Merge branch 'fix/hda' into topic/hda

13 files changed, 118 insertions, 30 deletions

diff --git a/sound/core/jack.c b/sound/core/jack.c
index 4902ae56873..53b53e97c89 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -141,6 +141,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 
 fail_input:
 	input_free_device(jack->input_dev);
+	kfree(jack->id);
 	kfree(jack);
 	return err;
 }
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 23f49f356e0..16c0bdfbb16 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1252,11 +1252,19 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) {
 static int inline vortex_adbdma_getlinearpos(vortex_t * vortex, int adbdma)
 {
 	stream_t *dma = &vortex->dma_adb[adbdma];
-	int temp;
+	int temp, page, delta;
 
 	temp = hwread(vortex->mmio, VORTEX_ADBDMA_STAT + (adbdma << 2));
-	temp = (dma->period_virt * dma->period_bytes) + (temp & (dma->period_bytes - 1));
-	return temp;
+	page = (temp & ADB_SUBBUF_MASK) >> ADB_SUBBUF_SHIFT;
+	if (dma->nr_periods >= 4)
+		delta = (page - dma->period_real) & 3;
+	else {
+		delta = (page - dma->period_real);
+		if (delta < 0)
+			delta += dma->nr_periods;
+	}
+	return (dma->period_virt + delta) * dma->period_bytes
+		+ (temp & (dma->period_bytes - 1));
 }
 
 static void vortex_adbdma_startfifo(vortex_t * vortex, int adbdma)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0baffcdee8f..fcedad9a5fe 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2308,6 +2308,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x8410, "ASUS", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index fbe97d32140..4d5004e693f 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -3114,6 +3114,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO),
 	SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP),
 	SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS),
 	SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
@@ -3410,7 +3412,7 @@ static void cx_auto_parse_output(struct hda_codec *codec)
 		}
 	}
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.max_channels = nums * 2;
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
 	if (cfg->hp_outs > 0)
 		spec->auto_mute = 1;
@@ -3729,9 +3731,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	return 0;
 }
 
-static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
+static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
 			      const char *dir, int cidx,
-			      hda_nid_t nid, int hda_dir)
+			      hda_nid_t nid, int hda_dir, int amp_idx)
 {
 	static char name[32];
 	static struct snd_kcontrol_new knew[] = {
@@ -3743,7 +3745,8 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 
 	for (i = 0; i < 2; i++) {
 		struct snd_kcontrol *kctl;
-		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, hda_dir);
+		knew[i].private_value = HDA_COMPOSE_AMP_VAL(nid, 3, amp_idx,
+							    hda_dir);
 		knew[i].subdevice = HDA_SUBDEV_AMP_FLAG;
 		knew[i].index = cidx;
 		snprintf(name, sizeof(name), "%s%s %s", basename, dir, sfx[i]);
@@ -3759,6 +3762,9 @@ static int cx_auto_add_volume(struct hda_codec *codec, const char *basename,
 	return 0;
 }
 
+#define cx_auto_add_volume(codec, str, dir, cidx, nid, hda_dir)		\
+	cx_auto_add_volume_idx(codec, str, dir, cidx, nid, hda_dir, 0)
+
 #define cx_auto_add_pb_volume(codec, nid, str, idx)			\
 	cx_auto_add_volume(codec, str, " Playback", idx, nid, HDA_OUTPUT)
 
@@ -3808,29 +3814,60 @@ static int cx_auto_build_input_controls(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	static const char *prev_label;
-	int i, err, cidx;
+	int i, err, cidx, conn_len;
+	hda_nid_t conn[HDA_MAX_CONNECTIONS];
+
+	int multi_adc_volume = 0; /* If the ADC nid has several input volumes */
+	int adc_nid = spec->adc_nids[0];
+
+	conn_len = snd_hda_get_connections(codec, adc_nid, conn,
+					   HDA_MAX_CONNECTIONS);
+	if (conn_len < 0)
+		return conn_len;
+
+	multi_adc_volume = cfg->num_inputs > 1 && conn_len > 1;
+	if (!multi_adc_volume) {
+		err = cx_auto_add_volume(codec, "Capture", "", 0, adc_nid,
+					 HDA_INPUT);
+		if (err < 0)
+			return err;
+	}
 
-	err = cx_auto_add_volume(codec, "Capture", "", 0, spec->adc_nids[0],
-				 HDA_INPUT);
-	if (err < 0)
-		return err;
 	prev_label = NULL;
 	cidx = 0;
 	for (i = 0; i < cfg->num_inputs; i++) {
 		hda_nid_t nid = cfg->inputs[i].pin;
 		const char *label;
-		if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+		int j;
+		int pin_amp = get_wcaps(codec, nid) & AC_WCAP_IN_AMP;
+		if (!pin_amp && !multi_adc_volume)
 			continue;
+
 		label = hda_get_autocfg_input_label(codec, cfg, i);
 		if (label == prev_label)
 			cidx++;
 		else
 			cidx = 0;
 		prev_label = label;
-		err = cx_auto_add_volume(codec, label, " Capture", cidx,
-					 nid, HDA_INPUT);
-		if (err < 0)
-			return err;
+
+		if (pin_amp) {
+			err = cx_auto_add_volume(codec, label, " Boost", cidx,
+						 nid, HDA_INPUT);
+			if (err < 0)
+				return err;
+		}
+
+		if (!multi_adc_volume)
+			continue;
+		for (j = 0; j < conn_len; j++) {
+			if (conn[j] == nid) {
+				err = cx_auto_add_volume_idx(codec, label,
+				    " Capture", cidx, adc_nid, HDA_INPUT, j);
+				if (err < 0)
+					return err;
+				break;
+			}
+		}
 	}
 	return 0;
 }
@@ -3902,6 +3939,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_cxt5066 },
 	{ .id = 0x14f15069, .name = "CX20585",
 	  .patch = patch_cxt5066 },
+	{ .id = 0x14f1506e, .name = "CX20590",
+	  .patch = patch_cxt5066 },
 	{ .id = 0x14f15097, .name = "CX20631",
 	  .patch = patch_conexant_auto },
 	{ .id = 0x14f15098, .name = "CX20632",
@@ -3928,6 +3967,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066");
 MODULE_ALIAS("snd-hda-codec-id:14f15067");
 MODULE_ALIAS("snd-hda-codec-id:14f15068");
 MODULE_ALIAS("snd-hda-codec-id:14f15069");
+MODULE_ALIAS("snd-hda-codec-id:14f1506e");
 MODULE_ALIAS("snd-hda-codec-id:14f15097");
 MODULE_ALIAS("snd-hda-codec-id:14f15098");
 MODULE_ALIAS("snd-hda-codec-id:14f150a1");
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index a76c3260d94..63b0054200a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -567,7 +567,7 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = cfg->inputs[i].pin;
 		if (spec->smart51_enabled && is_smart51_pins(spec, nid))
 			ctl = PIN_OUT;
-		else if (i == AUTO_PIN_MIC)
+		else if (cfg->inputs[i].type == AUTO_PIN_MIC)
 			ctl = PIN_VREF50;
 		else
 			ctl = PIN_IN;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 3351f77607b..37b8aa8a680 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
 	{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
 	/* AIF3 output */
@@ -1578,6 +1584,13 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{ "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+	{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
+	{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
+	{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+	{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
 	{ "AIF2DACL", NULL, "AIF2DAC Mux" },
 	{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -3129,6 +3142,11 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 	case WM8994:
 		snd_soc_dapm_add_routes(dapm, wm8994_intercon,
 					ARRAY_SIZE(wm8994_intercon));
+
+		if (wm8994->revision < 4)
+			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+						ARRAY_SIZE(wm8994_revd_intercon));
+			
 		break;
 	case WM8958:
 		snd_soc_dapm_add_routes(dapm, wm8958_intercon,
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index b36f0b39b09..fe7984221eb 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = {
 		.ops = &evm_spdif_ops,
 	},
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcasp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-0018",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
 	.cpu_dai_name= "davinci-mcasp.0",
@@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = {
 
 static struct snd_soc_card da830_snd_soc_card = {
 	.name = "DA830/OMAP-L137 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da830_evm_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
 	.name = "DA850/OMAP-L138 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da850_evm_dai,
 	.num_links = 1,
 };
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index c4b60610beb..c3f6f1e7279 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1449,6 +1449,7 @@ static int soc_post_component_init(struct snd_soc_card *card,
 		rtd = &card->rtd_aux[num];
 		name = aux_dev->name;
 	}
+	rtd->card = card;
 
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
@@ -1471,7 +1472,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 
 	/* register the rtd device */
 	rtd->codec = codec;
-	rtd->card = card;
 	rtd->dev.parent = card->dev;
 	rtd->dev.release = rtd_release;
 	rtd->dev.init_name = name;
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 68b97477577..66eabafb1c2 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -785,7 +785,7 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	}
 
 	dev->pcm->private_data = dev;
-	strcpy(dev->pcm->name, dev->product_name);
+	strlcpy(dev->pcm->name, dev->product_name, sizeof(dev->pcm->name));
 
 	memset(dev->sub_playback, 0, sizeof(dev->sub_playback));
 	memset(dev->sub_capture, 0, sizeof(dev->sub_capture));
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2f218c77fff..a1a47088fd0 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -136,7 +136,7 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *device)
 	if (ret < 0)
 		return ret;
 
-	strcpy(rmidi->name, device->product_name);
+	strlcpy(rmidi->name, device->product_name, sizeof(rmidi->name));
 
 	rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX;
 	rmidi->private_data = device;
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 800f7cb4f25..c0f8270bc19 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -323,6 +323,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
 		return -ENOMEM;
 	}
 
+	mutex_init(&chip->shutdown_mutex);
 	chip->index = idx;
 	chip->dev = dev;
 	chip->card = card;
@@ -531,6 +532,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
 	chip = ptr;
 	card = chip->card;
 	mutex_lock(&register_mutex);
+	mutex_lock(&chip->shutdown_mutex);
 	chip->shutdown = 1;
 	chip->num_interfaces--;
 	if (chip->num_interfaces <= 0) {
@@ -548,9 +550,11 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
 			snd_usb_mixer_disconnect(p);
 		}
 		usb_chip[chip->index] = NULL;
+		mutex_unlock(&chip->shutdown_mutex);
 		mutex_unlock(&register_mutex);
 		snd_card_free_when_closed(card);
 	} else {
+		mutex_unlock(&chip->shutdown_mutex);
 		mutex_unlock(&register_mutex);
 	}
 }
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 4132522ac90..e3f680526cb 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -361,6 +361,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (changed) {
+		mutex_lock(&subs->stream->chip->shutdown_mutex);
 		/* format changed */
 		snd_usb_release_substream_urbs(subs, 0);
 		/* influenced: period_bytes, channels, rate, format, */
@@ -368,6 +369,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 						  params_rate(hw_params),
 						  snd_pcm_format_physical_width(params_format(hw_params)) *
 							params_channels(hw_params));
+		mutex_unlock(&subs->stream->chip->shutdown_mutex);
 	}
 
 	return ret;
@@ -385,8 +387,9 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream)
 	subs->cur_audiofmt = NULL;
 	subs->cur_rate = 0;
 	subs->period_bytes = 0;
-	if (!subs->stream->chip->shutdown)
-		snd_usb_release_substream_urbs(subs, 0);
+	mutex_lock(&subs->stream->chip->shutdown_mutex);
+	snd_usb_release_substream_urbs(subs, 0);
+	mutex_unlock(&subs->stream->chip->shutdown_mutex);
 	return snd_pcm_lib_free_vmalloc_buffer(substream);
 }
 
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index db3eb21627e..6e66fffe87f 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -36,6 +36,7 @@ struct snd_usb_audio {
 	struct snd_card *card;
 	u32 usb_id;
 	int shutdown;
+	struct mutex shutdown_mutex;
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
 	int num_interfaces;
 	int num_suspended_intf;