Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This update contains overall only driver-specific fixes.  Slightly
  large LOC are seen in usb-audio driver for a couple of new device
  quirks and cs42l71 ASoC driver for enhanced features.  The others are
  a few small (regression) fixes HD-audio, and yet other small / trival
  ASoC fixes."

* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
  ALSA: HDA: Fix sound resume hang
  ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
  ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
  ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
  ASoC: atmel-ssc: change disable to disable in dts node
  ASoC: Prevent pop_wait overwrite
  ALSA: usb-audio: ignore-quirk for HP Wireless Audio
  ALSA: hda - Always turn on pins for HDMI/DP
  ALSA: hda - Fix pin configuration of HP Pavilion dv7
  ASoC: core: Fix splitting of log messages
  ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
  ASoC: cs42l73: Add DAPM events for power down.
  ASoC: cs42l73: Add DMIC's as DAPM inputs.
  ASoC: sigmadsp: Fix endianness conversion issue
  ASoC: tpa6130a2: Use devm_* APIs

19 files changed, 371 insertions, 82 deletions

diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi
index 8e6251f1f7a3..32ec62cf5385 100644
--- a/arch/arm/boot/dts/at91sam9263.dtsi
+++ b/arch/arm/boot/dts/at91sam9263.dtsi
@@ -368,14 +368,14 @@
 				compatible = "atmel,at91rm9200-ssc";
 				reg = <0xfff98000 0x4000>;
 				interrupts = <16 4 5>;
-				status = "disable";
+				status = "disabled";
 			};
 
 			ssc1: ssc@fff9c000 {
 				compatible = "atmel,at91rm9200-ssc";
 				reg = <0xfff9c000 0x4000>;
 				interrupts = <17 4 5>;
-				status = "disable";
+				status = "disabled";
 			};
 
 			macb0: ethernet@fffbc000 {
diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi
index fa1ae0c5479c..231858ffd850 100644
--- a/arch/arm/boot/dts/at91sam9g45.dtsi
+++ b/arch/arm/boot/dts/at91sam9g45.dtsi
@@ -425,14 +425,14 @@
 				compatible = "atmel,at91sam9g45-ssc";
 				reg = <0xfff9c000 0x4000>;
 				interrupts = <16 4 5>;
-				status = "disable";
+				status = "disabled";
 			};
 
 			ssc1: ssc@fffa0000 {
 				compatible = "atmel,at91sam9g45-ssc";
 				reg = <0xfffa0000 0x4000>;
 				interrupts = <17 4 5>;
-				status = "disable";
+				status = "disabled";
 			};
 
 			adc0: adc@fffb0000 {
diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi
index 617ede541ca2..40ac3a4eb1ab 100644
--- a/arch/arm/boot/dts/at91sam9x5.dtsi
+++ b/arch/arm/boot/dts/at91sam9x5.dtsi
@@ -92,7 +92,7 @@
 				compatible = "atmel,at91sam9g45-ssc";
 				reg = <0xf0010000 0x4000>;
 				interrupts = <28 4 5>;
-				status = "disable";
+				status = "disabled";
 			};
 
 			tcb0: timer@f8008000 {
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 628db7bca4fd..3953cea0ecfb 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -242,7 +242,6 @@ struct snd_soc_dai {
 	unsigned int symmetric_rates:1;
 	struct snd_pcm_runtime *runtime;
 	unsigned int active;
-	unsigned char pop_wait:1;
 	unsigned char probed:1;
 
 	struct snd_soc_dapm_widget *playback_widget;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 91244a096c19..769e27c774a3 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1039,6 +1039,7 @@ struct snd_soc_pcm_runtime {
 	struct snd_soc_dpcm_runtime dpcm[2];
 
 	long pmdown_time;
+	unsigned char pop_wait:1;
 
 	/* runtime devices */
 	struct snd_pcm *pcm;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 0f3d3db0df71..cca87277baf0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2876,7 +2876,7 @@ static int azx_free(struct azx *chip)
 	azx_notifier_unregister(chip);
 
 	chip->init_failed = 1; /* to be sure */
-	complete(&chip->probe_wait);
+	complete_all(&chip->probe_wait);
 
 	if (use_vga_switcheroo(chip)) {
 		if (chip->disabled && chip->bus)
@@ -3504,7 +3504,7 @@ static int azx_probe(struct pci_dev *pci,
 		pm_runtime_put_noidle(&pci->dev);
 
 	dev++;
-	complete(&chip->probe_wait);
+	complete_all(&chip->probe_wait);
 	return 0;
 
 out_free:
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 0fcfa6f406b8..b6c21ea187ca 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -431,9 +431,11 @@ static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid)
 	if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP)
 		snd_hda_codec_write(codec, pin_nid, 0,
 				AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
-	/* Disable pin out until stream is active*/
+	/* Enable pin out: some machines with GM965 gets broken output when
+	 * the pin is disabled or changed while using with HDMI
+	 */
 	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
 }
 
 static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid)
@@ -1341,7 +1343,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx = hinfo_to_pin_index(spec, hinfo);
 	hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid;
-	int pinctl;
 	bool non_pcm;
 
 	non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
@@ -1350,11 +1351,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 
 	hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream);
 
-	pinctl = snd_hda_codec_read(codec, pin_nid, 0,
-				    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT);
-
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
 
@@ -1374,7 +1370,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 	int cvt_idx, pin_idx;
 	struct hdmi_spec_per_cvt *per_cvt;
 	struct hdmi_spec_per_pin *per_pin;
-	int pinctl;
 
 	if (hinfo->nid) {
 		cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid);
@@ -1391,11 +1386,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 			return -EINVAL;
 		per_pin = &spec->pins[pin_idx];
 
-		pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
-					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_codec_write(codec, per_pin->pin_nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL,
-				    pinctl & ~PIN_OUT);
 		snd_hda_spdif_ctls_unassign(codec, pin_idx);
 		per_pin->chmap_set = false;
 		memset(per_pin->chmap, 0, sizeof(per_pin->chmap));
@@ -1691,6 +1681,30 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = {
 	.unsol_event		= hdmi_unsol_event,
 };
 
+static void intel_haswell_fixup_connect_list(struct hda_codec *codec)
+{
+	unsigned int vendor_param;
+	hda_nid_t list[3] = {0x2, 0x3, 0x4};
+
+	vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0);
+	if (vendor_param == -1 || vendor_param & 0x02)
+		return;
+
+	/* enable DP1.2 mode */
+	vendor_param |= 0x02;
+	snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param);
+
+	vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0);
+	if (vendor_param == -1 || !(vendor_param & 0x02))
+		return;
+
+	/* override 3 pins connection list */
+	snd_hda_override_conn_list(codec, 0x05, 3, list);
+	snd_hda_override_conn_list(codec, 0x06, 3, list);
+	snd_hda_override_conn_list(codec, 0x07, 3, list);
+}
+
+
 static int patch_generic_hdmi(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
@@ -1700,6 +1714,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -ENOMEM;
 
 	codec->spec = spec;
+
+	if (codec->vendor_id == 0x80862807)
+		intel_haswell_fixup_connect_list(codec);
+
 	if (hdmi_parse_codec(codec) < 0) {
 		codec->spec = NULL;
 		kfree(spec);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7743775f6abb..6ee34593774a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4373,6 +4373,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->single_adc_amp = 1;
 	spec->mixer_nid = mixer_nid;
 	snd_hda_gen_init(&spec->gen);
 	snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32);
@@ -6569,8 +6570,8 @@ static void alc861vd_fixup_dallas(struct hda_codec *codec,
 				  const struct alc_fixup *fix, int action)
 {
 	if (action == ALC_FIXUP_ACT_PRE_PROBE) {
-		snd_hda_override_pin_caps(codec, 0x18, 0x00001714);
-		snd_hda_override_pin_caps(codec, 0x19, 0x0000171c);
+		snd_hda_override_pin_caps(codec, 0x18, 0x00000734);
+		snd_hda_override_pin_caps(codec, 0x19, 0x0000073c);
 	}
 }
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index df13c0f84899..a86547ca17c8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1725,7 +1725,7 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1658,
 			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1659,
-			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+			  "HP Pavilion dv7", STAC_HP_DV7_4000),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165A,
 			  "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165B,
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index a0791ecf6d95..6361dab48bd1 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -40,6 +40,7 @@ struct  cs42l73_private {
 	u32 sysclk;
 	u8 mclksel;
 	u32 mclk;
+	int shutdwn_delay;
 };
 
 static const struct reg_default cs42l73_reg_defaults[] = {
@@ -588,7 +589,60 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = {
 	SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum),
 };
 
+static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		/* 150 ms delay between setting PDN and MCLKDIS */
+		priv->shutdwn_delay = 150;
+		break;
+	default:
+		pr_err("Invalid event = 0x%x\n", event);
+	}
+	return 0;
+}
+
+static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		/* 50 ms delay between setting PDN and MCLKDIS */
+		if (priv->shutdwn_delay < 50)
+			priv->shutdwn_delay = 50;
+		break;
+	default:
+		pr_err("Invalid event = 0x%x\n", event);
+	}
+	return 0;
+}
+
+
+static int cs42l73_hp_amp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		/* 30 ms delay between setting PDN and MCLKDIS */
+		if (priv->shutdwn_delay < 30)
+			priv->shutdwn_delay = 30;
+		break;
+	default:
+		pr_err("Invalid event = 0x%x\n", event);
+	}
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("DMICA"),
+	SND_SOC_DAPM_INPUT("DMICB"),
 	SND_SOC_DAPM_INPUT("LINEINA"),
 	SND_SOC_DAPM_INPUT("LINEINB"),
 	SND_SOC_DAPM_INPUT("MIC1"),
@@ -604,9 +658,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
 			CS42L73_PWRCTL2, 3, 1),
 	SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL,  0,
 			CS42L73_PWRCTL2, 3, 1),
-	SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL,  0,
-			CS42L73_PWRCTL2, 4, 1),
-	SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL,  0,
+	SND_SOC_DAPM_AIF_OUT("VSPINOUT", NULL,  0,
 			CS42L73_PWRCTL2, 4, 1),
 
 	SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -632,8 +684,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
 	SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-	SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("VSP Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0,
 				CS42L73_PWRCTL2, 0, 1),
@@ -649,7 +700,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0,
 				CS42L73_PWRCTL2, 2, 1),
 
-	SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0,
+	SND_SOC_DAPM_AIF_IN("VSPINOUT", NULL, 0,
 				CS42L73_PWRCTL2, 4, 1),
 
 	SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
@@ -674,16 +725,20 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0),
 
-	SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1,
-			    &hp_amp_ctl),
+	SND_SOC_DAPM_SWITCH_E("HP Amp",  CS42L73_PWRCTL3, 0, 1,
+			    &hp_amp_ctl, cs42l73_hp_amp_event,
+			SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1,
 			    &lo_amp_ctl),
-	SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1,
-			    &spk_amp_ctl),
-	SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1,
-			    &ear_amp_ctl),
-	SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
-			    &spklo_amp_ctl),
+	SND_SOC_DAPM_SWITCH_E("SPK Amp", CS42L73_PWRCTL3, 2, 1,
+			&spk_amp_ctl, cs42l73_spklo_spk_amp_event,
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH_E("EAR Amp", CS42L73_PWRCTL3, 3, 1,
+			    &ear_amp_ctl, cs42l73_ear_amp_event,
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SWITCH_E("SPKLO Amp", CS42L73_PWRCTL3, 4, 1,
+			    &spklo_amp_ctl, cs42l73_spklo_spk_amp_event,
+			SND_SOC_DAPM_POST_PMD),
 
 	SND_SOC_DAPM_OUTPUT("HPOUTA"),
 	SND_SOC_DAPM_OUTPUT("HPOUTB"),
@@ -705,7 +760,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 
 	{"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"},
 	{"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"},
-	{"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"},
+	{"ESL DAC", "ESL-VSP Mono Volume", "VSPINOUT"},
 	/* Loopback */
 	{"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"},
 	{"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"},
@@ -727,7 +782,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 
 	{"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"},
 	{"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"},
-	{"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"},
+	{"SPK DAC", "SPK-VSP Mono Volume", "VSPINOUT"},
 	/* Loopback */
 	{"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"},
 	{"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"},
@@ -770,8 +825,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 	{"HL Right Mixer", NULL, "ASPINR"},
 	{"HL Left Mixer", NULL, "XSPINL"},
 	{"HL Right Mixer", NULL, "XSPINR"},
-	{"HL Left Mixer", NULL, "VSPIN"},
-	{"HL Right Mixer", NULL, "VSPIN"},
+	{"HL Left Mixer", NULL, "VSPINOUT"},
+	{"HL Right Mixer", NULL, "VSPINOUT"},
 
 	{"ASPINL", NULL, "ASP Playback"},
 	{"ASPINM", NULL, "ASP Playback"},
@@ -779,7 +834,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 	{"XSPINL", NULL, "XSP Playback"},
 	{"XSPINM", NULL, "XSP Playback"},
 	{"XSPINR", NULL, "XSP Playback"},
-	{"VSPIN", NULL, "VSP Playback"},
+	{"VSPINOUT", NULL, "VSP Playback"},
 
 	/* Capture Paths */
 	{"MIC1", NULL, "MIC1 Bias"},
@@ -795,6 +850,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 
 	{"ADC Left", NULL, "PGA Left"},
 	{"ADC Right", NULL, "PGA Right"},
+	{"DMIC Left", NULL, "DMICA"},
+	{"DMIC Right", NULL, "DMICB"},
 
 	{"Input Left Capture", "ADC Left Input", "ADC Left"},
 	{"Input Right Capture", "ADC Right Input", "ADC Right"},
@@ -819,21 +876,18 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
 	{"XSPOUTR", NULL, "XSPR Output Mixer"},
 
 	/* Voice Capture */
-	{"VSPL Output Mixer", NULL, "Input Left Capture"},
-	{"VSPR Output Mixer", NULL, "Input Left Capture"},
+	{"VSP Output Mixer", NULL, "Input Left Capture"},
+	{"VSP Output Mixer", NULL, "Input Right Capture"},
 
-	{"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"},
-	{"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"},
+	{"VSPINOUT", "VSP-IP Volume", "VSP Output Mixer"},
 
-	{"VSPOUTL", NULL, "VSPL Output Mixer"},
-	{"VSPOUTR", NULL, "VSPR Output Mixer"},
+	{"VSPINOUT", NULL, "VSP Output Mixer"},
 
 	{"ASP Capture", NULL, "ASPOUTL"},
 	{"ASP Capture", NULL, "ASPOUTR"},
 	{"XSP Capture", NULL, "XSPOUTL"},
 	{"XSP Capture", NULL, "XSPOUTR"},
-	{"VSP Capture", NULL, "VSPOUTL"},
-	{"VSP Capture", NULL, "VSPOUTR"},
+	{"VSP Capture", NULL, "VSPINOUT"},
 };
 
 struct cs42l73_mclk_div {
@@ -1167,6 +1221,14 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_OFF:
 		snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1);
+		if (cs42l73->shutdwn_delay > 0) {
+			mdelay(cs42l73->shutdwn_delay);
+			cs42l73->shutdwn_delay = 0;
+		} else {
+			mdelay(15); /* Min amount of time requred to power
+				     * down.
+				     */
+		}
 		snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1);
 		break;
 	}
diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c
index 5be42bf56996..4068f2491232 100644
--- a/sound/soc/codecs/sigmadsp.c
+++ b/sound/soc/codecs/sigmadsp.c
@@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware);
 static int sigma_action_write_regmap(void *control_data,
 	const struct sigma_action *sa, size_t len)
 {
-	return regmap_raw_write(control_data, le16_to_cpu(sa->addr),
+	return regmap_raw_write(control_data, be16_to_cpu(sa->addr),
 		sa->payload, len - 2);
 }
 
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 8d75aa152c8c..c58bee8346ce 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -398,7 +398,8 @@ static int tpa6130a2_probe(struct i2c_client *client,
 						TPA6130A2_MUTE_L;
 
 	if (data->power_gpio >= 0) {
-		ret = gpio_request(data->power_gpio, "tpa6130a2 enable");
+		ret = devm_gpio_request(dev, data->power_gpio,
+					"tpa6130a2 enable");
 		if (ret < 0) {
 			dev_err(dev, "Failed to request power GPIO (%d)\n",
 				data->power_gpio);
@@ -419,16 +420,16 @@ static int tpa6130a2_probe(struct i2c_client *client,
 		break;
 	}
 
-	data->supply = regulator_get(dev, regulator);
+	data->supply = devm_regulator_get(dev, regulator);
 	if (IS_ERR(data->supply)) {
 		ret = PTR_ERR(data->supply);
 		dev_err(dev, "Failed to request supply: %d\n", ret);
-		goto err_regulator;
+		goto err_gpio;
 	}
 
 	ret = tpa6130a2_power(1);
 	if (ret != 0)
-		goto err_power;
+		goto err_gpio;
 
 
 	/* Read version */
@@ -440,15 +441,10 @@ static int tpa6130a2_probe(struct i2c_client *client,
 	/* Disable the chip */
 	ret = tpa6130a2_power(0);
 	if (ret != 0)
-		goto err_power;
+		goto err_gpio;
 
 	return 0;
 
-err_power:
-	regulator_put(data->supply);
-err_regulator:
-	if (data->power_gpio >= 0)
-		gpio_free(data->power_gpio);
 err_gpio:
 	tpa6130a2_client = NULL;
 
@@ -457,14 +453,7 @@ err_gpio:
 
 static int tpa6130a2_remove(struct i2c_client *client)
 {
-	struct tpa6130a2_data *data = i2c_get_clientdata(client);
-
 	tpa6130a2_power(0);
-
-	if (data->power_gpio >= 0)
-		gpio_free(data->power_gpio);
-
-	regulator_put(data->supply);
 	tpa6130a2_client = NULL;
 
 	return 0;
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 967d0e173e1b..5fbfb06e8083 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream)
 					SNDRV_PCM_STREAM_PLAYBACK,
 					SND_SOC_DAPM_STREAM_STOP);
 		} else
-			codec_dai->pop_wait = 1;
+			rtd->pop_wait = 1;
 			schedule_delayed_work(&rtd->delayed_work,
 				msecs_to_jiffies(rtd->pmdown_time));
 	} else {
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 9c768bcb98a6..91d592ff67b7 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -4155,9 +4155,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 		ret = of_property_read_string_index(np, propname,
 			2 * i, &routes[i].sink);
 		if (ret) {
-			dev_err(card->dev, "ASoC: Property '%s' index %d"
-				" could not be read: %d\n", propname, 2 * i,
-				ret);
+			dev_err(card->dev,
+				"ASoC: Property '%s' index %d could not be read: %d\n",
+				propname, 2 * i, ret);
 			kfree(routes);
 			return -EINVAL;
 		}
@@ -4165,8 +4165,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 			(2 * i) + 1, &routes[i].source);
 		if (ret) {
 			dev_err(card->dev,
-				"ASoC: Property '%s' index %d could not be"
-				" read: %d\n", propname, (2 * i) + 1, ret);
+				"ASoC: Property '%s' index %d could not be read: %d\n",
+				propname, (2 * i) + 1, ret);
 			kfree(routes);
 			return -EINVAL;
 		}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 5c3ca2a34661..d7711fce119b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work)
 	dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n",
 		 codec_dai->driver->playback.stream_name,
 		 codec_dai->playback_active ? "active" : "inactive",
-		 codec_dai->pop_wait ? "yes" : "no");
+		 rtd->pop_wait ? "yes" : "no");
 
 	/* are we waiting on this codec DAI stream */
-	if (codec_dai->pop_wait == 1) {
-		codec_dai->pop_wait = 0;
+	if (rtd->pop_wait == 1) {
+		rtd->pop_wait = 0;
 		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
 					  SND_SOC_DAPM_STREAM_STOP);
 	}
@@ -408,7 +408,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 						  SND_SOC_DAPM_STREAM_STOP);
 		} else {
 			/* start delayed pop wq here for playback streams */
-			codec_dai->pop_wait = 1;
+			rtd->pop_wait = 1;
 			schedule_delayed_work(&rtd->delayed_work,
 				msecs_to_jiffies(rtd->pmdown_time));
 		}
@@ -480,8 +480,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 
 	/* cancel any delayed stream shutdown that is pending */
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
-	    codec_dai->pop_wait) {
-		codec_dai->pop_wait = 0;
+	    rtd->pop_wait) {
+		rtd->pop_wait = 0;
 		cancel_delayed_work(&rtd->delayed_work);
 	}
 
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 34b9bb7fe87c..c183d34842ac 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -2181,6 +2181,10 @@ int snd_usbmidi_create(struct snd_card *card,
 		umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
 		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
 		break;
+	case QUIRK_MIDI_MBOX2:
+		umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		break;
 	case QUIRK_MIDI_RAW_BYTES:
 		umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
 		/*
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 49f9af995d7a..cdcf6b45e8a8 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -99,6 +99,42 @@
 },
 
 /*
+ * HP Wireless Audio
+ * When not ignored, causes instability issues for some users, forcing them to
+ * blacklist the entire module.
+ */
+{
+	USB_DEVICE(0x0424, 0xb832),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Standard Microsystems Corp.",
+		.product_name = "HP Wireless Audio",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			/* Mixer */
+			{
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE,
+			},
+			/* Playback */
+			{
+				.ifnum = 1,
+				.type = QUIRK_IGNORE_INTERFACE,
+			},
+			/* Capture */
+			{
+				.ifnum = 2,
+				.type = QUIRK_IGNORE_INTERFACE,
+			},
+			/* HID Device, .ifnum = 3 */
+			{
+				.ifnum = -1,
+			}
+		}
+	}
+},
+
+/*
  * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
  * class matches do not take effect without an explicit ID match.
  */
@@ -2885,6 +2921,93 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 
 	}
 },
+
+/* DIGIDESIGN MBOX 2 */
+{
+	USB_DEVICE(0x0dba, 0x3000),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.vendor_name = "Digidesign",
+		.product_name = "Mbox 2",
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = (const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 0,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 1,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S24_3BE,
+					.channels = 2,
+					.iface = 2,
+					.altsetting = 2,
+					.altset_idx = 1,
+					.attributes = 0x00,
+					.endpoint = 0x03,
+					.ep_attr = USB_ENDPOINT_SYNC_ASYNC,
+					.maxpacksize = 0x128,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) {
+						48000
+					}
+				}
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 4,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+				.formats = SNDRV_PCM_FMTBIT_S24_3BE,
+					.channels = 2,
+					.iface = 4,
+					.altsetting = 2,
+					.altset_idx = 1,
+					.attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
+					.endpoint = 0x85,
+					.ep_attr = USB_ENDPOINT_SYNC_SYNC,
+					.maxpacksize = 0x128,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) {
+						48000
+					}
+				}
+			},
+			{
+				.ifnum = 5,
+				.type = QUIRK_IGNORE_INTERFACE
+			},
+			{
+				.ifnum = 6,
+				.type = QUIRK_MIDI_MBOX2,
+				.data = &(const struct snd_usb_midi_endpoint_info) {
+					.out_ep =  0x02,
+					.out_cables = 0x0001,
+					.in_ep = 0x81,
+					.in_interval = 0x01,
+					.in_cables = 0x0001
+				}
+			},
+			{
+				.ifnum = -1
+			}
+		}
+	}
+},
 {
 	/* Tascam US122 MKII - playback-only support */
 	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 007fcecdf5cd..f104c68fe1e0 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -306,6 +306,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_MIDI_YAMAHA] = create_any_midi_quirk,
 		[QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk,
 		[QUIRK_MIDI_NOVATION] = create_any_midi_quirk,
+		[QUIRK_MIDI_MBOX2] = create_any_midi_quirk,
 		[QUIRK_MIDI_RAW_BYTES] = create_any_midi_quirk,
 		[QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
 		[QUIRK_MIDI_CME] = create_any_midi_quirk,
@@ -497,6 +498,92 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev)
 	return -EAGAIN;
 }
 
+static void mbox2_setup_48_24_magic(struct usb_device *dev)
+{
+	u8 srate[3];
+	u8 temp[12];
+
+	/* Choose 48000Hz permanently */
+	srate[0] = 0x80;
+	srate[1] = 0xbb;
+	srate[2] = 0x00;
+
+	/* Send the magic! */
+	snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+		0x01, 0x22, 0x0100, 0x0085, &temp, 0x0003);
+	snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+		0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003);
+	snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+		0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003);
+	snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0),
+		0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003);
+	return;
+}
+
+/* Digidesign Mbox 2 needs to load firmware onboard
+ * and driver must wait a few seconds for initialisation.
+ */
+
+#define MBOX2_FIRMWARE_SIZE    646
+#define MBOX2_BOOT_LOADING     0x01 /* Hard coded into the device */
+#define MBOX2_BOOT_READY       0x02 /* Hard coded into the device */
+
+int snd_usb_mbox2_boot_quirk(struct usb_device *dev)
+{
+	struct usb_host_config *config = dev->actconfig;
+	int err;
+	u8 bootresponse;
+	int fwsize;
+	int count;
+
+	fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength);
+
+	if (fwsize != MBOX2_FIRMWARE_SIZE) {
+		snd_printk(KERN_ERR "usb-audio: Invalid firmware size=%d.\n", fwsize);
+		return -ENODEV;
+	}
+
+	snd_printd("usb-audio: Sending Digidesign Mbox 2 boot sequence...\n");
+
+	count = 0;
+	bootresponse = MBOX2_BOOT_LOADING;
+	while ((bootresponse == MBOX2_BOOT_LOADING) && (count < 10)) {
+		msleep(500); /* 0.5 second delay */
+		snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0),
+			/* Control magic - load onboard firmware */
+			0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012);
+		if (bootresponse == MBOX2_BOOT_READY)
+			break;
+		snd_printd("usb-audio: device not ready, resending boot sequence...\n");
+		count++;
+	}
+
+	if (bootresponse != MBOX2_BOOT_READY) {
+		snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse);
+		return -ENODEV;
+	}
+
+	snd_printdd("usb-audio: device initialised!\n");
+
+	err = usb_get_descriptor(dev, USB_DT_DEVICE, 0,
+		&dev->descriptor, sizeof(dev->descriptor));
+	config = dev->actconfig;
+	if (err < 0)
+		snd_printd("error usb_get_descriptor: %d\n", err);
+
+	err = usb_reset_configuration(dev);
+	if (err < 0)
+		snd_printd("error usb_reset_configuration: %d\n", err);
+	snd_printdd("mbox2_boot: new boot length = %d\n",
+		le16_to_cpu(get_cfg_desc(config)->wTotalLength));
+
+	mbox2_setup_48_24_magic(dev);
+
+	snd_printk(KERN_INFO "usb-audio: Digidesign Mbox 2: 24bit 48kHz");
+
+	return 0; /* Successful boot */
+}
+
 /*
  * Setup quirks
  */
@@ -655,6 +742,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev,
 	case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */
 		return snd_usb_cm6206_boot_quirk(dev);
 
+	case USB_ID(0x0dba, 0x3000):
+		/* Digidesign Mbox 2 */
+		return snd_usb_mbox2_boot_quirk(dev);
+
 	case USB_ID(0x133e, 0x0815):
 		/* Access Music VirusTI Desktop */
 		return snd_usb_accessmusic_boot_quirk(dev);
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 1ac3fd9cc5a6..a8172c119796 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -76,6 +76,7 @@ enum quirk_type {
 	QUIRK_MIDI_YAMAHA,
 	QUIRK_MIDI_MIDIMAN,
 	QUIRK_MIDI_NOVATION,
+	QUIRK_MIDI_MBOX2,
 	QUIRK_MIDI_RAW_BYTES,
 	QUIRK_MIDI_EMAGIC,
 	QUIRK_MIDI_CME,