[GIT PULL] ALSA fixes

[Takashi Iwai <tiwai@suse.de>]

Hi Linus,

please pull ALSA updates for 2.6.28 from:

  git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus

This contains only the following small bug fixes.

Thanks!

Takashi

==

Harvey Harrison (1):
      ALSA: hda - correct bracketing in spdif test in patch_sigmatel.c

Jarkko Nikula (2):
      ALSA: ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver
      ALSA: ASoC: tlv320aic3x: Fix DSP DAI format and signal polarities matching

Johannes Berg (1):
      ALSA: aoa i2sbus: don't overwrite module parameter

Mark Brown (1):
      ALSA: Ensure PXA runtime data is initialised

Takashi Iwai (1):
      ALSA: hda - Fix conflicting volume controls on ALC260

 sound/aoa/soundbus/i2sbus/i2sbus-core.c |    6 +++---
 sound/arm/pxa2xx-pcm-lib.c              |    2 +-
 sound/pci/hda/patch_realtek.c           |   22 ++++++++++++++--------
 sound/pci/hda/patch_sigmatel.c          |    2 +-
 sound/soc/codecs/tlv320aic3x.c          |   16 ++++++++++------
 sound/soc/omap/omap-mcbsp.c             |    7 ++-----
 6 files changed, 31 insertions(+), 24 deletions(-)

diff --git a/sound/aoa/soundbus/i2sbus/i2sbus-core.c b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
index e6beb92..b4590df 100644
--- a/sound/aoa/soundbus/i2sbus/i2sbus-core.c
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -159,7 +159,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	struct i2sbus_dev *dev;
 	struct device_node *child = NULL, *sound = NULL;
 	struct resource *r;
-	int i, layout = 0, rlen;
+	int i, layout = 0, rlen, ok = force;
 	static const char *rnames[] = { "i2sbus: %s (control)",
 					"i2sbus: %s (tx)",
 					"i2sbus: %s (rx)" };
@@ -192,7 +192,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 			layout = *layout_id;
 			snprintf(dev->sound.modalias, 32,
 				 "sound-layout-%d", layout);
-			force = 1;
+			ok = 1;
 		}
 	}
 	/* for the time being, until we can handle non-layout-id
@@ -201,7 +201,7 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 	 * When there are two i2s busses and only one has a layout-id,
 	 * then this depends on the order, but that isn't important
 	 * either as the second one in that case is just a modem. */
-	if (!force) {
+	if (!ok) {
 		kfree(dev);
 		return -ENODEV;
 	}
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 1c93eb7..75a0d74 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
 		goto out;
 
 	ret = -ENOMEM;
-	rtd = kmalloc(sizeof(*rtd), GFP_KERNEL);
+	rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
 	if (!rtd)
 		goto out;
 	rtd->dma_desc_array =
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e72707c..ef4955c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4996,7 +4996,7 @@ static struct hda_verb alc260_test_init_verbs[] = {
  */
 
 static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
-					const char *pfx)
+					const char *pfx, int *vol_bits)
 {
 	hda_nid_t nid_vol;
 	unsigned long vol_val, sw_val;
@@ -5018,10 +5018,14 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid,
 	} else
 		return 0; /* N/A */
 
-	snprintf(name, sizeof(name), "%s Playback Volume", pfx);
-	err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
-	if (err < 0)
-		return err;
+	if (!(*vol_bits & (1 << nid_vol))) {
+		/* first control for the volume widget */
+		snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+		err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val);
+		if (err < 0)
+			return err;
+		*vol_bits |= (1 << nid_vol);
+	}
 	snprintf(name, sizeof(name), "%s Playback Switch", pfx);
 	err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val);
 	if (err < 0)
@@ -5035,6 +5039,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 {
 	hda_nid_t nid;
 	int err;
+	int vols = 0;
 
 	spec->multiout.num_dacs = 1;
 	spec->multiout.dac_nids = spec->private_dac_nids;
@@ -5042,21 +5047,22 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	nid = cfg->line_out_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Front");
+		err = alc260_add_playback_controls(spec, nid, "Front", &vols);
 		if (err < 0)
 			return err;
 	}
 
 	nid = cfg->speaker_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Speaker");
+		err = alc260_add_playback_controls(spec, nid, "Speaker", &vols);
 		if (err < 0)
 			return err;
 	}
 
 	nid = cfg->hp_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Headphone");
+		err = alc260_add_playback_controls(spec, nid, "Headphone",
+						   &vols);
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a2ac720..788fdc6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1282,7 +1282,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 		spec->multiout.share_spdif = 1;
 	}
-	if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) {
+	if (spec->dig_in_nid && !(spec->gpio_dir & 0x01)) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
 		if (err < 0)
 			return err;
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 05336ed..cff276e 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -863,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	/* interface format */
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_I2S:
+	/*
+	 * match both interface format and signal polarities since they
+	 * are fixed
+	 */
+	switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+		       SND_SOC_DAIFMT_INV_MASK)) {
+	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
+	case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x02 << 6);
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
+	case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x03 << 6);
 		break;
 	default:
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 853b33a..8485a8a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		regs->srgr2	|= FPER(wlen * 2 - 1);
-		regs->srgr1	|= FWID(0);
+		regs->srgr1	|= FWID(wlen * 2 - 2);
 		break;
 	}
 
@@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 {
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
-	unsigned int temp_fmt = fmt;
 
 	if (mcbsp_data->configured)
 		return 0;
@@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
 		regs->xcr2      |= XDATDLY(0);
-		/* Invert bit clock and FS polarity configuration for DSP_A */
-		temp_fmt ^= SND_SOC_DAIFMT_IB_IF;
 		break;
 	default:
 		/* Unsupported data format */
@@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	}
 
 	/* Set bit clock (CLKX/CLKR) and FS polarities */
-	switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
 		/*
 		 * Normal BCLK + FS.