Merge branch 'fix/misc' into topic/misc

11 files changed, 64 insertions, 21 deletions

diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index a03fe80a7a73..eb715e732106 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -501,6 +501,10 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
 	int err;
 
 	aaci_pcm_hw_free(substream);
+	if (aacirun->pcm_open) {
+		snd_ac97_pcm_close(aacirun->pcm);
+		aacirun->pcm_open = 0;
+	}
 
 	err = devdma_hw_alloc(NULL, substream,
 			      params_buffer_bytes(params));
@@ -514,7 +518,7 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream,
 	else
 		err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params),
 					params_channels(params),
-					aacirun->pcm->r[1].slots);
+					aacirun->pcm->r[0].slots);
 
 	if (err)
 		goto out;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 6479e65858d3..905859d4f4df 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -110,6 +110,7 @@ struct conexant_spec {
 
 	unsigned int dell_automute;
 	unsigned int port_d_mode;
+	unsigned char ext_mic_bias;
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1927,6 +1928,11 @@ static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
 static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
 #define CXT5066_SPDIF_OUT	0x21
 
+/* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal with
+ * the DC input range of the codec. */
+#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
+
 static struct hda_channel_mode cxt5066_modes[1] = {
 	{ 2, NULL },
 };
@@ -1980,9 +1986,10 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5066_automic(struct hda_codec *codec)
 {
-	static struct hda_verb ext_mic_present[] = {
+	struct conexant_spec *spec = codec->spec;
+	struct hda_verb ext_mic_present[] = {
 		/* enable external mic, port B */
-		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
 
 		/* switch to external mic input */
 		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -2235,7 +2242,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
 	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
 
 	/* Port B: external microphone */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
 
 	/* Port C: internal microphone */
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
@@ -2353,6 +2360,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->input_mux = &cxt5066_capture_source;
 
 	spec->port_d_mode = PIN_HP;
+	spec->ext_mic_bias = PIN_VREF80;
 
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2384,6 +2392,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->port_d_mode = 0;
+		spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 9fb60276f5c9..6afdab09bab7 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -397,6 +397,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
 static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 	{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+	{ .id = 0x10de0005, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -406,6 +407,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 
 MODULE_ALIAS("snd-hda-codec-id:10de0002");
 MODULE_ALIAS("snd-hda-codec-id:10de0003");
+MODULE_ALIAS("snd-hda-codec-id:10de0005");
 MODULE_ALIAS("snd-hda-codec-id:10de0006");
 MODULE_ALIAS("snd-hda-codec-id:10de0007");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ff20048504b6..70583719282b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4684,9 +4684,9 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
 		}
 	}
 	if (spec->autocfg.dig_in_pin)
@@ -6249,7 +6249,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
 	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
 	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
-	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
+	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_AUTO), /* no quirk */
 	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600),
@@ -8911,10 +8911,11 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
 	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
-	/* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
-	 * no perfect solution yet
+	/* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
+	 * so apparently no perfect solution yet
 	 */
 	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+	SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
 	{} /* terminator */
 };
 
@@ -9813,9 +9814,9 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
 			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
-			spec->slave_dig_outs[i - 1] = dig_nid;
-			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+			if (i >= ARRAY_SIZE(spec->slave_dig_outs) - 1)
 				break;
+			spec->slave_dig_outs[i - 1] = dig_nid;
 		}
 	}
 	if (spec->autocfg.dig_in_pin)
@@ -11460,6 +11461,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
 	SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06),
+	SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO),
+	SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO),
 	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
 			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8eb6508cd991..86de305fc9f2 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1590,6 +1590,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 				"Dell Studio 17", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02be,
 				"Dell Studio 1555", STAC_DELL_M6_DMIC),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd,
+				"Dell Studio 1557", STAC_DELL_M6_DMIC),
 	{} /* terminator */
 };
 
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 9da2dae64c5b..d063149e7047 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -382,8 +382,8 @@ struct snd_ice1712 {
 #ifdef CONFIG_PM
 	int (*pm_suspend)(struct snd_ice1712 *);
 	int (*pm_resume)(struct snd_ice1712 *);
-	int pm_suspend_enabled:1;
-	int pm_saved_is_spdif_master:1;
+	unsigned int pm_suspend_enabled:1;
+	unsigned int pm_saved_is_spdif_master:1;
 	unsigned int pm_saved_spdif_ctrl;
 	unsigned char pm_saved_spdif_cfg;
 	unsigned int pm_saved_route;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 35606ae60868..58ffb6de400f 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = {
 #define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11)        | (1<<15))
 static const unsigned short sr_valid_mask[] = {
 	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 0*/
-	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
 	LOWER_GROUP,			/* Usb, bosr - 0*/
+	LOWER_GROUP|UPPER_GROUP,	/* Normal, bosr - 1*/
 	UPPER_GROUP,			/* Usb, bosr - 1*/
 };
 /*
@@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
-	int i;
 	u16 reg;
 
 	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+	for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
 		u16 val = tlv320aic23_read_reg_cache(codec, reg);
 		tlv320aic23_write(codec, reg, val);
 	}
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 9114c263077b..13aa380de162 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit);
 
 MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
 MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index ad219aaf7cb8..0cd06f5dd356 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
  *  |P| <--- TWL4030 <--------- Line In and MICs
  */
 static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
-	SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
 			   0, 0, NULL, 0, omap3pandora_hp_event,
 			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "CARKITR");
 	snd_soc_dapm_nc_pin(codec, "HFL");
 	snd_soc_dapm_nc_pin(codec, "HFR");
+	snd_soc_dapm_nc_pin(codec, "VIBRA");
 
 	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d89f6dc00908..66d4c165f99b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 			if (!w->power_check)
 				continue;
 
-			power = w->power_check(w);
-			if (power)
-				sys_power = 1;
+			/* If we're suspending then pull down all the 
+			 * power. */
+			switch (event) {
+			case SND_SOC_DAPM_STREAM_SUSPEND:
+				power = 0;
+				break;
+
+			default:
+				power = w->power_check(w);
+				if (power)
+					sys_power = 1;
+				break;
+			}
 
 			if (w->power == power)
 				continue;
@@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 		case SND_SOC_DAPM_STREAM_RESUME:
 			sys_power = 1;
 			break;
+		case SND_SOC_DAPM_STREAM_SUSPEND:
+			sys_power = 0;
+			break;
 		case SND_SOC_DAPM_STREAM_NOP:
 			sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+			break;
 		default:
 			break;
 		}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 9efcfd08d747..c998220b99c6 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -1071,6 +1071,15 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig
 	channels = (ftr[0] - 7) / csize - 1;
 
 	master_bits = snd_usb_combine_bytes(ftr + 6, csize);
+	/* master configuration quirks */
+	switch (state->chip->usb_id) {
+	case USB_ID(0x08bb, 0x2702):
+		snd_printk(KERN_INFO
+			   "usbmixer: master volume quirk for PCM2702 chip\n");
+		/* disable non-functional volume control */
+		master_bits &= ~(1 << (USB_FEATURE_VOLUME - 1));
+		break;
+	}
 	if (channels > 0)
 		first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize);
 	else