Merge tag 'asoc-v3.10-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v3.10

A series of driver specific updates, none particularly critical, plus
one fix to the compressed API code to handle capture streams properly
which is very safe for mainline as there's no current users.

9 files changed, 30 insertions, 24 deletions

diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 0f6f481cec0..030f53c96ec 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 */
@@ -225,7 +225,7 @@ 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" };
@@ -413,7 +413,7 @@ 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 +441,7 @@ 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, hl_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 60985c05907..4277012c471 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 ce0d36412c9..8d14a76c724 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/wm5110.c b/sound/soc/codecs/wm5110.c
index 731884e0477..ba38f067966 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),
@@ -976,6 +976,8 @@ 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 1eb152cb109..dfd997aaadf 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;
@@ -3833,6 +3838,11 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data)
 			dev_dbg(codec->dev, "Ignoring removed jack\n");
 			return IRQ_HANDLED;
 		}
+	} else if (!(reg & WM8958_MICD_STS)) {
+		snd_soc_jack_report(wm8994->micdet[0].jack, 0,
+				    SND_JACK_MECHANICAL | SND_JACK_HEADSET |
+				    wm8994->btn_mask);
+		goto out;
 	}
 
 	if (wm8994->mic_detecting)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 56ecfc72f2e..81490febac6 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/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c
index 902fab02b85..c6fa03e2114 100644
--- a/sound/soc/fsl/imx-ssi.c
+++ b/sound/soc/fsl/imx-ssi.c
@@ -540,11 +540,6 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	clk_prepare_enable(ssi->clk);
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!res) {
-		ret = -ENODEV;
-		goto failed_get_resource;
-	}
-
 	ssi->base = devm_ioremap_resource(&pdev->dev, res);
 	if (IS_ERR(ssi->base)) {
 		ret = PTR_ERR(ssi->base);
@@ -633,7 +628,6 @@ failed_pdev_fiq_alloc:
 	snd_soc_unregister_component(&pdev->dev);
 failed_register:
 	release_mem_region(res->start, resource_size(res));
-failed_get_resource:
 	clk_disable_unprepare(ssi->clk);
 failed_clk:
 
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index befe68f5928..4c9dad3263c 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -471,11 +471,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	dev_set_drvdata(&pdev->dev, priv);
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!mem) {
-		dev_err(&pdev->dev, "platform_get_resource failed\n");
-		return -ENXIO;
-	}
-
 	priv->io = devm_ioremap_resource(&pdev->dev, mem);
 	if (IS_ERR(priv->io))
 		return PTR_ERR(priv->io);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 3853f7eb3f2..06a8000aa07 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;