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 1358987c49d8..3647baa9bfed 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 5ef70b5d27e4..278c0a0575f5 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 29c4b02c4790..0ac228b7dc04 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 2cf87f5afed4..fde9a7a29cb6 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 a39edcc32a93..da5fa1ac4eda 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 e09aba19375c..ddfef57c4c9f 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 a3ddac0deffd..27817266867a 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);