Merge branch 'master' into for-3.9-async

To receive f56c3196f251012de9b3ebaff55732a9074fdaae ("async: fix
__lowest_in_progress()").

Signed-off-by: Tejun Heo <tj@kernel.org>

18 files changed, 274 insertions, 172 deletions

diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index adf397b9d0e6..1d8bb5917594 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_DSP_A:
 		mode = 0;
 		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		mode = 1;
-		break;
 	case SND_SOC_DAIFMT_I2S:
 		mode = 2;
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		mode = 3;
-		break;
 	default:
 		arizona_aif_err(dai, "Unsupported DAI format %d\n",
 				fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
 				    ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
 		snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
-				    ARIZONA_AIF1_RATE_MASK, 8);
+				    ARIZONA_AIF1_RATE_MASK,
+				    8 << ARIZONA_AIF1_RATE_SHIFT);
 		break;
 	default:
 		arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index 41dae1ed3b71..4deebeb07177 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -34,15 +34,15 @@
 
 #define ARIZONA_FLL_SRC_MCLK1      0
 #define ARIZONA_FLL_SRC_MCLK2      1
-#define ARIZONA_FLL_SRC_SLIMCLK    2
-#define ARIZONA_FLL_SRC_FLL1       3
-#define ARIZONA_FLL_SRC_FLL2       4
-#define ARIZONA_FLL_SRC_AIF1BCLK   5
-#define ARIZONA_FLL_SRC_AIF2BCLK   6
-#define ARIZONA_FLL_SRC_AIF3BCLK   7
-#define ARIZONA_FLL_SRC_AIF1LRCLK  8
-#define ARIZONA_FLL_SRC_AIF2LRCLK  9
-#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+#define ARIZONA_FLL_SRC_SLIMCLK    3
+#define ARIZONA_FLL_SRC_FLL1       4
+#define ARIZONA_FLL_SRC_FLL2       5
+#define ARIZONA_FLL_SRC_AIF1BCLK   8
+#define ARIZONA_FLL_SRC_AIF2BCLK   9
+#define ARIZONA_FLL_SRC_AIF3BCLK  10
+#define ARIZONA_FLL_SRC_AIF1LRCLK 12
+#define ARIZONA_FLL_SRC_AIF2LRCLK 13
+#define ARIZONA_FLL_SRC_AIF3LRCLK 14
 
 #define ARIZONA_MIXER_VOL_MASK             0x00FE
 #define ARIZONA_MIXER_VOL_SHIFT                 1
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 4f1127935fdf..ac8742a1f25a 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 	struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
 	int ret;
 	int gpio_nreset = -EINVAL;
-	int amutec_eq_bmutec = 0;
+	bool amutec_eq_bmutec = false;
 
 #ifdef CONFIG_OF
 	if (of_match_device(cs4271_dt_ids, codec->dev)) {
 		gpio_nreset = of_get_named_gpio(codec->dev->of_node,
 						"reset-gpio", 0);
 
-		if (!of_get_property(codec->dev->of_node,
+		if (of_get_property(codec->dev->of_node,
 				     "cirrus,amutec-eq-bmutec", NULL))
-			amutec_eq_bmutec = 1;
+			amutec_eq_bmutec = true;
 	}
 #endif
 
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 99bb1c69499e..9811a5478c87 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = {
 
 static int cs42l52_get_clk(int mclk, int rate)
 {
-	int i, ret = 0;
+	int i, ret = -EINVAL;
 	u_int mclk1, mclk2 = 0;
 
 	for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) {
@@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate)
 			}
 		}
 	}
-	if (ret > ARRAY_SIZE(clk_map_table))
-		return -EINVAL;
 	return ret;
 }
 
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/lm49453.c b/sound/soc/codecs/lm49453.c
index d75257d40a49..e19490cfb3a8 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = {
 	{ 101, 0x00 },
 	{ 102, 0x00 },
 	{ 103, 0x01 },
-	{ 105, 0x01 },
-	{ 106, 0x00 },
-	{ 107, 0x01 },
+	{ 104, 0x01 },
+	{ 105, 0x00 },
+	{ 106, 0x01 },
 	{ 107, 0x00 },
 	{ 108, 0x00 },
 	{ 109, 0x00 },
@@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = {
 	{ 184, 0x00 },
 	{ 185, 0x00 },
 	{ 186, 0x00 },
-	{ 189, 0x00 },
+	{ 187, 0x00 },
 	{ 188, 0x00 },
-	{ 194, 0x00 },
-	{ 195, 0x00 },
-	{ 196, 0x00 },
-	{ 197, 0x00 },
-	{ 200, 0x00 },
-	{ 201, 0x00 },
-	{ 202, 0x00 },
-	{ 203, 0x00 },
-	{ 204, 0x00 },
-	{ 205, 0x00 },
-	{ 208, 0x00 },
+	{ 189, 0x00 },
+	{ 208, 0x06 },
 	{ 209, 0x00 },
-	{ 210, 0x00 },
-	{ 211, 0x00 },
-	{ 213, 0x00 },
-	{ 214, 0x00 },
-	{ 215, 0x00 },
-	{ 216, 0x00 },
-	{ 217, 0x00 },
-	{ 218, 0x00 },
-	{ 219, 0x00 },
+	{ 210, 0x08 },
+	{ 211, 0x54 },
+	{ 212, 0x14 },
+	{ 213, 0x0d },
+	{ 214, 0x0d },
+	{ 215, 0x14 },
+	{ 216, 0x60 },
 	{ 221, 0x00 },
 	{ 222, 0x00 },
+	{ 223, 0x00 },
 	{ 224, 0x00 },
-	{ 225, 0x00 },
-	{ 226, 0x00 },
-	{ 227, 0x00 },
-	{ 228, 0x00 },
-	{ 229, 0x00 },
-	{ 230, 0x13 },
-	{ 231, 0x00 },
-	{ 232, 0x80 },
-	{ 233, 0x0C },
-	{ 234, 0xDD },
-	{ 235, 0x00 },
-	{ 236, 0x04 },
-	{ 237, 0x00 },
-	{ 238, 0x00 },
-	{ 239, 0x00 },
-	{ 240, 0x00 },
-	{ 241, 0x00 },
-	{ 242, 0x00 },
-	{ 243, 0x00 },
-	{ 244, 0x00 },
-	{ 245, 0x00 },
 	{ 248, 0x00 },
 	{ 249, 0x00 },
-	{ 254, 0x00 },
+	{ 250, 0x00 },
 	{ 255, 0x00 },
 };
 
@@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0),
 };
 
 /* TLV Declarations */
-static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1);
-static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1);
+static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0);
+static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0);
 
 static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = {
 /* Sidetone supports mono only */
 SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG,
-		     0, 0x3F, 0, digital_tlv),
+		     0, 0x3F, 0, stn_tlv),
 };
 
 static const struct snd_kcontrol_new lm49453_snd_controls[] = {
 	/* mic1 and mic2 supports mono only */
-	SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6,
-			0, digital_tlv),
-	SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6,
-			0, digital_tlv),
+	SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv),
+	SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv),
+
+	SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63,
+			0, adc_dac_tlv),
+	SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63,
+			0, adc_dac_tlv),
 
 	SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG,
-			  LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG,
-			  LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
 	SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum),
 	SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum),
@@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = {
 					  2, 1, 0),
 
 	SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG,
-			  LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG,
-			  LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-			  LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 	SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG,
-			  LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv),
+			  LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv),
 
 	SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG,
-			0, 6, 0, digital_tlv),
+			0, 63, 0, adc_dac_tlv),
 
 	SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG,
 			0, 3, 0, port_tlv),
@@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 	}
 
 	snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG,
-			    LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5),
+			    LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5),
 			    (aif_val | mode | clk_phase));
 
 	snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index cb1675cd8e1c..92bbfec9b107 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
 			5, 1, 0),
 
 	SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
-			0, 4, 0, mic_gain_tlv),
+			0, 3, 0, mic_gain_tlv),
 };
 
 /* mute the codec used by alsa core */
@@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			SGTL5000_HP_ZCD_EN |
 			SGTL5000_ADC_ZCD_EN);
 
-	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
 
 	/*
 	 * disable DAP
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/sta529.c b/sound/soc/codecs/sta529.c
index ab355c4f0b2d..40c07be9b581 100644
--- a/sound/soc/codecs/sta529.c
+++ b/sound/soc/codecs/sta529.c
@@ -74,9 +74,10 @@
 				SNDRV_PCM_FMTBIT_S32_LE)
 #define	S2PC_VALUE		0x98
 #define CLOCK_OUT		0x60
-#define LEFT_J_DATA_FORMAT	0x10
-#define I2S_DATA_FORMAT		0x12
-#define RIGHT_J_DATA_FORMAT	0x14
+#define DATA_FORMAT_MSK		0x0E
+#define LEFT_J_DATA_FORMAT	0x00
+#define I2S_DATA_FORMAT		0x02
+#define RIGHT_J_DATA_FORMAT	0x04
 #define CODEC_MUTE_VAL		0x80
 
 #define POWER_CNTLMSAK		0x40
@@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+	snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode);
 
 	return 0;
 }
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/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 1cbe88f01d63..12bcae63a7f0 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue)
 
 	ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY);
 	if (wm2000->speech_clarity)
-		ret &= ~WM2000_SPEECH_CLARITY;
-	else
 		ret |= WM2000_SPEECH_CLARITY;
+	else
+		ret &= ~WM2000_SPEECH_CLARITY;
 	wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret);
 
 	wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33);
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
index afcf31df77e0..e6cefe1ac677 100644
--- a/sound/soc/codecs/wm2200.c
+++ b/sound/soc/codecs/wm2200.c
@@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_DSP_A:
 		fmt_val = 0;
 		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		fmt_val = 1;
-		break;
 	case SND_SOC_DAIFMT_I2S:
 		fmt_val = 2;
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		fmt_val = 3;
-		break;
 	default:
 		dev_err(codec->dev, "Unsupported DAI format %d\n",
 			fmt & SND_SOC_DAIFMT_FORMAT_MASK);
@@ -1626,7 +1620,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 			    WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
 			    lrclk);
 	snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
-			    WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+			    WM2200_AIF1_FMT_MASK, fmt_val);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5a5f36936235..54397a508073 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	case SND_SOC_DAIFMT_DSP_A:
 		mask = 0;
 		break;
-	case SND_SOC_DAIFMT_DSP_B:
-		mask = 1;
-		break;
 	case SND_SOC_DAIFMT_I2S:
 		mask = 2;
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		mask = 3;
-		break;
 	default:
 		dev_err(codec->dev, "Unsupported DAI format %d\n",
 			fmt & SND_SOC_DAIFMT_FORMAT_MASK);
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index 688ade080589..7a9048dad1cd 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -36,6 +36,9 @@
 struct wm5102_priv {
 	struct arizona_priv core;
 	struct arizona_fll fll[2];
+
+	unsigned int spk_ena:2;
+	unsigned int spk_ena_pending:1;
 };
 
 static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
@@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
 };
 
+static int wm5102_spk_ev(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol,
+			 int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+	struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+	if (arizona->rev < 1)
+		return 0;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (!wm5102->spk_ena) {
+			snd_soc_write(codec, 0x4f5, 0x25a);
+			wm5102->spk_ena_pending = true;
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		if (wm5102->spk_ena_pending) {
+			msleep(75);
+			snd_soc_write(codec, 0x4f5, 0xda);
+			wm5102->spk_ena_pending = false;
+			wm5102->spk_ena++;
+		}
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		wm5102->spk_ena--;
+		if (!wm5102->spk_ena)
+			snd_soc_write(codec, 0x4f5, 0x25a);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		if (!wm5102->spk_ena)
+			snd_soc_write(codec, 0x4f5, 0x0da);
+		break;
+	}
+
+	return 0;
+}
+
+
 ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
 ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
@@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
 		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
-		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
 		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ffc89fab96fb..7b198c38f3ef 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 	const struct wm_adsp_region *mem;
 	const char *region_name;
 	char *file, *text;
+	void *buf;
 	unsigned int reg;
 	int regions = 0;
 	int ret, offset, type, sizes;
@@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp)
 		}
 
 		if (reg) {
-			ret = regmap_raw_write(regmap, reg, region->data,
+			buf = kmemdup(region->data, le32_to_cpu(region->len),
+				      GFP_KERNEL);
+			if (!buf) {
+				adsp_err(dsp, "Out of memory\n");
+				return -ENOMEM;
+			}
+
+			ret = regmap_raw_write(regmap, reg, buf,
 					       le32_to_cpu(region->len));
+
+			kfree(buf);
+
 			if (ret != 0) {
 				adsp_err(dsp,
 					"%s.%d: Failed to write %d bytes at %d in %s: %d\n",
@@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 	const char *region_name;
 	int ret, pos, blocks, type, offset, reg;
 	char *file;
+	void *buf;
 
 	file = kzalloc(PAGE_SIZE, GFP_KERNEL);
 	if (file == NULL)
@@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 		}
 
 		if (reg) {
+			buf = kmemdup(blk->data, le32_to_cpu(blk->len),
+				      GFP_KERNEL);
+			if (!buf) {
+				adsp_err(dsp, "Out of memory\n");
+				return -ENOMEM;
+			}
+
 			ret = regmap_raw_write(regmap, reg, blk->data,
 					       le32_to_cpu(blk->len));
 			if (ret != 0) {
@@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp)
 					"%s.%d: Failed to write to %x in %s\n",
 					file, blocks, reg, region_name);
 			}
+
+			kfree(buf);
 		}
 
 		pos += le32_to_cpu(blk->len) + sizeof(*blk);
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..2370063b5824 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients);
 	ret = device_add(rtd->dev);
 	if (ret < 0) {
+		/* calling put_device() here to free the rtd->dev */
+		put_device(rtd->dev);
 		dev_err(card->dev,
 			"ASoC: failed to register runtime device: %d\n", ret);
 		return ret;
@@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num)
 	/* unregister the rtd device */
 	if (rtd->dev_registered) {
 		device_remove_file(rtd->dev, &dev_attr_codec_reg);
-		device_del(rtd->dev);
+		device_unregister(rtd->dev);
 		rtd->dev_registered = 0;
 	}
 
@@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
 	platform_max = mc->platform_max;
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 1;
+	uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1;
 	uinfo->value.integer.min = 0;
 	uinfo->value.integer.max = platform_max - min;
 
@@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	unsigned int reg = mc->reg;
+	unsigned int rreg = mc->rreg;
 	unsigned int shift = mc->shift;
 	int min = mc->min;
 	int max = mc->max;
 	unsigned int mask = (1 << fls(max)) - 1;
 	unsigned int invert = mc->invert;
 	unsigned int val, val_mask;
+	int ret;
 
 	val = ((ucontrol->value.integer.value[0] + min) & mask);
 	if (invert)
@@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
 	val_mask = mask << shift;
 	val = val << shift;
 
-	return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+	ret = snd_soc_update_bits_locked(codec, reg, val_mask, val);
+	if (ret != 0)
+		return ret;
+
+	if (snd_soc_volsw_is_stereo(mc)) {
+		val = ((ucontrol->value.integer.value[1] + min) & mask);
+		if (invert)
+			val = max - val;
+		val_mask = mask << shift;
+		val = val << shift;
+
+		ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val);
+	}
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
 
@@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	unsigned int reg = mc->reg;
+	unsigned int rreg = mc->rreg;
 	unsigned int shift = mc->shift;
 	int min = mc->min;
 	int max = mc->max;
@@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
 	ucontrol->value.integer.value[0] =
 		ucontrol->value.integer.value[0] - min;
 
+	if (snd_soc_volsw_is_stereo(mc)) {
+		ucontrol->value.integer.value[1] =
+			(snd_soc_read(codec, rreg) >> shift) & mask;
+		if (invert)
+			ucontrol->value.integer.value[1] =
+				max - ucontrol->value.integer.value[1];
+		ucontrol->value.integer.value[1] =
+			ucontrol->value.integer.value[1] - min;
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
@@ -4155,9 +4184,9 @@ 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 +4194,8 @@ 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..cf191e6aebbe 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);
 	}
 
@@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream)
 		if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) &&
+		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) &&
 		    (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP))
 			continue;