Merge 3.10-rc6 into driver-core-next

We want these fixes here too.

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

11 files changed, 80 insertions, 48 deletions

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec09..987f728718c5 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = {
 	{ CS42L52_BEEP_VOL, 0x00 },	/* r1D Beep Volume off Time */
 	{ CS42L52_BEEP_TONE_CTL, 0x00 },	/* r1E Beep Tone Cfg. */
 	{ CS42L52_TONE_CTL, 0x00 },	/* r1F Tone Ctl */
-	{ CS42L52_MASTERA_VOL, 0x88 },	/* r20 Master A Volume */
+	{ CS42L52_MASTERA_VOL, 0x00 },	/* r20 Master A Volume */
 	{ CS42L52_MASTERB_VOL, 0x00 },	/* r21 Master B Volume */
 	{ CS42L52_HPA_VOL, 0x00 },	/* r22 Headphone A Volume */
 	{ CS42L52_HPB_VOL, 0x00 },	/* r23 Headphone B Volume */
@@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0);
 
 static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
 
+static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
+
 static const unsigned int limiter_tlv[] = {
 	TLV_DB_RANGE_HEAD(2),
 	0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -225,7 +227,7 @@ static const char * const mic_bias_level_text[] = {
 };
 
 static const struct soc_enum mic_bias_level_enum =
-	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0,
+	SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0,
 			ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text);
 
 static const char * const cs42l52_mic_text[] = { "Single", "Differential" };
@@ -260,7 +262,7 @@ static const char * const hp_gain_num_text[] = {
 };
 
 static const struct soc_enum hp_gain_enum =
-	SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4,
+	SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5,
 		ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text);
 
 static const char * const beep_pitch_text[] = {
@@ -413,7 +415,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 	SOC_ENUM("Headphone Analog Gain", hp_gain_enum),
 
 	SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL,
-			      CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv),
+			      CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv),
 
 	SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL,
 			      CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv),
@@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
 
 	SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume",
 			    CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL,
-				6, 0x7f, 0x19, hl_tlv),
+				0, 0x7f, 0x19, mix_tlv),
 	SOC_DOUBLE_R("PCM Mixer Switch",
 		     CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1),
 
diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h
index 60985c059071..4277012c4719 100644
--- a/sound/soc/codecs/cs42l52.h
+++ b/sound/soc/codecs/cs42l52.h
@@ -157,7 +157,7 @@
 #define CS42L52_PB_CTL1_INV_PCMA		(1 << 2)
 #define CS42L52_PB_CTL1_MSTB_MUTE		(1 << 1)
 #define CS42L52_PB_CTL1_MSTA_MUTE		(1 << 0)
-#define CS42L52_PB_CTL1_MUTE_MASK		0xFFFD
+#define CS42L52_PB_CTL1_MUTE_MASK		0x03
 #define CS42L52_PB_CTL1_MUTE			3
 #define CS42L52_PB_CTL1_UNMUTE			0
 
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index ce0d36412c97..8d14a76c7249 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	dev_dbg(codec->dev, "irq = %d\n", max98090->irq);
 
 	ret = request_threaded_irq(max98090->irq, NULL,
-		max98090_interrupt, IRQF_TRIGGER_FALLING,
+		max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
 		"max98090_interrupt", codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "request_irq failed: %d\n",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 65d09d60b7c6..1514bf845e4b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 
 			break;
 		}
-
-		if (found)
-			snd_soc_dapm_sync(widget->dapm);
 	}
 
-	ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
-
 	mutex_unlock(&widget->codec->mutex);
+
+	if (found)
+		snd_soc_dapm_sync(widget->dapm);
+
+	ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
 	return ret;
 }
 
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
index e895d3939eef..100fdadda56a 100644
--- a/sound/soc/codecs/wm5102.c
+++ b/sound/soc/codecs/wm5102.c
@@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
 ARIZONA_DSP_WIDGETS(DSP1, "DSP1"),
 
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
-		       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux),
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm5102_aec_loopback_mux),
 
 SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM,
 		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev,
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e04776..88ad7db52dde 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE),
-ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE),
 
 ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
 ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
@@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
 		 NULL, 0),
 
 SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1,
-		       ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux),
+		       ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0,
+		       &wm5110_aec_loopback_mux),
 
 SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
 		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
@@ -976,6 +977,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec)
 	if (ret != 0)
 		return ret;
 
+	arizona_init_spk(codec);
+
 	snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS");
 
 	priv->core.arizona->dapm = &codec->dapm;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1eb152cb1097..29e95f93d482 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int drc = wm8994_get_drc(kcontrol->id.name);
 
+	if (drc < 0)
+		return drc;
 	ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
 
 	return 0;
@@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
+	if (block < 0)
+		return block;
+
 	ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
 
 	return 0;
@@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w,
 {
 	struct snd_soc_codec *codec = w->codec;
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	struct wm8994 *control = codec->control_data;
+	struct wm8994 *control = wm8994->wm8994;
 	int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
 	int i;
 	int dac;
@@ -3831,8 +3836,14 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 				ret);
 		} else if (!(ret & WM1811_JACKDET_LVL)) {
 			dev_dbg(codec->dev, "Ignoring removed jack\n");
-			return IRQ_HANDLED;
+			goto out;
 		}
+	} else if (!(reg & WM8958_MICD_STS)) {
+		snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+				    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+				    wm8994->btn_mask);
+		wm8994->mic_detecting = true;
+		goto out;
 	}
 
 	if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e9..81490febac6d 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 				       int word_length)
 {
 	u32 fmt;
-	u32 rotate = (word_length / 4) & 0x7;
+	u32 tx_rotate = (word_length / 4) & 0x7;
+	u32 rx_rotate = (32 - word_length) / 4;
 	u32 mask = (1ULL << word_length) - 1;
 
 	/*
@@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev,
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
 				TXSSZ(fmt), TXSSZ(0x0F));
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG,
-				TXROT(rotate), TXROT(7));
+				TXROT(tx_rotate), TXROT(7));
 		mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG,
-				RXROT(rotate), RXROT(7));
+				RXROT(rx_rotate), RXROT(7));
 		mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG,
 				mask);
 	}
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f28..06a8000aa07b 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
 			goto err;
 	}
 
-	snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
-				SND_SOC_DAPM_STREAM_START);
+	if (cstream->direction == SND_COMPRESS_PLAYBACK)
+		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK,
+					SND_SOC_DAPM_STREAM_START);
+	else
+		snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE,
+					SND_SOC_DAPM_STREAM_START);
 
 	/* cancel any delayed stream shutdown that is pending */
 	rtd->pop_wait = 0;
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a80c883bb8be..c7051c457b75 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -55,7 +55,8 @@ static int dapm_up_seq[] = {
 	[snd_soc_dapm_clock_supply] = 1,
 	[snd_soc_dapm_micbias] = 2,
 	[snd_soc_dapm_dai_link] = 2,
-	[snd_soc_dapm_dai] = 3,
+	[snd_soc_dapm_dai_in] = 3,
+	[snd_soc_dapm_dai_out] = 3,
 	[snd_soc_dapm_aif_in] = 3,
 	[snd_soc_dapm_aif_out] = 3,
 	[snd_soc_dapm_mic] = 4,
@@ -92,7 +93,8 @@ static int dapm_down_seq[] = {
 	[snd_soc_dapm_value_mux] = 9,
 	[snd_soc_dapm_aif_in] = 10,
 	[snd_soc_dapm_aif_out] = 10,
-	[snd_soc_dapm_dai] = 10,
+	[snd_soc_dapm_dai_in] = 10,
+	[snd_soc_dapm_dai_out] = 10,
 	[snd_soc_dapm_dai_link] = 11,
 	[snd_soc_dapm_clock_supply] = 12,
 	[snd_soc_dapm_regulator_supply] = 12,
@@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_in:
+	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_hp:
 	case snd_soc_dapm_mic:
 	case snd_soc_dapm_spk:
@@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget,
 	switch (widget->id) {
 	case snd_soc_dapm_adc:
 	case snd_soc_dapm_aif_out:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_out:
 		if (widget->active) {
 			widget->outputs = snd_soc_dapm_suspend_check(widget);
 			return widget->outputs;
@@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget,
 	switch (widget->id) {
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_aif_in:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_in:
 		if (widget->active) {
 			widget->inputs = snd_soc_dapm_suspend_check(widget);
 			return widget->inputs;
@@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w)
 	return out != 0 && in != 0;
 }
 
-static int dapm_dai_check_power(struct snd_soc_dapm_widget *w)
-{
-	DAPM_UPDATE_STAT(w, power_checks);
-
-	if (w->active)
-		return w->active;
-
-	return dapm_generic_check_power(w);
-}
-
 /* Check to see if an ADC has power */
 static int dapm_adc_check_power(struct snd_soc_dapm_widget *w)
 {
@@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
-	case snd_soc_dapm_dai:
+	case snd_soc_dapm_dai_in:
+	case snd_soc_dapm_dai_out:
 	case snd_soc_dapm_dai_link:
 		list_add(&path->list, &dapm->card->paths);
 		list_add(&path->list_sink, &wsink->sources);
@@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 		break;
 	case snd_soc_dapm_adc:
 	case snd_soc_dapm_aif_out:
+	case snd_soc_dapm_dai_out:
 		w->power_check = dapm_adc_check_power;
 		break;
 	case snd_soc_dapm_dac:
 	case snd_soc_dapm_aif_in:
+	case snd_soc_dapm_dai_in:
 		w->power_check = dapm_dac_check_power;
 		break;
 	case snd_soc_dapm_pga:
@@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
 	case snd_soc_dapm_clock_supply:
 		w->power_check = dapm_supply_check_power;
 		break;
-	case snd_soc_dapm_dai:
-		w->power_check = dapm_dai_check_power;
-		break;
 	default:
 		w->power_check = dapm_always_on_check_power;
 		break;
@@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 	template.reg = SND_SOC_NOPM;
 
 	if (dai->driver->playback.stream_name) {
-		template.id = snd_soc_dapm_dai;
+		template.id = snd_soc_dapm_dai_in;
 		template.name = dai->driver->playback.stream_name;
 		template.sname = dai->driver->playback.stream_name;
 
@@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 	}
 
 	if (dai->driver->capture.stream_name) {
-		template.id = snd_soc_dapm_dai;
+		template.id = snd_soc_dapm_dai_out;
 		template.name = dai->driver->capture.stream_name;
 		template.sname = dai->driver->capture.stream_name;
 
@@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 
 	/* For each DAI widget... */
 	list_for_each_entry(dai_w, &card->widgets, list) {
-		if (dai_w->id != snd_soc_dapm_dai)
+		switch (dai_w->id) {
+		case snd_soc_dapm_dai_in:
+		case snd_soc_dapm_dai_out:
+			break;
+		default:
 			continue;
+		}
 
 		dai = dai_w->priv;
 
@@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 			if (w->dapm != dai_w->dapm)
 				continue;
 
-			if (w->id == snd_soc_dapm_dai)
+			switch (w->id) {
+			case snd_soc_dapm_dai_in:
+			case snd_soc_dapm_dai_out:
 				continue;
+			default:
+				break;
+			}
 
 			if (!w->sname)
 				continue;
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 73bb8eefa491..ccb6be4d658d 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream,
 	/* Create any new FE <--> BE connections */
 	for (i = 0; i < list->num_widgets; i++) {
 
-		if (list->widgets[i]->id != snd_soc_dapm_dai)
+		switch (list->widgets[i]->id) {
+		case snd_soc_dapm_dai_in:
+		case snd_soc_dapm_dai_out:
+			break;
+		default:
 			continue;
+		}
 
 		/* is there a valid BE rtd for this widget */
 		be = dpcm_get_be(card, list->widgets[i], stream);
@@ -2011,9 +2016,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		if (cpu_dai->driver->capture.channels_min)
 			capture = 1;
 	} else {
-		if (codec_dai->driver->playback.channels_min)
+		if (codec_dai->driver->playback.channels_min &&
+		    cpu_dai->driver->playback.channels_min)
 			playback = 1;
-		if (codec_dai->driver->capture.channels_min)
+		if (codec_dai->driver->capture.channels_min &&
+		    cpu_dai->driver->capture.channels_min)
 			capture = 1;
 	}