Merge branch 'fix/misc' into topic/misc

Conflicts:
	sound/pci/hda/patch_realtek.c

16 files changed, 191 insertions, 57 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1793574..8728876 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -316,10 +316,10 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
 	if (!params->info)
 		params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES;
 	if (!params->fifo_size) {
-		if (snd_mask_min(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT]) ==
-		    snd_mask_max(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT]) &&
-                    snd_mask_min(&params->masks[SNDRV_PCM_HW_PARAM_CHANNELS]) ==
-                    snd_mask_max(&params->masks[SNDRV_PCM_HW_PARAM_CHANNELS])) {
+		m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+		i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+		if (snd_mask_min(m) == snd_mask_max(m) &&
+                    snd_interval_min(i) == snd_interval_max(i)) {
 			changed = substream->ops->ioctl(substream,
 					SNDRV_PCM_IOCTL1_FIFO_SIZE, params);
 			if (changed < 0)
diff --git a/sound/core/sound.c b/sound/core/sound.c
index 7872a02..563d196 100644
--- a/sound/core/sound.c
+++ b/sound/core/sound.c
@@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void)
 	unregister_chrdev(major, "alsa");
 }
 
-module_init(alsa_sound_init)
-module_exit(alsa_sound_exit)
+subsys_initcall(alsa_sound_init);
+module_exit(alsa_sound_exit);
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 7fe1226..0c164e5 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -93,7 +93,7 @@ static int snd_oss_kernel_minor(int type, struct snd_card *card, int dev)
 	default:
 		return -EINVAL;
 	}
-	if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OSS_MINORS))
+	if (minor < 0 || minor >= SNDRV_OSS_MINORS)
 		return -EINVAL;
 	return minor;
 }
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 81e2bfc..49d572a 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -297,6 +297,7 @@ static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = {
 MODULE_DEVICE_TABLE(pci, snd_atiixp_ids);
 
 static struct snd_pci_quirk atiixp_quirks[] __devinitdata = {
+	SND_PCI_QUIRK(0x105b, 0x0c81, "Foxconn RC4107MA-RS2", 0),
 	SND_PCI_QUIRK(0x15bd, 0x3100, "DFI RS482", 0),
 	{ } /* terminator */
 };
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 2783ce6..8dab82d 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1864,7 +1864,9 @@ static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
 	/* The hardware doesn't tell us which substream caused the irq,
 	thus we have to check all running substreams. */
 	for (ss = 0; ss < DSP_MAXPIPES; ss++) {
-		if ((substream = chip->substream[ss])) {
+		substream = chip->substream[ss];
+		if (substream && ((struct audiopipe *)substream->runtime->
+				private_data)->state == PIPE_STATE_STARTED) {
 			period = pcm_pointer(substream) /
 				substream->runtime->period_size;
 			if (period != chip->last_period[ss]) {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ac05bef..8d8e0b5 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -426,6 +426,7 @@ struct azx {
 
 	/* flags */
 	int position_fix;
+	int poll_count;
 	unsigned int running :1;
 	unsigned int initialized :1;
 	unsigned int single_cmd :1;
@@ -506,7 +507,7 @@ static char *driver_short_names[] __devinitdata = {
 #define get_azx_dev(substream) (substream->runtime->private_data)
 
 static int azx_acquire_irq(struct azx *chip, int do_disconnect);
-
+static int azx_send_cmd(struct hda_bus *bus, unsigned int val);
 /*
  * Interface for HD codec
  */
@@ -664,11 +665,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 {
 	struct azx *chip = bus->private_data;
 	unsigned long timeout;
+	int do_poll = 0;
 
  again:
 	timeout = jiffies + msecs_to_jiffies(1000);
 	for (;;) {
-		if (chip->polling_mode) {
+		if (chip->polling_mode || do_poll) {
 			spin_lock_irq(&chip->reg_lock);
 			azx_update_rirb(chip);
 			spin_unlock_irq(&chip->reg_lock);
@@ -676,6 +678,9 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		if (!chip->rirb.cmds[addr]) {
 			smp_rmb();
 			bus->rirb_error = 0;
+
+			if (!do_poll)
+				chip->poll_count = 0;
 			return chip->rirb.res[addr]; /* the last value */
 		}
 		if (time_after(jiffies, timeout))
@@ -688,6 +693,16 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus,
 		}
 	}
 
+	if (!chip->polling_mode && chip->poll_count < 2) {
+		snd_printdd(SFX "azx_get_response timeout, "
+			   "polling the codec once: last cmd=0x%08x\n",
+			   chip->last_cmd[addr]);
+		do_poll = 1;
+		chip->poll_count++;
+		goto again;
+	}
+
+
 	if (!chip->polling_mode) {
 		snd_printk(KERN_WARNING SFX "azx_get_response timeout, "
 			   "switching to polling mode: last cmd=0x%08x\n",
@@ -1878,6 +1893,9 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 
 	if (!bdl_pos_adj[chip->dev_index])
 		return 1; /* no delayed ack */
+	if (WARN_ONCE(!azx_dev->period_bytes,
+		      "hda-intel: zero azx_dev->period_bytes"))
+		return 0; /* this shouldn't happen! */
 	if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		return 0; /* NG - it's below the period boundary */
 	return 1; /* OK, it's fine */
@@ -2043,7 +2061,7 @@ static int azx_acquire_irq(struct azx *chip, int do_disconnect)
 {
 	if (request_irq(chip->pci->irq, azx_interrupt,
 			chip->msi ? 0 : IRQF_SHARED,
-			"HDA Intel", chip)) {
+			"hda_intel", chip)) {
 		printk(KERN_ERR "hda-intel: unable to grab IRQ %d, "
 		       "disabling device\n", chip->pci->irq);
 		if (do_disconnect)
@@ -2332,6 +2350,7 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
  */
 static struct snd_pci_quirk msi_black_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */
+	SND_PCI_QUIRK(0x1043, 0x829c, "ASUS", 0), /* nvidia */
 	{}
 };
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 141ff44..f628c33 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1097,6 +1097,16 @@ static void alc889_coef_init(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
 }
 
+/* turn on/off EAPD control (only if available) */
+static void set_eapd(struct hda_codec *codec, hda_nid_t nid, int on)
+{
+	if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+		return;
+	if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
+				    on ? 2 : 0);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	unsigned int tmp;
@@ -1114,25 +1124,22 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 	case ALC_INIT_DEFAULT:
 		switch (codec->vendor_id) {
 		case 0x10ec0260:
-			snd_hda_codec_write(codec, 0x0f, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x10, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x0f, 1);
+			set_eapd(codec, 0x10, 1);
 			break;
 		case 0x10ec0262:
 		case 0x10ec0267:
 		case 0x10ec0268:
 		case 0x10ec0269:
+		case 0x10ec0270:
 		case 0x10ec0272:
 		case 0x10ec0660:
 		case 0x10ec0662:
 		case 0x10ec0663:
 		case 0x10ec0862:
 		case 0x10ec0889:
-			snd_hda_codec_write(codec, 0x14, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
-			snd_hda_codec_write(codec, 0x15, 0,
-					    AC_VERB_SET_EAPD_BTLENABLE, 2);
+			set_eapd(codec, 0x14, 1);
+			set_eapd(codec, 0x15, 1);
 			break;
 		}
 		switch (codec->vendor_id) {
@@ -1234,6 +1241,8 @@ static void alc_init_auto_mic(struct hda_codec *codec)
 			return; /* invalid entry */
 		}
 	}
+	if (!ext || !fixed)
+		return;
 	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
 		return; /* no unsol support */
 	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
@@ -4917,6 +4926,49 @@ static void fixup_automic_adc(struct hda_codec *codec)
 	spec->auto_mic = 0; /* disable auto-mic to be sure */
 }
 
+/* choose the ADC/MUX containing the input pin and initialize the setup */
+static void fixup_single_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin;
+	int i;
+
+	/* search for the input pin; there must be only one */
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (spec->autocfg.input_pins[i]) {
+			pin = spec->autocfg.input_pins[i];
+			break;
+		}
+	}
+	if (!pin)
+		return;
+
+	/* set the default connection to that pin */
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t cap = spec->capsrc_nids ?
+			spec->capsrc_nids[i] : spec->adc_nids[i];
+		int idx;
+
+		idx = get_connection_index(codec, cap, pin);
+		if (idx < 0)
+			continue;
+		/* use only this ADC */
+		if (spec->capsrc_nids)
+			spec->capsrc_nids += i;
+		spec->adc_nids += i;
+		spec->num_adc_nids = 1;
+		/* select or unmute this route */
+		if (get_wcaps_type(get_wcaps(codec, cap)) == AC_WID_AUD_MIX) {
+			snd_hda_codec_amp_stereo(codec, cap, HDA_INPUT, idx,
+						 HDA_AMP_MUTE, 0);
+		} else {
+			snd_hda_codec_write_cache(codec, cap, 0,
+					  AC_VERB_SET_CONNECT_SEL, idx);
+		}
+		return;
+	}
+}
+
 static void set_capture_mixer(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -4929,14 +4981,15 @@ static void set_capture_mixer(struct hda_codec *codec)
 		  alc_capture_mixer3 },
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
-		int mux;
-		if (spec->auto_mic) {
-			mux = 0;
+		int mux = 0;
+		if (spec->auto_mic)
 			fixup_automic_adc(codec);
-		} else if (spec->input_mux && spec->input_mux->num_items > 1)
-			mux = 1;
-		else
-			mux = 0;
+		else if (spec->input_mux) {
+			if (spec->input_mux->num_items > 1)
+				mux = 1;
+			else if (spec->input_mux->num_items == 1)
+				fixup_single_adc(codec);
+		}
 		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
 	}
 }
@@ -7201,8 +7254,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
 	HDA_BIND_MUTE   ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
 	HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
-	HDA_BIND_MUTE   ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -7603,6 +7656,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
 	/* Front Mic pin: input vref at 80% */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
@@ -7787,6 +7841,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
+static void alc885_mb5_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x14, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+
+}
+
+static void alc885_mb5_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	/* Headphone insertion or removal. */
+	if ((res >> 26) == ALC880_HP_EVENT)
+		alc885_mb5_automute(codec);
+}
+
 static void alc885_imac91_automute(struct hda_codec *codec)
 {
  	unsigned int present;
@@ -9233,6 +9308,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.input_mux = &mb5_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc885_mb5_unsol_event,
+		.init_hook = alc885_mb5_automute,
 	},
 	[ALC885_MACPRO] = {
 		.mixers = { alc882_macpro_mixer },
@@ -9510,6 +9587,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
 		.channel_mode = alc883_3ST_6ch_modes,
 		.need_dac_fix = 1,
+		.const_channel_count = 6,
 		.num_mux_defs =
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
@@ -10420,7 +10498,7 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
-	spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
+	spec->autocfg.speaker_pins[0] = 0x14;
 }
 
 static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -11310,7 +11388,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 #define alc262_auto_create_input_ctls \
-	alc880_auto_create_input_ctls
+	alc882_auto_create_input_ctls
 
 /*
  * generic initialization of ADC, input mixers and output mixers
@@ -11849,9 +11927,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc262_modes),
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
+		.unsol_event = alc_sku_unsol_event,
 		.setup = alc262_hp_t5735_setup,
-		.init_hook = alc_automute_amp,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_HP_RP5700] = {
 		.mixers = { alc262_hp_rp5700_mixer },
@@ -12605,6 +12683,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 		dac = 0x02;
 		break;
 	case 0x15:
+	case 0x21:
 		dac = 0x03;
 		break;
 	default:
@@ -14991,6 +15070,8 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+	if (!spec->cap_mixer)
+		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
 
 	spec->vmaster_nid = 0x03;
@@ -15633,7 +15714,7 @@ 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, 0x15, 0x22, 0);
 }
 
 
@@ -17391,7 +17472,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
 		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4),
+	SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO),
 	SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 117919a..dbffb5b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4737,6 +4737,26 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	}
 }
 
+static int hp_blike_system(u32 subsystem_id);
+
+static void set_hp_led_gpio(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	switch (codec->vendor_id) {
+	case 0x111d7608:
+		/* GPIO 0 */
+		spec->gpio_led = 0x01;
+		break;
+	case 0x111d7600:
+	case 0x111d7601:
+	case 0x111d7602:
+	case 0x111d7603:
+		/* GPIO 3 */
+		spec->gpio_led = 0x08;
+		break;
+	}
+}
+
 /*
  * This method searches for the mute LED GPIO configuration
  * provided as OEM string in SMBIOS. The format of that string
@@ -4748,6 +4768,14 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
  *
  * So, HP B-series like systems may have HP_Mute_LED_0 (current models)
  * or  HP_Mute_LED_0_3 (future models) OEM SMBIOS strings
+ *
+ *
+ * The dv-series laptops don't seem to have the HP_Mute_LED* strings in
+ * SMBIOS - at least the ones I have seen do not have them - which include
+ * my own system (HP Pavilion dv6-1110ax) and my cousin's
+ * HP Pavilion dv9500t CTO.
+ * Need more information on whether it is true across the entire series.
+ * -- kunal
  */
 static int find_mute_led_gpio(struct hda_codec *codec)
 {
@@ -4758,28 +4786,27 @@ static int find_mute_led_gpio(struct hda_codec *codec)
 		while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING,
 								NULL, dev))) {
 			if (sscanf(dev->name, "HP_Mute_LED_%d_%d",
-			      &spec->gpio_led_polarity,
-			      &spec->gpio_led) == 2) {
+				  &spec->gpio_led_polarity,
+				  &spec->gpio_led) == 2) {
 				spec->gpio_led = 1 << spec->gpio_led;
 				return 1;
 			}
 			if (sscanf(dev->name, "HP_Mute_LED_%d",
-			      &spec->gpio_led_polarity) == 1) {
-				switch (codec->vendor_id) {
-				case 0x111d7608:
-					/* GPIO 0 */
-					spec->gpio_led = 0x01;
-					return 1;
-				case 0x111d7600:
-				case 0x111d7601:
-				case 0x111d7602:
-				case 0x111d7603:
-					/* GPIO 3 */
-					spec->gpio_led = 0x08;
-					return 1;
-				}
+				  &spec->gpio_led_polarity) == 1) {
+				set_hp_led_gpio(codec);
+				return 1;
 			}
 		}
+
+		/*
+		 * Fallback case - if we don't find the DMI strings,
+		 * we statically set the GPIO - if not a B-series system.
+		 */
+		if (!hp_blike_system(codec->subsystem_id)) {
+			set_hp_led_gpio(codec);
+			spec->gpio_led_polarity = 1;
+			return 1;
+		}
 	}
 	return 0;
 }
@@ -5603,6 +5630,8 @@ again:
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
+	snd_printdd("Found board config: %d\n", spec->board_config);
+
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index bb08a28..960a227 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1058,7 +1058,7 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate)
 				 rptr.retwords[2] != M &&
 				 rptr.retwords[3] != N &&
 				 i++ < MAX_WRITE_RETRY);
-			if (i == MAX_WRITE_RETRY) {
+			if (i > MAX_WRITE_RETRY) {
 				snd_printdd("sent samplerate %d: %d failed\n",
 					    *intdec, rate);
 				return -EIO;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index a9dc5fb..da589d8 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 	u16 reg;
 
 	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
+	for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index ebbf11b..718ef91 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 		iface |= 0x3 << 8;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x3 << 8;	/* lg not sure which mode */
+		iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index ce5515e..3595bd5 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev)
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
-	u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults),
+	u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults),
 				 GFP_KERNEL);
 
 	/* Bring the codec back up to standby first to minimise pop/clicks */
@@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev)
 		for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
 			if (tmp_cache[i] != reg_cache[i])
 				snd_soc_write(codec, i, tmp_cache[i]);
+		kfree(tmp_cache);
 	} else {
 		dev_err(&i2c->dev, "Failed to allocate temporary cache\n");
 	}
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 3db8a6c..19283e5 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -25,7 +25,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o
-obj-$(CONFIG_SND_OMAP_SOC_OMAP3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 71b2c16..68980c1 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
+	{"PCM DAC", NULL, "APLL Enable"},
 	{"Headphone Amplifier", NULL, "PCM DAC"},
 	{"Line Out", NULL, "PCM DAC"},
 	{"Headphone Jack", NULL, "Headphone Amplifier"},
diff --git a/sound/sound_core.c b/sound/sound_core.c
index dbca7c9..7c2d677 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void)
 	class_destroy(sound_class);
 }
 
-module_init(init_soundcore);
+subsys_initcall(init_soundcore);
 module_exit(cleanup_soundcore);
 
 
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 8a8f6251..c6b9c8c 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -1959,7 +1959,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
 	struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
 	struct snd_usb_substream *subs = &as->substream[direction];
 
-	if (subs->interface >= 0) {
+	if (!as->chip->shutdown && subs->interface >= 0) {
 		usb_set_interface(subs->dev, subs->interface, 0);
 		subs->interface = -1;
 	}