Merge branch 'fix/hda' into topic/hda

7 files changed, 50 insertions, 19 deletions

diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index 86ee16ca365..440030818db 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -209,6 +209,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle,
 		isight->packet_index = -1;
 		return;
 	}
+	fw_iso_context_queue_flush(isight->context);
 
 	if (++index >= QUEUE_LENGTH)
 		index = 0;
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 4967eabe774..55f0647458c 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -54,7 +54,7 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 {
 	return 0;
 }
-void snd_hda_detach_beep_device(struct hda_codec *codec)
+static inline void snd_hda_detach_beep_device(struct hda_codec *codec)
 {
 }
 #endif
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0f90fac34a7..b0cf726d4b9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4884,7 +4884,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
 	SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
 	SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
-	SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST),
 	SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
 	SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
 	SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
@@ -12601,6 +12600,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
  */
 enum {
 	PINFIX_FSC_H270,
+	PINFIX_HP_Z200,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -12613,9 +12613,17 @@ static const struct alc_fixup alc262_fixups[] = {
 			{ }
 		}
 	},
+	[PINFIX_HP_Z200] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x16, 0x99130120 }, /* internal speaker */
+			{ }
+		}
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200),
 	SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270),
 	{}
 };
@@ -12732,6 +12740,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = {
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series",
 			   ALC262_HP_BPC),
+	SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200",
+			   ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
 			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -13873,7 +13883,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
 	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-	SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
 	SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
 	{}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 995974d0b12..de004ed2ef8 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol,
 	struct via_spec *spec = codec->spec;
 	hda_nid_t nid = kcontrol->private_value;
 	unsigned int pinsel = ucontrol->value.enumerated.item[0];
+	unsigned int parm0, parm1;
 	/* Get Independent Mode index of headphone pin widget */
 	spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel
 		? 1 : 0;
-	if (spec->codec_type == VT1718S)
+	if (spec->codec_type == VT1718S) {
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0);
+		/* Set correct mute switch for MW3 */
+		parm0 = spec->hp_independent_mode ?
+			       AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0);
+		parm1 = spec->hp_independent_mode ?
+			       AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm0);
+		snd_hda_codec_write(codec, 0x1b, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, parm1);
+	}
 	else
 		snd_hda_codec_write(codec, nid, 0,
 				    AC_VERB_SET_CONNECT_SEL, pinsel);
@@ -832,10 +843,13 @@ static int via_hp_build(struct hda_codec *codec)
 	knew->subdevice = HDA_SUBDEV_NID_FLAG | nid;
 	knew->private_value = nid;
 
-	knew = via_clone_control(spec, &via_hp_mixer[1]);
-	if (knew == NULL)
-		return -ENOMEM;
-	knew->subdevice = side_mute_channel(spec);
+	nid = side_mute_channel(spec);
+	if (nid) {
+		knew = via_clone_control(spec, &via_hp_mixer[1]);
+		if (knew == NULL)
+			return -ENOMEM;
+		knew->subdevice = nid;
+	}
 
 	return 0;
 }
@@ -4274,9 +4288,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-
-	/* Setup default input of Front HP to MW9 */
-	{0x28, AC_VERB_SET_CONNECT_SEL, 0x1},
 	/* PW9 PW10 Output enable */
 	{0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
 	{0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN},
@@ -4285,10 +4296,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* Enable Boost Volume backdoor */
 	{0x1, 0xf88, 0x8},
 	/* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
@@ -4298,8 +4309,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = {
 	/* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */
 	{0x34, AC_VERB_SET_CONNECT_SEL, 0x2},
 	{0x35, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Unmute MW4's index 0 */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ }
 };
 
@@ -4447,6 +4456,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec,
 			if (err < 0)
 				return err;
 		} else if (i == AUTO_SEQ_FRONT) {
+			/* add control to mixer index 0 */
+			err = via_add_control(spec, VIA_CTL_WIDGET_VOL,
+					      "Master Front Playback Volume",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
+			err = via_add_control(spec, VIA_CTL_WIDGET_MUTE,
+					      "Master Front Playback Switch",
+					      HDA_COMPOSE_AMP_VAL(0x21, 3, 5,
+								  HDA_INPUT));
+			if (err < 0)
+				return err;
 			/* Front */
 			sprintf(name, "%s Playback Volume", chname[i]);
 			err = via_add_control(
diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c
index 34b24286d27..2692e5ae5f2 100644
--- a/sound/pci/lola/lola.c
+++ b/sound/pci/lola/lola.c
@@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip)
 	lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */
 }
 
-static int lola_parse_tree(struct lola *chip)
+static int __devinit lola_parse_tree(struct lola *chip)
 {
 	unsigned int val;
 	int nid, err;
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index a91719d5918..1e3ae3327dd 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -270,7 +270,6 @@ static int usb6fire_fw_ezusb_upload(
 	data = 0x00; /* resume ezusb cpu */
 	ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1);
 	if (ret < 0) {
-		release_firmware(fw);
 		snd_printk(KERN_ERR PREFIX "unable to upload ezusb "
 				"firmware %s: end message.\n", fwname);
 		return ret;
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index b137b25865c..d144cdb2f15 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub)
 	alsa_rt->hw = pcm_hw;
 
 	if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (rt->rate >= 0)
+		if (rt->rate < ARRAY_SIZE(rates))
 			alsa_rt->hw.rates = rates_alsaid[rt->rate];
 		alsa_rt->hw.channels_max = OUT_N_CHANNELS;
 		sub = &rt->playback;
 	} else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) {
-		if (rt->rate >= 0)
+		if (rt->rate < ARRAY_SIZE(rates))
 			alsa_rt->hw.rates = rates_alsaid[rt->rate];
 		alsa_rt->hw.channels_max = IN_N_CHANNELS;
 		sub = &rt->capture;