Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
  ALSA: hda/realtek - Skip invalid digital out pins
  ALSA: hda/sigmatel - Automatically retrieve digital I/O widgets
  ALSA: hda - Remove unused variables
  ALSA: hda/realtek - Don't create alt-stream for capture when unnecessary
  ALSA: hda - Add support for 92HD65 / 92HD66 family of codecs
  ALSA: hda - Disable power-widget control for IDT 92HD83/93 as default
  ALSA: hda - Check NO_PRESENCE pincfg default bit
  ASoC: Ensure we always delay for WM8962 FLL when starting from SYSCLK
  ASoC: Ensure the WM8962 oscillator and PLLs start up disabled
  ASoC: Ensure WM8962 PLL registers are reset
  ALSA: intel8x0 - Fix inclusion of kvm_para.h
  ALSA: hda_hwdep: Fix possible buffer overflow
  ASoC: Fix return value of wm5100_gpio_direction_out()
  ASoC: WM8904: Set `invert' bit for Capture Switch
  ASoC: Leave input audio data bit length settings untouched in wm8711_set_dai_fmt
  ASoC: wm8711: Fix wrong mask for setting input audio data bit length select
  ALSA: intel8x0: Improve performance in virtual environment
  ALSA: hdspm - Enable all firmware ranges for PCI MADI/AES cards
  ALSA: hdsp - Correct HDSP_VERSION_BIT constant, thus partly fixing RPM detection
  ALSA: hdspm - Fix MADI channel format in the status ioctl
  ...

7 files changed, 34 insertions, 18 deletions

diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index ab27dbcd1262..336de8f69a02 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -430,6 +430,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface_reg |= TLV320AIC23_MS_MASTER;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
+		iface_reg &= ~TLV320AIC23_MS_MASTER;
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 7a49390bc30d..87d5ef188e29 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1023,6 +1023,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		aic3x->master = 0;
+		iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER);
 		break;
 	default:
 		return -EINVAL;
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index 5d88c99aaea6..42d9039a49e9 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -2361,13 +2361,17 @@ static int wm5100_gpio_direction_out(struct gpio_chip *chip,
 {
 	struct wm5100_priv *wm5100 = gpio_to_wm5100(chip);
 	struct snd_soc_codec *codec = wm5100->codec;
-	int val;
+	int val, ret;
 
 	val = (1 << WM5100_GP1_FN_SHIFT) | (!!value << WM5100_GP1_LVL_SHIFT);
 
-	return snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset,
-				   WM5100_GP1_FN_MASK | WM5100_GP1_DIR |
-				   WM5100_GP1_LVL, val);
+	ret = snd_soc_update_bits(codec, WM5100_GPIO_CTRL_1 + offset,
+				  WM5100_GP1_FN_MASK | WM5100_GP1_DIR |
+				  WM5100_GP1_LVL, val);
+	if (ret < 0)
+		return ret;
+	else
+		return 0;
 }
 
 static int wm5100_gpio_get(struct gpio_chip *chip, unsigned offset)
diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c
index 8d0347cf0e9a..076bdb9930a1 100644
--- a/sound/soc/codecs/wm8711.c
+++ b/sound/soc/codecs/wm8711.c
@@ -151,7 +151,7 @@ static int wm8711_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_codec *codec = dai->codec;
 	struct wm8711_priv *wm8711 =  snd_soc_codec_get_drvdata(codec);
-	u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc;
+	u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfff3;
 	int i = get_coeff(wm8711->sysclk, params_rate(params));
 	u16 srate = (coeff_div[i].sr << 2) |
 		(coeff_div[i].bosr << 1) | coeff_div[i].usb;
@@ -232,7 +232,7 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 iface = 0;
+	u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0x000c;
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 9fc8f4c0a9a9..285ef87e6704 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -867,7 +867,7 @@ SOC_ENUM("Right Capture Mode", rin_mode),
 SOC_DOUBLE_R("Capture Volume", WM8904_ANALOGUE_LEFT_INPUT_0,
 	     WM8904_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0),
 SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0,
-	     WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 0),
+	     WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 1),
 
 SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0),
 SOC_ENUM("High Pass Filter Mode", hpf_mode),
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index dc5cb3150857..de9ec9b8b7d9 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -621,7 +621,7 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 
 	switch (div_id) {
 	case WM8940_BCLKDIV:
-		reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFEF3;
+		reg = snd_soc_read(codec, WM8940_CLOCK) & 0xFFE3;
 		ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 2));
 		break;
 	case WM8940_MCLKDIV:
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index f60dfa16545e..91d3c6dbeba3 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1961,7 +1961,13 @@ static int wm8962_readable_register(struct snd_soc_codec *codec, unsigned int re
 
 static int wm8962_reset(struct snd_soc_codec *codec)
 {
-	return snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0x6243);
+	int ret;
+
+	ret = snd_soc_write(codec, WM8962_SOFTWARE_RESET, 0x6243);
+	if (ret != 0)
+		return ret;
+
+	return snd_soc_write(codec, WM8962_PLL_SOFTWARE_RESET, 0);
 }
 
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, -2325, 75, 0);
@@ -2360,15 +2366,14 @@ static int sysclk_event(struct snd_soc_dapm_widget *w,
 
 			snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1,
 					    WM8962_FLL_ENA, WM8962_FLL_ENA);
-			if (wm8962->irq) {
-				timeout = msecs_to_jiffies(5);
-				timeout = wait_for_completion_timeout(&wm8962->fll_lock,
-								      timeout);
-
-				if (timeout == 0)
-					dev_err(codec->dev,
-						"Timed out starting FLL\n");
-			}
+
+			timeout = msecs_to_jiffies(5);
+			timeout = wait_for_completion_timeout(&wm8962->fll_lock,
+							      timeout);
+
+			if (wm8962->irq && timeout == 0)
+				dev_err(codec->dev,
+					"Timed out starting FLL\n");
 		}
 		break;
 
@@ -4029,6 +4034,11 @@ static int wm8962_probe(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8962_CLOCKING2,
 			    WM8962_CLKREG_OVD, WM8962_CLKREG_OVD);
 
+	/* Ensure that the oscillator and PLLs are disabled */
+	snd_soc_update_bits(codec, WM8962_PLL2,
+			    WM8962_OSC_ENA | WM8962_PLL2_ENA | WM8962_PLL3_ENA,
+			    0);
+
 	regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 
 	if (pdata) {