 
 #include <sound/soc.h>
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 74ffed41340..9018fa5bf0d 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -160,7 +160,7 @@ static int __devinit psc_i2s_of_probe(struct platform_device *op,
 	rc = snd_soc_register_dais(&op->dev, psc_i2s_dai, ARRAY_SIZE(psc_i2s_dai));
 	if (rc != 0) {
 		pr_err("Failed to register DAI\n");
-		return 0;
+		return rc;
 	}
 
 	psc_dma = dev_get_drvdata(&op->dev);
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 0d7dcf1e486..7d7847a1e66 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -498,6 +498,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "platform device add failed\n");
 		goto error;
 	}
+	dev_set_drvdata(&pdev->dev, sound_device);
 
 	of_node_put(codec_np);
 
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
index 63b9eaa1ebc..026b756961e 100644
--- a/sound/soc/fsl/p1022_ds.c
+++ b/sound/soc/fsl/p1022_ds.c
@@ -498,6 +498,7 @@ static int p1022_ds_probe(struct platform_device *pdev)
 		dev_err(&pdev->dev, "platform device add failed\n");
 		goto error;
 	}
+	dev_set_drvdata(&pdev->dev, sound_device);
 
 	of_node_put(codec_np);
 
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 25f27ec1dd6..ba4d85e317e 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -76,6 +76,7 @@ static __init int pcm030_fabric_init(void)
 	rc = platform_device_add(pdev);
 	if (rc) {
 		pr_err("pcm030_fabric_init: platform_device_add() failed\n");
+		platform_device_put(pdev);
 		return -ENODEV;
 	}
 	return 0;
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index b59675257ce..dd4fffdbd17 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -34,8 +34,8 @@ static int eukrea_tlv320_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_dai *codec_dai = rtd->dai->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int ret;
 
 	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
@@ -79,10 +79,10 @@ static struct snd_soc_ops eukrea_tlv320_snd_ops = {
 static struct snd_soc_dai_link eukrea_tlv320_dai = {
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
-	.codec_dai	= "tlv320aic23-hifi",
+	.codec_dai_name	= "tlv320aic23-hifi",
 	.platform_name	= "imx-pcm-audio.0",
 	.codec_name	= "tlv320aic23-codec.0-001a",
-	.cpu_dai = "imx-ssi.0",
+	.cpu_dai_name	= "imx-ssi.0",
 	.ops		= &eukrea_tlv320_snd_ops,
 };
 
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index fd493ee1428..671ef8dd524 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -20,6 +20,7 @@
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
+#include <linux/dmaengine.h>
 
 #include <sound/core.h>
 #include <sound/initval.h>
@@ -27,165 +28,146 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
-#include <mach/dma-mx1-mx2.h>
+#include <mach/dma.h>
 
 #include "imx-ssi.h"
 
 struct imx_pcm_runtime_data {
-	int sg_count;
-	struct scatterlist *sg_list;
-	int period;
+	int period_bytes;
 	int periods;
-	unsigned long dma_addr;
 	int dma;
-	struct snd_pcm_substream *substream;
 	unsigned long offset;
 	unsigned long size;
-	unsigned long period_cnt;
 	void *buf;
 	int period_time;
+	struct dma_async_tx_descriptor *desc;
+	struct dma_chan *dma_chan;
+	struct imx_dma_data dma_data;
 };
 
-/* Called by the DMA framework when a period has elapsed */
-static void imx_ssi_dma_progression(int channel, void *data,
-					struct scatterlist *sg)
+static void audio_dma_irq(void *data)
 {
-	struct snd_pcm_substream *substream = data;
+	struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
-	if (!sg)
-		return;
-
-	runtime = iprtd->substream->runtime;
+	iprtd->offset += iprtd->period_bytes;
+	iprtd->offset %= iprtd->period_bytes * iprtd->periods;
 
-	iprtd->offset = sg->dma_address - runtime->dma_addr;
-
-	snd_pcm_period_elapsed(iprtd->substream);
+	snd_pcm_period_elapsed(substream);
 }
 
-static void imx_ssi_dma_callback(int channel, void *data)
+static bool filter(struct dma_chan *chan, void *param)
 {
-	pr_err("%s shouldn't be called\n", __func__);
-}
+	struct imx_pcm_runtime_data *iprtd = param;
 
-static void snd_imx_dma_err_callback(int channel, void *data, int err)
-{
-	struct snd_pcm_substream *substream = data;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = 
-		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
-	int ret;
+	if (!imx_dma_is_general_purpose(chan))
+		return false;
 
-	pr_err("DMA timeout on channel %d -%s%s%s%s\n",
-		 channel,
-		 err & IMX_DMA_ERR_BURST ?    " burst" : "",
-		 err & IMX_DMA_ERR_REQUEST ?  " request" : "",
-		 err & IMX_DMA_ERR_TRANSFER ? " transfer" : "",
-		 err & IMX_DMA_ERR_BUFFER ?   " buffer" : "");
+        chan->private = &iprtd->dma_data;
 
-	imx_dma_disable(iprtd->dma);
-	ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
-			IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
-			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			DMA_MODE_WRITE : DMA_MODE_READ);
-	if (!ret)
-		imx_dma_enable(iprtd->dma);
+        return true;
 }
 
-static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
+static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct imx_pcm_dma_params *dma_params;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+	struct dma_slave_config slave_config;
+	dma_cap_mask_t mask;
+	enum dma_slave_buswidth buswidth;
 	int ret;
 
 	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
-	iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
-	if (iprtd->dma < 0) {
-		pr_err("Failed to claim the audio DMA\n");
-		return -ENODEV;
-	}
+	iprtd->dma_data.peripheral_type = IMX_DMATYPE_SSI;
+	iprtd->dma_data.priority = DMA_PRIO_HIGH;
+	iprtd->dma_data.dma_request = dma_params->dma;
 
-	ret = imx_dma_setup_handlers(iprtd->dma,
-				imx_ssi_dma_callback,
-				snd_imx_dma_err_callback, substream);
-	if (ret)
-		goto out;
+	/* Try to grab a DMA channel */
+	dma_cap_zero(mask);
+	dma_cap_set(DMA_SLAVE, mask);
+	iprtd->dma_chan = dma_request_channel(mask, filter, iprtd);
+	if (!iprtd->dma_chan)
+		return -EINVAL;
 
-	ret = imx_dma_setup_progression_handler(iprtd->dma,
-			imx_ssi_dma_progression);
-	if (ret) {
-		pr_err("Failed to setup the DMA handler\n");
-		goto out;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+	case SNDRV_PCM_FORMAT_S24_LE:
+		buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES;
+		break;
+	default:
+		return 0;
 	}
 
-	ret = imx_dma_config_channel(iprtd->dma,
-			IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO,
-			IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR,
-			dma_params->dma, 1);
-	if (ret < 0) {
-		pr_err("Cannot configure DMA channel: %d\n", ret);
-		goto out;
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		slave_config.direction = DMA_TO_DEVICE;
+		slave_config.dst_addr = dma_params->dma_addr;
+		slave_config.dst_addr_width = buswidth;
+		slave_config.dst_maxburst = dma_params->burstsize;
+	} else {
+		slave_config.direction = DMA_FROM_DEVICE;
+		slave_config.src_addr = dma_params->dma_addr;
+		slave_config.src_addr_width = buswidth;
+		slave_config.src_maxburst = dma_params->burstsize;
 	}
 
-	imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2);
+	ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config);
+	if (ret)
+		return ret;
 
 	return 0;
-out:
-	imx_dma_free(iprtd->dma);
-	return ret;
 }
 
 static int snd_imx_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_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
-	int i;
 	unsigned long dma_addr;
+	struct dma_chan *chan;
+	struct imx_pcm_dma_params *dma_params;
+	int ret;
 
-	imx_ssi_dma_alloc(substream);
+	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	ret = imx_ssi_dma_alloc(substream, params);
+	if (ret)
+		return ret;
+	chan = iprtd->dma_chan;
 
 	iprtd->size = params_buffer_bytes(params);
 	iprtd->periods = params_periods(params);
-	iprtd->period = params_period_bytes(params);
+	iprtd->period_bytes = params_period_bytes(params);
 	iprtd->offset = 0;
 	iprtd->period_time = HZ / (params_rate(params) /
 			params_period_size(params));
 
 	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
 
-	if (iprtd->sg_count != iprtd->periods) {
-		kfree(iprtd->sg_list);
-
-		iprtd->sg_list = kcalloc(iprtd->periods + 1,
-				sizeof(struct scatterlist), GFP_KERNEL);
-		if (!iprtd->sg_list)
-			return -ENOMEM;
-		iprtd->sg_count = iprtd->periods + 1;
-	}
-
-	sg_init_table(iprtd->sg_list, iprtd->sg_count);
 	dma_addr = runtime->dma_addr;
 
-	for (i = 0; i < iprtd->periods; i++) {
-		iprtd->sg_list[i].page_link = 0;
-		iprtd->sg_list[i].offset = 0;
-		iprtd->sg_list[i].dma_address = dma_addr;
-		iprtd->sg_list[i].length = iprtd->period;
-		dma_addr += iprtd->period;
+	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
+
+	iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr,
+			iprtd->period_bytes * iprtd->periods,
+			iprtd->period_bytes,
+			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+			DMA_TO_DEVICE : DMA_FROM_DEVICE);
+	if (!iprtd->desc) {
+		dev_err(&chan->dev->device, "cannot prepare slave dma\n");
+		return -EINVAL;
 	}
 
-	/* close the loop */
-	iprtd->sg_list[iprtd->sg_count - 1].offset = 0;
-	iprtd->sg_list[iprtd->sg_count - 1].length = 0;
-	iprtd->sg_list[iprtd->sg_count - 1].page_link =
-			((unsigned long) iprtd->sg_list | 0x01) & ~0x02;
+	iprtd->desc->callback = audio_dma_irq;
+	iprtd->desc->callback_param = substream;
+
 	return 0;
 }
 
@@ -194,41 +176,21 @@ static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
-	if (iprtd->dma >= 0) {
-		imx_dma_free(iprtd->dma);
-		iprtd->dma = -EINVAL;
+	if (iprtd->dma_chan) {
+		dma_release_channel(iprtd->dma_chan);
+		iprtd->dma_chan = NULL;
 	}
 
-	kfree(iprtd->sg_list);
-	iprtd->sg_list = NULL;
-
 	return 0;
 }
 
 static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct imx_pcm_dma_params *dma_params;
-	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
-	int err;
 
 	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
 
-	iprtd->substream = substream;
-	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
-	iprtd->period_cnt = 0;
-
-	pr_debug("%s: buf: %p period: %d periods: %d\n",
-			__func__, iprtd->buf, iprtd->period, iprtd->periods);
-
-	err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count,
-			IMX_DMA_LENGTH_LOOP, dma_params->dma_addr,
-			substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
-			DMA_MODE_WRITE : DMA_MODE_READ);
-	if (err)
-		return err;
-
 	return 0;
 }
 
@@ -241,14 +203,14 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		imx_dma_enable(iprtd->dma);
+		dmaengine_submit(iprtd->desc);
 
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		imx_dma_disable(iprtd->dma);
+		dmaengine_terminate_all(iprtd->dma_chan);
 
 		break;
 	default:
@@ -263,6 +225,9 @@ static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 
+	pr_debug("%s: %ld %ld\n", __func__, iprtd->offset,
+			bytes_to_frames(substream->runtime, iprtd->offset));
+
 	return bytes_to_frames(substream->runtime, iprtd->offset);
 }
 
@@ -279,7 +244,7 @@ static struct snd_pcm_hardware snd_imx_hardware = {
 	.channels_max = 2,
 	.buffer_bytes_max = IMX_SSI_DMABUF_SIZE,
 	.period_bytes_min = 128,
-	.period_bytes_max = 16 * 1024,
+	.period_bytes_max = 65535, /* Limited by SDMA engine */
 	.periods_min = 2,
 	.periods_max = 255,
 	.fifo_size = 0,
@@ -304,11 +269,23 @@ static int snd_imx_open(struct snd_pcm_substream *substream)
 	}
 
 	snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
+
+	return 0;
+}
+
+static int snd_imx_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
+
+	kfree(iprtd);
+
 	return 0;
 }
 
 static struct snd_pcm_ops imx_pcm_ops = {
 	.open		= snd_imx_open,
+	.close		= snd_imx_close,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= snd_imx_pcm_hw_params,
 	.hw_free	= snd_imx_pcm_hw_free,
@@ -340,7 +317,6 @@ static struct platform_driver imx_pcm_driver = {
 			.name = "imx-pcm-audio",
 			.owner = THIS_MODULE,
 	},
-
 	.probe = imx_soc_platform_probe,
 	.remove = __devexit_p(imx_soc_platform_remove),
 };
@@ -356,4 +332,3 @@ static void __exit snd_imx_pcm_exit(void)
 	platform_driver_unregister(&imx_pcm_driver);
 }
 module_exit(snd_imx_pcm_exit);
-
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index d4bd345b0a8..390b6ffc265 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -439,7 +439,22 @@ void imx_pcm_free(struct snd_pcm *pcm)
 }
 EXPORT_SYMBOL_GPL(imx_pcm_free);
 
+static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
+{
+	struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
+	uint32_t val;
+
+	snd_soc_dai_set_drvdata(dai, ssi);
+
+	val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
+		SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
+	writel(val, ssi->base + SSI_SFCSR);
+
+	return 0;
+}
+
 static struct snd_soc_dai_driver imx_ssi_dai = {
+	.probe = imx_ssi_dai_probe,
 	.playback = {
 		.channels_min = 2,
 		.channels_max = 2,
@@ -455,20 +470,6 @@ static struct snd_soc_dai_driver imx_ssi_dai = {
 	.ops = &imx_ssi_pcm_dai_ops,
 };
 
-static int imx_ssi_dai_probe(struct snd_soc_dai *dai)
-{
-	struct imx_ssi *ssi = dev_get_drvdata(dai->dev);
-	uint32_t val;
-
-	snd_soc_dai_set_drvdata(dai, ssi);
-
-	val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) |
-		SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize);
-	writel(val, ssi->base + SSI_SFCSR);
-
-	return 0;
-}
-
 static struct snd_soc_dai_driver imx_ac97_dai = {
 	.probe = imx_ssi_dai_probe,
 	.ac97_control = 1,
@@ -677,9 +678,25 @@ static int imx_ssi_probe(struct platform_device *pdev)
 		goto failed_register;
 	}
 
-	ssi->soc_platform_pdev = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
-	if (!ssi->soc_platform_pdev)
+	ssi->soc_platform_pdev_fiq = platform_device_alloc("imx-fiq-pcm-audio", pdev->id);
+	if (!ssi->soc_platform_pdev_fiq) {
+		ret = -ENOMEM;
+		goto failed_pdev_fiq_alloc;
+	}
+
+	platform_set_drvdata(ssi->soc_platform_pdev_fiq, ssi);
+	ret = platform_device_add(ssi->soc_platform_pdev_fiq);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to add platform device\n");
+		goto failed_pdev_fiq_add;
+	}
+
+	ssi->soc_platform_pdev = platform_device_alloc("imx-pcm-audio", pdev->id);
+	if (!ssi->soc_platform_pdev) {
+		ret = -ENOMEM;
 		goto failed_pdev_alloc;
+	}
+
 	platform_set_drvdata(ssi->soc_platform_pdev, ssi);
 	ret = platform_device_add(ssi->soc_platform_pdev);
 	if (ret) {
@@ -692,6 +709,10 @@ static int imx_ssi_probe(struct platform_device *pdev)
 failed_pdev_add:
 	platform_device_put(ssi->soc_platform_pdev);
 failed_pdev_alloc:
+	platform_device_del(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_add:
+	platform_device_put(ssi->soc_platform_pdev_fiq);
+failed_pdev_fiq_alloc:
 	snd_soc_unregister_dai(&pdev->dev);
 failed_register:
 failed_ac97:
@@ -712,8 +733,8 @@ static int __devexit imx_ssi_remove(struct platform_device *pdev)
 	struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	struct imx_ssi *ssi = platform_get_drvdata(pdev);
 
-	platform_device_del(ssi->soc_platform_pdev);
-	platform_device_put(ssi->soc_platform_pdev);
+	platform_device_unregister(ssi->soc_platform_pdev);
+	platform_device_unregister(ssi->soc_platform_pdev_fiq);
 
 	snd_soc_unregister_dai(&pdev->dev);
 
diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h
index 53b780d9b2b..a4406a13489 100644
--- a/sound/soc/imx/imx-ssi.h
+++ b/sound/soc/imx/imx-ssi.h
@@ -185,6 +185,9 @@
 
 #define DRV_NAME "imx-ssi"
 
+#include <linux/dmaengine.h>
+#include <mach/dma.h>
+
 struct imx_pcm_dma_params {
 	int dma;
 	unsigned long dma_addr;
@@ -212,6 +215,7 @@ struct imx_ssi {
 	int enabled;
 
 	struct platform_device *soc_platform_pdev;
+	struct platform_device *soc_platform_pdev_fiq;
 };
 
 struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev,
diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c
index 6a65dd70551..9eabc28667e 100644
--- a/sound/soc/imx/phycore-ac97.c
+++ b/sound/soc/imx/phycore-ac97.c
@@ -20,9 +20,6 @@
 #include <sound/soc-dapm.h>
 #include <asm/mach-types.h>
 
-#include "../codecs/wm9712.h"
-#include "imx-ssi.h"
-
 static struct snd_soc_card imx_phycore;
 
 static struct snd_soc_ops imx_phycore_hifi_ops = {
@@ -41,11 +38,12 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = {
 };
 
 static struct snd_soc_card imx_phycore = {
-	.name		= "PhyCORE-audio",
+	.name		= "PhyCORE-ac97-audio",
 	.dai_link	= imx_phycore_dai_ac97,
 	.num_links	= ARRAY_SIZE(imx_phycore_dai_ac97),
 };
 
+static struct platform_device *imx_phycore_snd_ac97_device;
 static struct platform_device *imx_phycore_snd_device;
 
 static int __init imx_phycore_init(void)
@@ -56,29 +54,42 @@ static int __init imx_phycore_init(void)
 		/* return happy. We might run on a totally different machine */
 		return 0;
 
-	imx_phycore_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!imx_phycore_snd_device)
+	imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1);
+	if (!imx_phycore_snd_ac97_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(imx_phycore_snd_device, &imx_phycore);
-	ret = platform_device_add(imx_phycore_snd_device);
+	platform_set_drvdata(imx_phycore_snd_ac97_device, &imx_phycore);
+	ret = platform_device_add(imx_phycore_snd_ac97_device);
+	if (ret)
+		goto fail1;
 
 	imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1);
-	if (!imx_phycore_snd_device)
-		return -ENOMEM;
+	if (!imx_phycore_snd_device) {
+		ret = -ENOMEM;
+		goto fail2;
+	}
 	ret = platform_device_add(imx_phycore_snd_device);
 
 	if (ret) {
 		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
-		platform_device_put(imx_phycore_snd_device);
+		goto fail3;
 	}
 
+	return 0;
+
+fail3:
+	platform_device_put(imx_phycore_snd_device);
+fail2:
+	platform_device_del(imx_phycore_snd_ac97_device);
+fail1:
+	platform_device_put(imx_phycore_snd_ac97_device);
 	return ret;
 }
 
 static void __exit imx_phycore_exit(void)
 {
 	platform_device_unregister(imx_phycore_snd_device);
+	platform_device_unregister(imx_phycore_snd_ac97_device);
 }
 
 late_initcall(imx_phycore_init);
diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c
index 293dc748797..dac6732da96 100644
--- a/sound/soc/nuc900/nuc900-ac97.c
+++ b/sound/soc/nuc900/nuc900-ac97.c
@@ -49,7 +49,7 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97,
 	mutex_lock(&ac97_mutex);
 
 	val = nuc900_checkready();
-	if (!!val) {
+	if (val) {
 		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
 		goto out;
 	}
@@ -102,7 +102,7 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
 	mutex_lock(&ac97_mutex);
 
 	tmp = nuc900_checkready();
-	if (!!tmp)
+	if (tmp)
 		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
 
 	/* clear the R_WB bit and write register index */
@@ -149,7 +149,7 @@ static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97)
 	udelay(100);
 
 	val = nuc900_checkready();
-	if (!!val)
+	if (val)
 		dev_err(nuc900_audio->dev, "AC97 codec is not ready\n");
 
 	mutex_unlock(&ac97_mutex);
@@ -263,8 +263,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream *substream,
 	return ret;
 }
 
-static int nuc900_ac97_probe(struct platform_device *pdev,
-					struct snd_soc_dai *dai)
+static int nuc900_ac97_probe(struct snd_soc_dai *dai)
 {
 	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
 	unsigned long val;
@@ -284,12 +283,12 @@ static int nuc900_ac97_probe(struct platform_device *pdev,
 	return 0;
 }
 
-static void nuc900_ac97_remove(struct platform_device *pdev,
-						struct snd_soc_dai *dai)
+static int nuc900_ac97_remove(struct snd_soc_dai *dai)
 {
 	struct nuc900_audio *nuc900_audio = nuc900_ac97_data;
 
 	clk_disable(nuc900_audio->clk);
+	return 0;
 }
 
 static struct snd_soc_dai_ops nuc900_ac97_dai_ops = {
@@ -313,7 +312,7 @@ static struct snd_soc_dai_driver nuc900_ac97_dai = {
 		.channels_max	= 2,
 	},
 	.ops = &nuc900_ac97_dai_ops,
-}
+};
 
 static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev)
 {
@@ -384,7 +383,6 @@ out0:
 
 static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
 {
-
 	snd_soc_unregister_dai(&pdev->dev);
 
 	clk_put(nuc900_ac97_data->clk);
@@ -392,6 +390,7 @@ static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev)
 	release_mem_region(nuc900_ac97_data->res->start,
 				resource_size(nuc900_ac97_data->res));
 
+	kfree(nuc900_ac97_data);
 	nuc900_ac97_data = NULL;
 
 	return 0;
diff --git a/sound/soc/nuc900/nuc900-audio.h b/sound/soc/nuc900/nuc900-audio.h
index aeed8ead2b2..59f7e8ed1a6 100644
--- a/sound/soc/nuc900/nuc900-audio.h
+++ b/sound/soc/nuc900/nuc900-audio.h
@@ -110,4 +110,6 @@ struct nuc900_audio {
 
 };
 
+extern struct nuc900_audio *nuc900_ac97_data;
+
 #endif /*end _NUC900_AUDIO_H */
diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c
index 195d1ac9477..8263f56dc66 100644
--- a/sound/soc/nuc900/nuc900-pcm.c
+++ b/sound/soc/nuc900/nuc900-pcm.c
@@ -50,12 +50,12 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream *substream,
 	unsigned long flags;
 	int ret = 0;
 
-	spin_lock_irqsave(&nuc900_audio->lock, flags);
-
 	ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
 	if (ret < 0)
 		return ret;
 
+	spin_lock_irqsave(&nuc900_audio->lock, flags);
+
 	nuc900_audio->substream = substream;
 	nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr;
 	nuc900_audio->buffersize[substream->stream] =
@@ -169,6 +169,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct nuc900_audio *nuc900_audio = runtime->private_data;
 	unsigned long flags, val;
+	int ret = 0;
 
 	spin_lock_irqsave(&nuc900_audio->lock, flags);
 
@@ -197,10 +198,10 @@ static int nuc900_dma_prepare(struct snd_pcm_substream *substream)
 		AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val);
 		break;
 	default:
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 	spin_unlock_irqrestore(&nuc900_audio->lock, flags);
-	return 0;
+	return ret;
 }
 
 static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd)
@@ -332,7 +333,7 @@ static struct snd_soc_platform_driver nuc900_soc_platform = {
 	.ops		= &nuc900_dma_ops,
 	.pcm_new	= nuc900_dma_new,
 	.pcm_free	= nuc900_dma_free_dma_buffers,
-}
+};
 
 static int __devinit nuc900_soc_platform_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index d542ea2ff6b..a088db6d509 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -12,8 +12,8 @@ config SND_OMAP_SOC_MCPDM
 config SND_OMAP_SOC_N810
 	tristate "SoC Audio support for Nokia N810"
 	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
+	depends on OMAP_MUX
 	select SND_OMAP_SOC_MCBSP
-	select OMAP_MUX
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index d211c9fa5a9..7e84f24b9a8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -644,15 +644,23 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 
 	case OMAP_MCBSP_CLKR_SRC_CLKR:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR);
 		break;
 	case OMAP_MCBSP_CLKR_SRC_CLKX:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX);
 		break;
 	case OMAP_MCBSP_FSR_SRC_FSR:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR);
 		break;
 	case OMAP_MCBSP_FSR_SRC_FSX:
+		if (cpu_class_is_omap1())
+			break;
 		omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX);
 		break;
 	default:
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index dbd9d96b5f9..4ee33ce2cb9 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -306,6 +306,7 @@ static int __init omap3pandora_soc_init(void)
 		pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n",
 			dev_name(&omap3pandora_snd_device->dev),
 			PTR_ERR(omap3pandora_dac_reg));
+		ret = PTR_ERR(omap3pandora_dac_reg);
 		goto fail3;
 	}
 
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index f0e66255642..65ae00e976e 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -177,7 +177,8 @@ static int __init osk_soc_init(void)
 	tlv320aic23_mclk = clk_get(dev, "mclk");
 	if (IS_ERR(tlv320aic23_mclk)) {
 		printk(KERN_ERR "Could not get mclk clock\n");
-		return -ENODEV;
+		err = PTR_ERR(tlv320aic23_mclk);
+		goto err2;
 	}
 
 	/*
@@ -188,7 +189,7 @@ static int __init osk_soc_init(void)
 		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
 			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
 			err = -ECANCELED;
-			goto err1;
+			goto err3;
 		}
 	}
 
@@ -196,9 +197,12 @@ static int __init osk_soc_init(void)
 	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
 
 	return 0;
-err1:
+
+err3:
 	clk_put(tlv320aic23_mclk);
+err2:
 	platform_device_del(osk_snd_device);
+err1:
 	platform_device_put(osk_snd_device);
 
 	return err;
@@ -207,6 +211,7 @@ err1:
 
 static void __exit osk_soc_exit(void)
 {
+	clk_put(tlv320aic23_mclk);
 	platform_device_unregister(osk_snd_device);
 }
 
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 37f191bbfdd..580f4857130 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,6 +1,7 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
 	depends on ARCH_PXA
+	select SND_ARM
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 97e9423615c..f451acd4935 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -100,8 +100,13 @@ static int corgi_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	corgi_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index b8207ced407..5ef0526924b 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -72,9 +72,13 @@ static int magician_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	magician_ext_control(codec);
 
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index af84ee9c5e1..84edd0385a2 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -77,8 +77,13 @@ static int poodle_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	poodle_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index f470f360f4d..0b30d7de24e 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -108,8 +108,13 @@ static int spitz_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	spitz_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 73d0edd8ded..7b983f93545 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -81,8 +81,13 @@ static int tosa_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 
+	mutex_lock(&codec->mutex);
+
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
 	return 0;
 }
 
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 8a6b53ccd20..d85bf8a0abb 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -2,6 +2,7 @@ config SND_S3C24XX_SOC
 	tristate "SoC Audio for the Samsung S3CXXXX chips"
 	depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 || ARCH_S5PV210
 	select S3C64XX_DMA if ARCH_S3C64XX
+	select S3C2410_DMA if ARCH_S3C2410
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the S3C24XX AC97 or I2S interfaces. You will also need to
diff --git a/sound/soc/s3c24xx/rx1950_uda1380.c b/sound/soc/s3c24xx/rx1950_uda1380.c
index ffd5cf2fb0a..468cc11fdf4 100644
--- a/sound/soc/s3c24xx/rx1950_uda1380.c
+++ b/sound/soc/s3c24xx/rx1950_uda1380.c
@@ -50,7 +50,6 @@ static unsigned int rates[] = {
 	16000,
 	44100,
 	48000,
-	88200,
 };
 
 static struct snd_pcm_hw_constraint_list hw_rates = {
@@ -130,7 +129,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 };
 
 static struct platform_device *s3c24xx_snd_device;
-static struct clk *xtal;
 
 static int rx1950_startup(struct snd_pcm_substream *substream)
 {
@@ -179,10 +177,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
 	case 44100:
 	case 88200:
 		clk_source = S3C24XX_CLKSRC_MPLL;
-		fs_mode = S3C2410_IISMOD_256FS;
-		div = clk_get_rate(xtal) / (256 * rate);
-		if (clk_get_rate(xtal) % (256 * rate) > (128 * rate))
-			div++;
+		fs_mode = S3C2410_IISMOD_384FS;
+		div = 1;
 		break;
 	default:
 		printk(KERN_ERR "%s: rate %d is not supported\n",
@@ -210,7 +206,7 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream,
 
 	/* set MCLK division for sample rate */
 	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
-		S3C2410_IISMOD_384FS);
+		fs_mode);
 	if (ret < 0)
 		return ret;
 
@@ -295,17 +291,8 @@ static int __init rx1950_init(void)
 		goto err_plat_add;
 	}
 
-	xtal = clk_get(&s3c24xx_snd_device->dev, "xtal");
-
-	if (IS_ERR(xtal)) {
-		ret = PTR_ERR(xtal);
-		platform_device_unregister(s3c24xx_snd_device);
-		goto err_clk;
-	}
-
 	return 0;
 
-err_clk:
 err_plat_add:
 err_plat_alloc:
 err_gpio_conf:
@@ -320,7 +307,6 @@ static void __exit rx1950_exit(void)
 	platform_device_unregister(s3c24xx_snd_device);
 	snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
 		hp_jack_gpios);
-	clk_put(xtal);
 	gpio_free(S3C2410_GPA(1));
 }
 
diff --git a/sound/soc/s3c24xx/smdk_spdif.c b/sound/soc/s3c24xx/smdk_spdif.c
index f31d22ad7c8..c8bd90488a8 100644
--- a/sound/soc/s3c24xx/smdk_spdif.c
+++ b/sound/soc/s3c24xx/smdk_spdif.c
@@ -38,7 +38,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
 	}
 
 	mout_epll = clk_get(NULL, "mout_epll");
-	if (IS_ERR(fout_epll)) {
+	if (IS_ERR(mout_epll)) {
 		printk(KERN_WARNING "%s: Cannot find mout_epll.\n",
 				__func__);
 		ret = -EINVAL;
@@ -54,7 +54,7 @@ static int set_audio_clock_heirachy(struct platform_device *pdev)
 	}
 
 	sclk_spdif = clk_get(NULL, "sclk_spdif");
-	if (IS_ERR(fout_epll)) {
+	if (IS_ERR(sclk_spdif)) {
 		printk(KERN_WARNING "%s: Cannot find sclk_spdif.\n",
 				__func__);
 		ret = -EINVAL;
diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c
index 8778faa174a..3052f64b240 100644
--- a/sound/soc/s6000/s6000-i2s.c
+++ b/sound/soc/s6000/s6000-i2s.c
@@ -434,7 +434,7 @@ static struct snd_soc_dai_driver s6000_i2s_dai = {
 		.rate_max = 1562500,
 	},
 	.ops = &s6000_i2s_dai_ops,
-}
+};
 
 static int __devinit s6000_i2s_probe(struct platform_device *pdev)
 {
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 271fd222bf1..ab3ccaec72d 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -473,7 +473,7 @@ static int s6000_pcm_new(struct snd_card *card,
 	}
 
 	res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED,
-			  s6000_soc_platform.name, pcm);
+			  "s6000-audio", pcm);
 	if (res) {
 		printk(KERN_ERR "s6000-pcm couldn't get IRQ\n");
 		return res;
diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c
index 96c05e13753..c1244c5bc73 100644
--- a/sound/soc/s6000/s6105-ipcam.c
+++ b/sound/soc/s6000/s6105-ipcam.c
@@ -167,7 +167,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd)
 
 	snd_soc_dapm_sync(codec);
 
-	snd_ctl_add(codec->snd_card, snd_ctl_new1(&audio_out_mux, codec));
+	snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec));
 
 	return 0;
 }
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 507e709f280..4c2404b1b86 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -132,6 +132,8 @@ struct fsi_priv {
 	struct fsi_stream playback;
 	struct fsi_stream capture;
 
+	long rate;
+
 	u32 mst_ctrl;
 };
 
@@ -854,10 +856,17 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	int is_play = fsi_is_play(substream);
+	struct fsi_master *master = fsi_get_master(fsi);
+	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 
 	fsi_irq_disable(fsi, is_play);
 	fsi_clk_ctrl(fsi, 0);
 
+	set_rate = master->info->set_rate;
+	if (set_rate && fsi->rate)
+		set_rate(dai->dev, fsi_is_port_a(fsi), fsi->rate, 0);
+	fsi->rate = 0;
+
 	pm_runtime_put_sync(dai->dev);
 }
 
@@ -891,20 +900,20 @@ static int fsi_dai_hw_params(struct snd_pcm_substream *substream,
 {
 	struct fsi_priv *fsi = fsi_get_priv(substream);
 	struct fsi_master *master = fsi_get_master(fsi);
-	int (*set_rate)(int is_porta, int rate) = master->info->set_rate;
+	int (*set_rate)(struct device *dev, int is_porta, int rate, int enable);
 	int fsi_ver = master->core->ver;
-	int is_play = fsi_is_play(substream);
+	long rate = params_rate(params);
 	int ret;
 
-	/* if slave mode, set_rate is not needed */
-	if (!fsi_is_master_mode(fsi, is_play))
+	set_rate = master->info->set_rate;
+	if (!set_rate)
 		return 0;
 
-	/* it is error if no set_rate */
-	if (!set_rate)
-		return -EIO;
+	ret = set_rate(dai->dev, fsi_is_port_a(fsi), rate, 1);
+	if (ret < 0) /* error */
+		return ret;
 
-	ret = set_rate(fsi_is_port_a(fsi), params_rate(params));
+	fsi->rate = rate;
 	if (ret > 0) {
 		u32 data = 0;
 
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index 40bbdf1591d..05192d97b37 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -387,7 +387,7 @@ static int __devinit sh4_soc_dai_probe(struct platform_device *pdev)
 
 static int __devexit sh4_soc_dai_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_dai(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
+	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(sh4_ssi_dai));
 	return 0;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 614a8b30d87..b54ea9a0a1d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -67,25 +67,6 @@ 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;
-}
-
 /* codec register dump */
 static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 {
@@ -1016,7 +997,7 @@ static int soc_suspend(struct device *dev)
 
 	/* close any waiting streams and save state */
 	for (i = 0; i < card->num_rtd; i++) {
-		run_delayed_work(&card->rtd[i].delayed_work);
+		flush_delayed_work_sync(&card->rtd[i].delayed_work);
 		card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level;
 	}
 
@@ -1687,7 +1668,7 @@ static int soc_remove(struct platform_device *pdev)
 		/* make sure any delayed work runs */
 		for (i = 0; i < card->num_rtd; i++) {
 			struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
-			run_delayed_work(&rtd->delayed_work);
+			flush_delayed_work_sync(&rtd->delayed_work);
 		}
 
 		/* remove and free each DAI */
@@ -1718,7 +1699,7 @@ static int soc_poweroff(struct device *dev)
 	 * now, we're shutting down so no imminent restart. */
 	for (i = 0; i < card->num_rtd; i++) {
 		struct snd_soc_pcm_runtime *rtd = &card->rtd[i];
-		run_delayed_work(&rtd->delayed_work);
+		flush_delayed_work_sync(&rtd->delayed_work);
 	}
 
 	snd_soc_dapm_shutdown(card);
@@ -3043,8 +3024,10 @@ int snd_soc_register_dais(struct device *dev,
 	for (i = 0; i < count; i++) {
 
 		dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL);
-		if (dai == NULL)
-			return -ENOMEM;
+		if (dai == NULL) {
+			ret = -ENOMEM;
+			goto err;
+		}
 
 		/* create DAI component name */
 		dai->name = fmt_multiple_name(dev, &dai_drv[i]);
@@ -3263,9 +3246,6 @@ int snd_soc_register_codec(struct device *dev,
 	return 0;
 
 error:
-	for (i--; i >= 0; i--)
-		snd_soc_unregister_dai(dev);
-
 	if (codec->reg_cache)
 		kfree(codec->reg_cache);
 	kfree(codec->name);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7d85c6496af..75ed6491222 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -683,12 +683,12 @@ static int dapm_seq_compare(struct snd_soc_dapm_widget *a,
 			    struct snd_soc_dapm_widget *b,
 			    int sort[])
 {
-	if (a->codec != b->codec)
-		return (unsigned long)a - (unsigned long)b;
 	if (sort[a->id] != sort[b->id])
 		return sort[a->id] - sort[b->id];
 	if (a->reg != b->reg)
 		return a->reg - b->reg;
+	if (a->codec != b->codec)
+		return (unsigned long)a->codec - (unsigned long)b->codec;
 
 	return 0;
 }
diff --git a/sound/sound_core.c b/sound/sound_core.c
index c03bbaefdbc..5580aced873 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -104,7 +104,6 @@ module_exit(cleanup_soundcore);
 
 #include <linux/init.h>
 #include <linux/slab.h>
-#include <linux/smp_lock.h>
 #include <linux/types.h>
 #include <linux/kernel.h>
 #include <linux/sound.h>