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

6 files changed, 108 insertions, 57 deletions

diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index a0791ecf6d9..6361dab48bd 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 5be42bf5699..4068f249123 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 8d75aa152c8..c58bee8346c 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 967d0e173e1..5fbfb06e808 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 9c768bcb98a..91d592ff67b 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 5c3ca2a3466..d7711fce119 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);
 	}