Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Again very calm updates at this time.

  All small fixes for individual drivers, mostly ASoC codecs, in
  addition to soc-compress fix for capture streams which is safe to
  apply as there is no in-tree users yet."

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: cs42l52: fix default value for MASTERA_VOL.
  ASoC: wm8994: check for array index returned
  ASoC: wm8994: Fix reporting of accessory removal on WM8958
  ASoC: wm8994: use the correct pointer to get the control value
  ASoC: wm5110: Correct DSP4R Mixer control name
  ALSA: usb-6fire: Modify firmware version check
  ASoC: cs42l52: fix master playback mute mask.
  ASoC: cs42l52: fix bogus shifts in "Speaker Volume" and "PCM Mixer Volume" controls.
  ASoC: cs42l52: microphone bias is controlled by IFACE_CTL2 register.
  ASoC: davinci: fix sample rotation
  ASoC: wm5110: Add missing speaker initialisation
  ASoC: soc-compress: Send correct stream event for capture start
  ASoC: max98090: request IRQF_ONESHOT interrupt

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),
 

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
 

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",

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;

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)

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);
 	}

sound/soc/soc-compress.c

@@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream,
 			goto err;
 	}
 
+	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;

sound/usb/6fire/firmware.c

@@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = {
 	0x94, 0x01, 0x5c, 0x02  /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */
 };
 
-static const u8 known_fw_versions[][4] = {
-	{ 0x03, 0x01, 0x0b, 0x00 }
+static const u8 known_fw_versions[][2] = {
+	{ 0x03, 0x01 }
 };
 
 struct ihex_record {
@@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version)
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++)
-		if (!memcmp(version, known_fw_versions + i, 4))
+		if (!memcmp(version, known_fw_versions + i, 2))
 			return 0;
 
 	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "