24 files changed, 213 insertions, 103 deletions

diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index 582aace..7eca25f 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -37,8 +37,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
 MODULE_LICENSE("GPL");
 
-#define FREQ_LO		 (76U * 16000)
-#define FREQ_HI		(108U * 16000)
+#define FREQ_LO		((tea->tea5759 ? 760 :  875) * 1600U)
+#define FREQ_HI		((tea->tea5759 ? 910 : 1080) * 1600U)
 
 /*
  * definitions
@@ -120,9 +120,9 @@ static u32 snd_tea575x_read(struct snd_tea575x *tea)
 	return data;
 }
 
-static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
+static u32 snd_tea575x_val_to_freq(struct snd_tea575x *tea, u32 val)
 {
-	u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK;
+	u32 freq = val & TEA575X_BIT_FREQ_MASK;
 
 	if (freq == 0)
 		return freq;
@@ -139,6 +139,11 @@ static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
 	return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */
 }
 
+static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
+{
+	return snd_tea575x_val_to_freq(tea, snd_tea575x_read(tea));
+}
+
 static void snd_tea575x_set_freq(struct snd_tea575x *tea)
 {
 	u32 freq = tea->freq;
@@ -156,6 +161,7 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea)
 	tea->val &= ~TEA575X_BIT_FREQ_MASK;
 	tea->val |= freq & TEA575X_BIT_FREQ_MASK;
 	snd_tea575x_write(tea, tea->val);
+	tea->freq = snd_tea575x_val_to_freq(tea, tea->val);
 }
 
 /*
@@ -317,7 +323,6 @@ static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl)
 }
 
 static const struct v4l2_file_operations tea575x_fops = {
-	.owner		= THIS_MODULE,
 	.unlocked_ioctl	= video_ioctl2,
 	.open           = v4l2_fh_open,
 	.release        = v4l2_fh_release,
@@ -337,7 +342,6 @@ static const struct v4l2_ioctl_ops tea575x_ioctl_ops = {
 };
 
 static const struct video_device tea575x_radio = {
-	.fops           = &tea575x_fops,
 	.ioctl_ops 	= &tea575x_ioctl_ops,
 	.release        = video_device_release_empty,
 };
@@ -349,7 +353,7 @@ static const struct v4l2_ctrl_ops tea575x_ctrl_ops = {
 /*
  * initialize all the tea575x chips
  */
-int snd_tea575x_init(struct snd_tea575x *tea)
+int snd_tea575x_init(struct snd_tea575x *tea, struct module *owner)
 {
 	int retval;
 
@@ -374,6 +378,9 @@ int snd_tea575x_init(struct snd_tea575x *tea)
 	tea->vd.lock = &tea->mutex;
 	tea->vd.v4l2_dev = tea->v4l2_dev;
 	tea->vd.ctrl_handler = &tea->ctrl_handler;
+	tea->fops = tea575x_fops;
+	tea->fops.owner = owner;
+	tea->vd.fops = &tea->fops;
 	set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags);
 	/* disable hw_freq_seek if we can't use it */
 	if (tea->cannot_read_data)
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 67f47d8..52b5c0b 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2769,7 +2769,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
 	chip->tea.ops = &snd_es1968_tea_ops;
 	strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card));
 	sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
-	if (!snd_tea575x_init(&chip->tea))
+	if (!snd_tea575x_init(&chip->tea, THIS_MODULE))
 		printk(KERN_INFO "es1968: detected TEA575x radio\n");
 #endif
 
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index f696623..b32e802 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1254,7 +1254,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
 	if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 &&
 	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
-		if (snd_tea575x_init(&chip->tea)) {
+		if (snd_tea575x_init(&chip->tea, THIS_MODULE)) {
 			snd_printk(KERN_ERR "TEA575x radio not found\n");
 			snd_fm801_free(chip);
 			return -ENODEV;
@@ -1263,7 +1263,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		/* autodetect tuner connection */
 		for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) {
 			chip->tea575x_tuner = tea575x_tuner;
-			if (!snd_tea575x_init(&chip->tea)) {
+			if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) {
 				snd_printk(KERN_INFO "detected TEA575x radio type %s\n",
 					   get_tea575x_gpio(chip)->name);
 				break;
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 163b6b5..d030797 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -97,19 +97,6 @@ config SND_HDA_CODEC_REALTEK
 	  snd-hda-codec-realtek.
 	  This module is automatically loaded at probing.
 
-config SND_HDA_ENABLE_REALTEK_QUIRKS
-	bool "Build static quirks for Realtek codecs"
-	depends on SND_HDA_CODEC_REALTEK
-	default y
-	help
-	  Say Y here to build the static quirks codes for Realtek codecs.
-	  If you need the "model" preset that the default BIOS auto-parser
-	  can't handle, turn this option on.
-
-	  If your device works with model=auto option, basically you don't
-	  need the quirk code.  By turning this off, you can reduce the
-	  module size quite a lot.
-
 config SND_HDA_CODEC_ANALOG
 	bool "Build Analog Device HD-audio codec support"
 	default y
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 6e9ef3e..f7520b9 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -618,7 +618,6 @@ int snd_hda_gen_add_verbs(struct hda_gen_spec *spec,
 			  const struct hda_verb *list)
 {
 	const struct hda_verb **v;
-	snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
 	v = snd_array_new(&spec->verbs);
 	if (!v)
 		return -ENOMEM;
diff --git a/sound/pci/hda/hda_auto_parser.h b/sound/pci/hda/hda_auto_parser.h
index 2a7889d..632ad0a 100644
--- a/sound/pci/hda/hda_auto_parser.h
+++ b/sound/pci/hda/hda_auto_parser.h
@@ -157,4 +157,14 @@ void snd_hda_pick_fixup(struct hda_codec *codec,
 			const struct snd_pci_quirk *quirk,
 			const struct hda_fixup *fixlist);
 
+static inline void snd_hda_gen_init(struct hda_gen_spec *spec)
+{
+	snd_array_init(&spec->verbs, sizeof(struct hda_verb *), 8);
+}
+
+static inline void snd_hda_gen_free(struct hda_gen_spec *spec)
+{
+	snd_array_free(&spec->verbs);
+}
+
 #endif /* __SOUND_HDA_AUTO_PARSER_H */
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 41ca803..51cb2a2 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1184,6 +1184,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 {
 	if (!codec)
 		return;
+	snd_hda_jack_tbl_clear(codec);
 	restore_init_pincfgs(codec);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	cancel_delayed_work(&codec->power_work);
@@ -1192,6 +1193,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	list_del(&codec->list);
 	snd_array_free(&codec->mixers);
 	snd_array_free(&codec->nids);
+	snd_array_free(&codec->cvt_setups);
 	snd_array_free(&codec->conn_lists);
 	snd_array_free(&codec->spdif_out);
 	codec->bus->caddr_tbl[codec->addr] = NULL;
@@ -2333,6 +2335,8 @@ int snd_hda_codec_reset(struct hda_codec *codec)
 	/* free only driver_pins so that init_pins + user_pins are restored */
 	snd_array_free(&codec->driver_pins);
 	restore_pincfgs(codec);
+	snd_array_free(&codec->cvt_setups);
+	snd_array_free(&codec->spdif_out);
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
 	codec->preset = NULL;
@@ -4393,20 +4397,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec)
 	codec->power_jiffies += delta;
 }
 
-/**
- * snd_hda_power_up - Power-up the codec
- * @codec: HD-audio codec
- *
- * Increment the power-up counter and power up the hardware really when
- * not turned on yet.
- */
-void snd_hda_power_up(struct hda_codec *codec)
+/* Transition to powered up, if wait_power_down then wait for a pending
+ * transition to D3 to complete. A pending D3 transition is indicated
+ * with power_transition == -1. */
+static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down)
 {
 	struct hda_bus *bus = codec->bus;
 
 	spin_lock(&codec->power_lock);
 	codec->power_count++;
-	if (codec->power_on || codec->power_transition > 0) {
+	/* Return if power_on or transitioning to power_on, unless currently
+	 * powering down. */
+	if ((codec->power_on || codec->power_transition > 0) &&
+	    !(wait_power_down && codec->power_transition < 0)) {
 		spin_unlock(&codec->power_lock);
 		return;
 	}
@@ -4430,8 +4433,37 @@ void snd_hda_power_up(struct hda_codec *codec)
 	codec->power_transition = 0;
 	spin_unlock(&codec->power_lock);
 }
+
+/**
+ * snd_hda_power_up - Power-up the codec
+ * @codec: HD-audio codec
+ *
+ * Increment the power-up counter and power up the hardware really when
+ * not turned on yet.
+ */
+void snd_hda_power_up(struct hda_codec *codec)
+{
+	__snd_hda_power_up(codec, false);
+}
 EXPORT_SYMBOL_HDA(snd_hda_power_up);
 
+/**
+ * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending
+ *   D3 transition to complete.  This differs from snd_hda_power_up() when
+ *   power_transition == -1.  snd_hda_power_up sees this case as a nop,
+ *   snd_hda_power_up_d3wait waits for the D3 transition to complete then powers
+ *   back up.
+ * @codec: HD-audio codec
+ *
+ * Cancel any power down operation hapenning on the work queue, then power up.
+ */
+void snd_hda_power_up_d3wait(struct hda_codec *codec)
+{
+	/* This will cancel and wait for pending power_work to complete. */
+	__snd_hda_power_up(codec, true);
+}
+EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait);
+
 #define power_save(codec)	\
 	((codec)->bus->power_save ? *(codec)->bus->power_save : 0)
 
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 4fc3960..2fdaadb 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg);
  */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 void snd_hda_power_up(struct hda_codec *codec);
+void snd_hda_power_up_d3wait(struct hda_codec *codec);
 void snd_hda_power_down(struct hda_codec *codec);
 void snd_hda_update_power_acct(struct hda_codec *codec);
 #else
 static inline void snd_hda_power_up(struct hda_codec *codec) {}
+static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {}
 static inline void snd_hda_power_down(struct hda_codec *codec) {}
 #endif
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0276382..7757536 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 				   buff_step);
 	snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
 				   buff_step);
-	snd_hda_power_up(apcm->codec);
+	snd_hda_power_up_d3wait(apcm->codec);
 	err = hinfo->ops.open(hinfo, apcm->codec, substream);
 	if (err < 0) {
 		azx_release_device(azx_dev);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 172370b..2bf99fc 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -445,8 +445,10 @@ static int conexant_init(struct hda_codec *codec)
 
 static void conexant_free(struct hda_codec *codec)
 {
+	struct conexant_spec *spec = codec->spec;
+	snd_hda_gen_free(&spec->gen);
 	snd_hda_detach_beep_device(codec);
-	kfree(codec->spec);
+	kfree(spec);
 }
 
 static const struct snd_kcontrol_new cxt_capture_mixers[] = {
@@ -4466,6 +4468,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC),
 	{}
 };
 
@@ -4497,6 +4500,7 @@ static int patch_conexant_auto(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	snd_hda_gen_init(&spec->gen);
 
 	switch (codec->vendor_id) {
 	case 0x14f15045:
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f8f4906..aa4c25e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2289,6 +2289,7 @@ static void alc_free(struct hda_codec *codec)
 	alc_shutup(codec);
 	alc_free_kctls(codec);
 	alc_free_bind_ctls(codec);
+	snd_hda_gen_free(&spec->gen);
 	kfree(spec);
 	snd_hda_detach_beep_device(codec);
 }
@@ -4253,6 +4254,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
 		return -ENOMEM;
 	codec->spec = spec;
 	spec->mixer_nid = mixer_nid;
+	snd_hda_gen_init(&spec->gen);
 
 	err = alc_codec_rename_from_preset(codec);
 	if (err < 0) {
@@ -6686,6 +6688,31 @@ static const struct alc_model_fixup alc662_fixup_models[] = {
 	{}
 };
 
+static void alc662_fill_coef(struct hda_codec *codec)
+{
+	int val, coef;
+
+	coef = alc_get_coef0(codec);
+
+	switch (codec->vendor_id) {
+	case 0x10ec0662:
+		if ((coef & 0x00f0) == 0x0030) {
+			val = alc_read_coef_idx(codec, 0x4); /* EAPD Ctrl */
+			alc_write_coef_idx(codec, 0x4, val & ~(1<<10));
+		}
+		break;
+	case 0x10ec0272:
+	case 0x10ec0273:
+	case 0x10ec0663:
+	case 0x10ec0665:
+	case 0x10ec0670:
+	case 0x10ec0671:
+	case 0x10ec0672:
+		val = alc_read_coef_idx(codec, 0xd); /* EAPD Ctrl */
+		alc_write_coef_idx(codec, 0xd, val | (1<<14));
+		break;
+	}
+}
 
 /*
  */
@@ -6705,12 +6732,8 @@ static int patch_alc662(struct hda_codec *codec)
 
 	alc_fix_pll_init(codec, 0x20, 0x04, 15);
 
-	if ((alc_get_coef0(codec) & (1 << 14)) &&
-	    codec->bus->pci->subsystem_vendor == 0x1025 &&
-	    spec->cdefine.platform_type == 1) {
-		if (alc_codec_rename(codec, "ALC272X") < 0)
-			goto error;
-	}
+	spec->init_hook = alc662_fill_coef;
+	alc662_fill_coef(codec);
 
 	alc_pick_fixup(codec, alc662_fixup_models,
 		       alc662_fixup_tbl, alc662_fixups);
@@ -6718,6 +6741,13 @@ static int patch_alc662(struct hda_codec *codec)
 
 	alc_auto_parse_customize_define(codec);
 
+	if ((alc_get_coef0(codec) & (1 << 14)) &&
+	    codec->bus->pci->subsystem_vendor == 0x1025 &&
+	    spec->cdefine.platform_type == 1) {
+		if (alc_codec_rename(codec, "ALC272X") < 0)
+			goto error;
+	}
+
 	/* automatic parse from the BIOS config */
 	err = alc662_parse_auto_config(codec);
 	if (err < 0)
@@ -6800,6 +6830,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
 	{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
 	{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
+	{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7db8228..0767528 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4367,7 +4367,7 @@ static int stac92xx_init(struct hda_codec *codec)
 					 AC_PINCTL_IN_EN);
 	for (i = 0; i < spec->num_pwrs; i++)  {
 		hda_nid_t nid = spec->pwr_nids[i];
-		int pinctl, def_conf;
+		unsigned int pinctl, def_conf;
 
 		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		def_conf = get_defcfg_connect(def_conf);
@@ -4376,6 +4376,11 @@ static int stac92xx_init(struct hda_codec *codec)
 			stac_toggle_power_map(codec, nid, 0);
 			continue;
 		}
+		if (def_conf == AC_JACK_PORT_FIXED) {
+			/* no need for jack detection for fixed pins */
+			stac_toggle_power_map(codec, nid, 1);
+			continue;
+		}
 		/* power on when no jack detection is available */
 		/* or when the VREF is used for controlling LED */
 		if (!spec->hp_detect ||
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 64d2a4f..e9b62b5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -935,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	}
 
 found:
-	data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
-	snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
-		      data | (pll_p << PLLP_SHIFT));
+	snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p);
 	snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG,
 		      pll_r << PLLR_SHIFT);
 	snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT);
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 6f097fb..08c7f66 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -166,6 +166,7 @@
 
 /* PLL registers bitfields */
 #define PLLP_SHIFT		0
+#define PLLP_MASK		7
 #define PLLQ_SHIFT		3
 #define PLLR_SHIFT		0
 #define PLLJ_SHIFT		2
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index acbdc5f..32682c1 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1491,6 +1491,7 @@ static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = {
 
 static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = {
 	5644800,
+	3763200,
 	2882400,
 	1881600,
 	1411200,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 65d525d..812acd8 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 				return ret;
 			}
 
+			regcache_cache_only(wm8904->regmap, false);
 			regcache_sync(wm8904->regmap);
 
 			/* Enable bias */
@@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
 				    WM8904_BIAS_ENA, 0);
 
-#ifdef CONFIG_REGULATOR
-		/* Post 2.6.34 we will be able to get a callback when
-		 * the regulators are disabled which we can use but
-		 * for now just assume that the power will be cut if
-		 * the regulator API is in use.
-		 */
-		codec->cache_sync = 1;
-#endif
+		regcache_cache_only(wm8904->regmap, true);
+		regcache_mark_dirty(wm8904->regmap);
 
 		regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
 				       wm8904->supplies);
@@ -2084,10 +2079,8 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 	struct wm8904_pdata *pdata = wm8904->pdata;
-	u16 *reg_cache = codec->reg_cache;
 	int ret, i;
 
-	codec->cache_sync = 1;
 	codec->control_data = wm8904->regmap;
 
 	switch (wm8904->devtype) {
@@ -2150,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 		goto err_enable;
 	}
 
+	regcache_cache_only(wm8904->regmap, true);
 	/* Change some default settings - latch VU and enable ZC */
 	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
 			    WM8904_ADC_VU, WM8904_ADC_VU);
@@ -2180,14 +2174,18 @@ static int wm8904_probe(struct snd_soc_codec *codec)
 			if (!pdata->gpio_cfg[i])
 				continue;
 
-			reg_cache[WM8904_GPIO_CONTROL_1 + i]
-				= pdata->gpio_cfg[i] & 0xffff;
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_GPIO_CONTROL_1 + i,
+					   0xffff,
+					   pdata->gpio_cfg[i]);
 		}
 
 		/* Zero is the default value for these anyway */
 		for (i = 0; i < WM8904_MIC_REGS; i++)
-			reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i]
-				= pdata->mic_cfg[i];
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_MIC_BIAS_CONTROL_0 + i,
+					   0xffff,
+					   pdata->mic_cfg[i]);
 	}
 
 	/* Set Class W by default - this will be managed by the Class
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index aa8c98b..1436b6c 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -727,9 +727,6 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode)
 	if (!wm8994->jackdet || !wm8994->jack_cb)
 		return;
 
-	if (!wm8994->jackdet || !wm8994->jack_cb)
-		return;
-
 	if (wm8994->active_refcount)
 		mode = WM1811_JACKDET_MODE_AUDIO;
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index 8af422e..dc9b42b 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 		}
 	}
 
-	regcache_cache_only(codec->control_data, true);
-
 	/* Apply platform data settings */
 	snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
 			    WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
@@ -3051,7 +3049,6 @@ static int wm8996_remove(struct snd_soc_codec *codec)
 	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
 		regulator_unregister_notifier(wm8996->supplies[i].consumer,
 					      &wm8996->disable_nb[i]);
-	regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
 
 	return 0;
 }
@@ -3206,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c,
 	dev_info(&i2c->dev, "revision %c\n",
 		 (reg & WM8996_CHIP_REV_MASK) + 'A');
 
-	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
-
 	ret = wm8996_reset(wm8996);
 	if (ret < 0) {
 		dev_err(&i2c->dev, "Failed to issue reset\n");
 		goto err_regmap;
 	}
 
+	regcache_cache_only(wm8996->regmap, true);
+	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+
 	wm8996_init_gpio(wm8996);
 
 	ret = snd_soc_register_codec(&i2c->dev,
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 1c2aa7f..4da5fc5 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -33,7 +33,6 @@
 
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <mach/audio.h>
 
 #include "../../arm/pxa2xx-pcm.h"
 #include "pxa-ssp.h"
@@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div)
 {
 	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 
-	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+	if (ssp->type == PXA25x_SSP) {
 		sscr0 &= ~0x0000ff00;
 		sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
 	} else {
@@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp)
 	u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0);
 	u32 div;
 
-	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+	if (ssp->type == PXA25x_SSP)
 		div = ((sscr0 >> 8) & 0xff) * 2 + 2;
 	else
 		div = ((sscr0 >> 8) & 0xfff) + 1;
@@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		break;
 	case PXA_SSP_CLK_PLL:
 		/* Internal PLL is fixed */
-		if (cpu_is_pxa25x())
+		if (ssp->type == PXA25x_SSP)
 			priv->sysclk = 1843200;
 		else
 			priv->sysclk = 13000000;
@@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 
 	/* The SSP clock must be disabled when changing SSP clock mode
 	 * on PXA2xx.  On PXA3xx it must be enabled when doing so. */
-	if (!cpu_is_pxa3xx())
+	if (ssp->type != PXA3xx_SSP)
 		clk_disable(ssp->clk);
 	val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0;
 	pxa_ssp_write_reg(ssp, SSCR0, val);
-	if (!cpu_is_pxa3xx())
+	if (ssp->type != PXA3xx_SSP)
 		clk_enable(ssp->clk);
 
 	return 0;
@@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 	case PXA_SSP_AUDIO_DIV_SCDB:
 		val = pxa_ssp_read_reg(ssp, SSACD);
 		val &= ~SSACD_SCDB;
-#if defined(CONFIG_PXA3xx)
-		if (cpu_is_pxa3xx())
+		if (ssp->type == PXA3xx_SSP)
 			val &= ~SSACD_SCDX8;
-#endif
 		switch (div) {
 		case PXA_SSP_CLK_SCDB_1:
 			val |= SSACD_SCDB;
 			break;
 		case PXA_SSP_CLK_SCDB_4:
 			break;
-#if defined(CONFIG_PXA3xx)
 		case PXA_SSP_CLK_SCDB_8:
-			if (cpu_is_pxa3xx())
+			if (ssp->type == PXA3xx_SSP)
 				val |= SSACD_SCDX8;
 			else
 				return -EINVAL;
 			break;
-#endif
 		default:
 			return -EINVAL;
 		}
@@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 	struct ssp_device *ssp = priv->ssp;
 	u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70;
 
-#if defined(CONFIG_PXA3xx)
-	if (cpu_is_pxa3xx())
+	if (ssp->type == PXA3xx_SSP)
 		pxa_ssp_write_reg(ssp, SSACDD, 0);
-#endif
 
 	switch (freq_out) {
 	case 5622000:
@@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 		break;
 
 	default:
-#ifdef CONFIG_PXA3xx
 		/* PXA3xx has a clock ditherer which can be used to generate
 		 * a wider range of frequencies - calculate a value for it.
 		 */
-		if (cpu_is_pxa3xx()) {
+		if (ssp->type == PXA3xx_SSP) {
 			u32 val;
 			u64 tmp = 19968;
 			tmp *= 1000000;
@@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
 				val, freq_out);
 			break;
 		}
-#endif
 
 		return -EINVAL;
 	}
@@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	/* bit size */
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-#ifdef CONFIG_PXA3xx
-		if (cpu_is_pxa3xx())
+		if (ssp->type == PXA3xx_SSP)
 			sscr0 |= SSCR0_FPCKE;
-#endif
 		sscr0 |= SSCR0_DataSize(16);
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
@@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			* trying and failing a lot; some of the registers
 			* needed for that mode are only available on PXA3xx.
 			*/
-
-#ifdef CONFIG_PXA3xx
-			if (!cpu_is_pxa3xx())
+			if (ssp->type != PXA3xx_SSP)
 				return -EINVAL;
 
 			sspsp |= SSPSP_SFRMWDTH(width * 2);
@@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			sspsp |= SSPSP_EDMYSTOP(3);
 			sspsp |= SSPSP_DMYSTOP(3);
 			sspsp |= SSPSP_DMYSTRT(1);
-#else
-			return -EINVAL;
-#endif
 		} else {
 			/* The frame width is the width the LRCLK is
 			 * asserted for; the delay is expressed in
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 0b0df49..3b6da91 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
+static int tegra_wm8903_remove(struct snd_soc_card *card)
+{
+	struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]);
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	wm8903_mic_detect(codec, NULL, 0, 0);
+
+	return 0;
+}
+
 static struct snd_soc_dai_link tegra_wm8903_dai = {
 	.name = "WM8903",
 	.stream_name = "WM8903 PCM",
@@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = {
 	.dai_link = &tegra_wm8903_dai,
 	.num_links = 1,
 
+	.remove = tegra_wm8903_remove,
+
 	.controls = tegra_wm8903_controls,
 	.num_controls = ARRAY_SIZE(tegra_wm8903_controls),
 	.dapm_widgets = tegra_wm8903_dapm_widgets,
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index 6f9715a..56ad923 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload(
 	int ret;
 	u8 data;
 	struct usb_device *device = interface_to_usbdev(intf);
-	const struct firmware *fw = 0;
+	const struct firmware *fw = NULL;
 	struct ihex_record *rec = kmalloc(sizeof(struct ihex_record),
 			GFP_KERNEL);
 
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index 41daaa2..e71fe55 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -341,6 +341,14 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = audigy2nx_map,
 		.selector_map = audigy2nx_selectors,
 	},
+	{	/* Logitech, Inc. QuickCam Pro for Notebooks */
+		.id = USB_ID(0x046d, 0x0991),
+		.ignore_ctl_error = 1,
+	},
+	{	/* Logitech, Inc. QuickCam E 3500 */
+		.id = USB_ID(0x046d, 0x09a4),
+		.ignore_ctl_error = 1,
+	},
 	{
 		/* Hercules DJ Console (Windows Edition) */
 		.id = USB_ID(0x06f8, 0xb000),
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index cdf8b76..54607f8 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 		    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
 		     get_endpoint(alts, 1)->bSynchAddress != 0 &&
 		     !implicit_fb)) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
-				   dev->devnum, fmt->iface, fmt->altsetting);
+			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+				   dev->devnum, fmt->iface, fmt->altsetting,
+				   get_endpoint(alts, 1)->bmAttributes,
+				   get_endpoint(alts, 1)->bLength,
+				   get_endpoint(alts, 1)->bSynchAddress);
 			return -EINVAL;
 		}
 		ep = get_endpoint(alts, 1)->bEndpointAddress;
-		if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+		if (!implicit_fb &&
+		    get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
 		    (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
-		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) ||
-		     ( is_playback && !implicit_fb))) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n",
-				   dev->devnum, fmt->iface, fmt->altsetting);
+		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+				   dev->devnum, fmt->iface, fmt->altsetting,
+				   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
 			return -EINVAL;
 		}
 
@@ -1147,7 +1151,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
 	return -EINVAL;
 }
 
-int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream,
+					     int cmd)
 {
 	int err;
 	struct snd_usb_substream *subs = substream->runtime->private_data;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index d89ab4c..79780fa 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1831,6 +1831,36 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+{
+	USB_DEVICE(0x0582, 0x014d),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		/* .vendor_name = "BOSS", */
+		/* .product_name = "GT-100", */
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 1,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_INTERFACE
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_MIDI_FIXED_ENDPOINT,
+				.data = & (const struct snd_usb_midi_endpoint_info) {
+					.out_cables = 0x0001,
+					.in_cables  = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 
 /* Guillemot devices */
 {