Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix a copmile warning
  ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
  ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek
  ASoC: Manage WM8731 ACTIVE bit as a supply widget
  ASoC: Don't set invalid name string to snd_card->driver field
  ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
  ASoC: Tegra: I2S: Ensure clock is enabled when writing regs
  ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
  ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state
  ASoC: tlv320aic3x: Don't sync first two registers from register cache
  audio: tlv320aic26: fix PLL register configuration

9 files changed, 68 insertions, 45 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d21191dcfe88..b48fb43b5448 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol,
 
 static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol,
-				 getput_call_t func)
+				 getput_call_t func, bool check_adc_switch)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
-	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-	int err;
+	int i, err = 0;
 
 	mutex_lock(&codec->control_mutex);
-	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx],
-						      3, 0, HDA_INPUT);
-	err = func(kcontrol, ucontrol);
+	if (check_adc_switch && spec->dual_adc_switch) {
+		for (i = 0; i < spec->num_adc_nids; i++) {
+			kcontrol->private_value =
+				HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+						    3, 0, HDA_INPUT);
+			err = func(kcontrol, ucontrol);
+			if (err < 0)
+				goto error;
+		}
+	} else {
+		i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+		kcontrol->private_value =
+			HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
+					    3, 0, HDA_INPUT);
+		err = func(kcontrol, ucontrol);
+	}
+ error:
 	mutex_unlock(&codec->control_mutex);
 	return err;
 }
@@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_volume_get);
+				     snd_hda_mixer_amp_volume_get, false);
 }
 
 static int alc_cap_vol_put(struct snd_kcontrol *kcontrol,
 			   struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_volume_put);
+				     snd_hda_mixer_amp_volume_put, true);
 }
 
 /* capture mixer elements */
@@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_switch_get);
+				     snd_hda_mixer_amp_switch_get, false);
 }
 
 static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
 			  struct snd_ctl_elem_value *ucontrol)
 {
 	return alc_cap_getput_caller(kcontrol, ucontrol,
-				     snd_hda_mixer_amp_switch_put);
+				     snd_hda_mixer_amp_switch_put, true);
 }
 
 #define _DEFINE_CAPMIX(num) \
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index b5101efd1c87..f1fd95bb6416 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 	pr_debug("%s enter\n", __func__);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		diff = sport_curr_offset_tx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	} else {
 		diff = sport_curr_offset_rx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	}
+
+	/*
+	 * TX at least can report one frame beyond the end of the
+	 * buffer if we hit the wraparound case - clamp to within the
+	 * buffer as the ALSA APIs require.
+	 */
+	if (diff == snd_pcm_lib_buffer_bytes(substream))
+		diff = 0;
+
+	frames = bytes_to_frames(substream->runtime, diff);
+
 	return frames;
 }
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4be0570e3f1f..65f46047b1cb 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	default:
 		return -EINVAL;
 	}
-	snd_soc_update_bits(codec, PW_MGMT2, MS, data);
+	snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
 	snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
 
 	/* format type */
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index e2a7608d3944..7859bdcc93db 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 		dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL;
 	}
 
-	/* Configure PLL */
+	/**
+	 * Configure PLL
+	 * fsref = (mclk * PLLM) / 2048
+	 * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal)
+	 */
 	pval = 1;
-	jval = (fsref == 44100) ? 7 : 8;
-	dval = (fsref == 44100) ? 5264 : 1920;
+	/* compute J portion of multiplier */
+	jval = fsref / (aic26->mclk / 2048);
+	/* compute fractional DDDD component of multiplier */
+	dval = fsref - (jval * (aic26->mclk / 2048));
+	dval = (10000 * dval) / (aic26->mclk / 2048);
+	dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval);
 	qval = 0;
 	reg = 0x8000 | qval << 11 | pval << 8 | jval << 2;
 	aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index c3d96fc8c267..789453d44ec5 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -1114,12 +1114,19 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power)
 
 		/* Sync reg_cache with the hardware */
 		codec->cache_only = 0;
-		for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++)
+		for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
 			snd_soc_write(codec, i, cache[i]);
 		if (aic3x->model == AIC3X_MODEL_3007)
 			aic3x_init_3007(codec);
 		codec->cache_sync = 0;
 	} else {
+		/*
+		 * Do soft reset to this codec instance in order to clear
+		 * possible VDD leakage currents in case the supply regulators
+		 * remain on
+		 */
+		snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
+		codec->cache_sync = 1;
 		aic3x->power = 0;
 		/* HW writes are needless when bias is off */
 		codec->cache_only = 1;
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 2dc964b55e4f..76b4361e9b80 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
 SOC_DAPM_ENUM("Input Select", wm8731_insel_enum);
 
 static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
 	&wm8731_output_mixer_controls[0],
@@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
 static const struct snd_soc_dapm_route wm8731_intercon[] = {
 	{"DAC", NULL, "OSC", wm8731_check_osc},
 	{"ADC", NULL, "OSC", wm8731_check_osc},
+	{"DAC", NULL, "ACTIVE"},
+	{"ADC", NULL, "ACTIVE"},
 
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
@@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
-			      struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	/* set active */
-	snd_soc_write(codec, WM8731_ACTIVE, 0x0001);
-
-	return 0;
-}
-
-static void wm8731_shutdown(struct snd_pcm_substream *substream,
-			    struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-
-	/* deactivate */
-	if (!codec->active) {
-		udelay(50);
-		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
-	}
-}
-
 static int wm8731_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
-		snd_soc_write(codec, WM8731_ACTIVE, 0x0);
 		snd_soc_write(codec, WM8731_PWR, 0xffff);
 		regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies),
 				       wm8731->supplies);
@@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 	SNDRV_PCM_FMTBIT_S24_LE)
 
 static struct snd_soc_dai_ops wm8731_dai_ops = {
-	.prepare	= wm8731_pcm_prepare,
 	.hw_params	= wm8731_hw_params,
-	.shutdown	= wm8731_shutdown,
 	.digital_mute	= wm8731_mute,
 	.set_sysclk	= wm8731_set_dai_sysclk,
 	.set_fmt	= wm8731_set_dai_fmt,
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 970a95c5360b..c2fc0356c2a4 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
 				    WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
 				    reg);
+
+		msleep(5);
 	}
 
 	wm8994->fll[id].in = freq_in;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d75043ed7fc0..b194be09e74d 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 		 "%s", card->name);
 	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
 		 "%s", card->long_name ? card->long_name : card->name);
-	snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
-		 "%s", card->driver_name ? card->driver_name : card->name);
+	if (card->driver_name)
+		strlcpy(card->snd_card->driver, card->driver_name,
+			sizeof(card->snd_card->driver));
 
 	if (card->late_probe) {
 		ret = card->late_probe(card);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548c..95f03c10b4f7 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (i2sclock % (2 * srate))
 		reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
 
+	if (!i2s->clk_refs)
+		clk_enable(i2s->clk_i2s);
+
 	tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
 
 	tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
 		TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
 		TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
+	if (!i2s->clk_refs)
+		clk_disable(i2s->clk_i2s);
+
 	return 0;
 }