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

sound fixes for 3.3-rc5

Just a collection of boring small fixes for ASoC, HD-audio Realtek
and USB-audio drivers.

* tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: snd-usb-caiaq: Fix the return of XRUN
  ASoC: ak4642: fixup HeadPhone L/R dapm settings
  ALSA: hda/realtek - Fix surround output regression on Acer Aspire 5935
  ALSA: hda/realtek - Fix overflow of vol/sw check bitmap
  ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()
  ASoC: wm8962: Fix sidetone enumeration texts

7 files changed, 44 insertions, 24 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 1358987c49d..3647baa9bfe 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
 	ALC_AUTOMUTE_MIXER,	/* mute/unmute mixer widget AMP */
 };
 
+#define MAX_VOL_NIDS	0x40
+
 struct alc_spec {
 	/* codec parameterization */
 	const struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
 	const hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
 	hda_nid_t mixer_nid;		/* analog-mixer NID */
-	DECLARE_BITMAP(vol_ctls, 0x20 << 1);
-	DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+	DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+	DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
 
 	/* capture setup for dynamic dual-adc switch */
 	hda_nid_t cur_adc;
@@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
 static inline unsigned int get_ctl_pos(unsigned int data)
 {
 	hda_nid_t nid = get_amp_nid_(data);
-	unsigned int dir = get_amp_direction_(data);
+	unsigned int dir;
+	if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+		return 0;
+	dir = get_amp_direction_(data);
 	return (nid << 1) | dir;
 }
 
@@ -4436,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec,
 				   const struct alc_fixup *fix, int action)
 {
 	if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+		/* fake the connections during parsing the tree */
 		hda_nid_t conn1[2] = { 0x0c, 0x0d };
 		hda_nid_t conn2[2] = { 0x0e, 0x0f };
 		snd_hda_override_conn_list(codec, 0x14, 2, conn1);
 		snd_hda_override_conn_list(codec, 0x15, 2, conn1);
 		snd_hda_override_conn_list(codec, 0x18, 2, conn2);
 		snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+	} else if (action == ALC_FIXUP_ACT_PROBE) {
+		/* restore the connections */
+		hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+		snd_hda_override_conn_list(codec, 0x14, 5, conn);
+		snd_hda_override_conn_list(codec, 0x15, 5, conn);
+		snd_hda_override_conn_list(codec, 0x18, 5, conn);
+		snd_hda_override_conn_list(codec, 0x1a, 5, conn);
 	}
 }
 
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e..278c0a0575f 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 
 	SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
 			 0, 0xFF, 1, out_tlv),
-
-	SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
 };
 
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
-	SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+	SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
 
 static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
 	SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("HPOUTR"),
 	SND_SOC_DAPM_OUTPUT("LINEOUT"),
 
-	SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
-			   &ak4642_hpout_mixer_controls[0],
-			   ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+	SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+	SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+			    &ak4642_headphone_control),
 
-	SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
-			   &ak4642_hpout_mixer_controls[0],
-			   ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+	SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
 
 	SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
 			   &ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
 static const struct snd_soc_dapm_route ak4642_intercon[] = {
 
 	/* Outputs */
-	{"HPOUTL", NULL, "HPOUTL Mixer"},
-	{"HPOUTR", NULL, "HPOUTR Mixer"},
+	{"HPOUTL", NULL, "HPL Out"},
+	{"HPOUTR", NULL, "HPR Out"},
 	{"LINEOUT", NULL, "LINEOUT Mixer"},
 
-	{"HPOUTL Mixer", "DACH", "DAC"},
-	{"HPOUTR Mixer", "DACH", "DAC"},
+	{"HPL Out", NULL, "Headphone Enable"},
+	{"HPR Out", NULL, "Headphone Enable"},
+
+	{"Headphone Enable", "Switch", "DACH"},
+
+	{"DACH", NULL, "DAC"},
+
 	{"LINEOUT Mixer", "DACL", "DAC"},
 };
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 29c4b02c479..0ac228b7dc0 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2564,7 +2564,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
-static const char *st_text[] = { "None", "Right", "Left" };
+static const char *st_text[] = { "None", "Left", "Right" };
 
 static const struct soc_enum str_enum =
 	SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text);
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index 2cf87f5afed..fde9a7a29cb 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -311,8 +311,10 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
 
 	spin_lock(&dev->spinlock);
 
-	if (dev->input_panic || dev->output_panic)
+	if (dev->input_panic || dev->output_panic) {
 		ptr = SNDRV_PCM_POS_XRUN;
+		goto unlock;
+	}
 
 	if (sub->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		ptr = bytes_to_frames(sub->runtime,
@@ -321,6 +323,7 @@ snd_usb_caiaq_pcm_pointer(struct snd_pcm_substream *sub)
 		ptr = bytes_to_frames(sub->runtime,
 					dev->audio_in_buf_pos[index]);
 
+unlock:
 	spin_unlock(&dev->spinlock);
 	return ptr;
 }
diff --git a/sound/usb/card.h b/sound/usb/card.h
index a39edcc32a9..da5fa1ac4ed 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -1,6 +1,7 @@
 #ifndef __USBAUDIO_CARD_H
 #define __USBAUDIO_CARD_H
 
+#define MAX_NR_RATES	1024
 #define MAX_PACKS	20
 #define MAX_PACKS_HS	(MAX_PACKS * 8)	/* in high speed mode */
 #define MAX_URBS	8
diff --git a/sound/usb/format.c b/sound/usb/format.c
index e09aba19375..ddfef57c4c9 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -209,8 +209,6 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 	return 0;
 }
 
-#define MAX_UAC2_NR_RATES 1024
-
 /*
  * Helper function to walk the array of sample rate triplets reported by
  * the device. The problem is that we need to parse whole array first to
@@ -255,7 +253,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
 			fp->rates |= snd_pcm_rate_to_rate_bit(rate);
 
 			nr_rates++;
-			if (nr_rates >= MAX_UAC2_NR_RATES) {
+			if (nr_rates >= MAX_NR_RATES) {
 				snd_printk(KERN_ERR "invalid uac2 rates\n");
 				break;
 			}
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index a3ddac0deff..27817266867 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -132,10 +132,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	unsigned *rate_table = NULL;
 
 	fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
-	if (! fp) {
+	if (!fp) {
 		snd_printk(KERN_ERR "cannot memdup\n");
 		return -ENOMEM;
 	}
+	if (fp->nr_rates > MAX_NR_RATES) {
+		kfree(fp);
+		return -EINVAL;
+	}
 	if (fp->nr_rates > 0) {
 		rate_table = kmemdup(fp->rate_table,
 				     sizeof(int) * fp->nr_rates, GFP_KERNEL);