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

sound fixes #2 for 3.3-rc3

A collection of small fixes, mostly for regressions.
In addition, a few ASoC wm8994 updates are included, too.

* tag 'sound-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: wm8994: Disable line output discharge prior to ramping VMID
  ASoC: wm8994: Fix typo in VMID ramp setting
  ALSA: oxygen, virtuoso: fix exchanged L/R volumes of aux and CD inputs
  ALSA: usb-audio: add Edirol UM-3G support
  ALSA: hda - add support for Uniwill ECS M31EI notebook
  ALSA: hda - Fix error handling in patch_ca0132.c
  ASoC: wm8994: Enabling VMID should take a runtime PM reference
  ALSA: hda/realtek - Fix a wrong condition
  ALSA: emu8000: Remove duplicate linux/moduleparam.h include from emu8000_patch.c
  ALSA: hda/realtek - Add missing Bass and CLFE as vmaster slaves
  ASoC: wm_hubs: Correct line input to line output 2 paths
  ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode
  ASoC: wm8962: Fix word length configuration
  ASoC: core: Better support for idle_bias_off suspend ignores
  ASoC: wm8994: Remove ASoC level register cache sync
  ASoC: wm_hubs: Fix routing of input PGAs to line output mixer

10 files changed, 76 insertions, 41 deletions

diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index e09f144177f5..c99c6078be33 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -22,7 +22,6 @@
 #include "emu8000_local.h"
 #include <asm/uaccess.h>
 #include <linux/moduleparam.h>
-#include <linux/moduleparam.h>
 
 static int emu8000_reset_addr;
 module_param(emu8000_reset_addr, int, 0444);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 35abe3c62908..21d91d580da8 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol,
 
         err = chipio_read(codec, REG_CODEC_MUTE, &data);
         if (err < 0)
-                return err;
+                goto exit;
 
         /* *valp 0 is mute, 1 is unmute */
         data = (data & 0x7f) | (*valp ? 0 : 0x80);
-        chipio_write(codec, REG_CODEC_MUTE, data);
+        err = chipio_write(codec, REG_CODEC_MUTE, data);
         if (err < 0)
-                return err;
+                goto exit;
 
         spec->curr_hp_switch = *valp;
 
+ exit:
         snd_hda_power_down(codec);
-        return 1;
+        return err < 0 ? err : 1;
 }
 
 static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol,
@@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol,
 
         err = chipio_read(codec, REG_CODEC_MUTE, &data);
         if (err < 0)
-                return err;
+                goto exit;
 
         /* *valp 0 is mute, 1 is unmute */
         data = (data & 0xef) | (*valp ? 0 : 0x10);
-        chipio_write(codec, REG_CODEC_MUTE, data);
+        err = chipio_write(codec, REG_CODEC_MUTE, data);
         if (err < 0)
-                return err;
+                goto exit;
 
         spec->curr_speaker_switch = *valp;
 
+ exit:
         snd_hda_power_down(codec);
-        return 1;
+        return err < 0 ? err : 1;
 }
 
 static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol,
@@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol,
 
         err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data);
         if (err < 0)
-                return err;
+                goto exit;
 
         val = 31 - left_vol;
         data = (data & 0xe0) | val;
-        chipio_write(codec, REG_CODEC_HP_VOL_L, data);
+        err = chipio_write(codec, REG_CODEC_HP_VOL_L, data);
         if (err < 0)
-                return err;
+                goto exit;
 
         val = 31 - right_vol;
         data = (data & 0xe0) | val;
-        chipio_write(codec, REG_CODEC_HP_VOL_R, data);
+        err = chipio_write(codec, REG_CODEC_HP_VOL_R, data);
         if (err < 0)
-                return err;
+                goto exit;
 
         spec->curr_hp_volume[0] = left_vol;
         spec->curr_hp_volume[1] = right_vol;
 
+ exit:
         snd_hda_power_down(codec);
-        return 1;
+        return err < 0 ? err : 1;
 }
 
 static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid)
@@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
                 if (err < 0)
                         return err;
                 err = add_in_volume(codec, spec->dig_in, "IEC958");
+                if (err < 0)
+                        return err;
         }
         return 0;
 }
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a8e82be3d2fc..9350f3c3bdf8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1855,6 +1855,8 @@ static const char * const alc_slave_vols[] = {
         "Speaker Playback Volume",
         "Mono Playback Volume",
         "Line-Out Playback Volume",
+        "CLFE Playback Volume",
+        "Bass Speaker Playback Volume",
         "PCM Playback Volume",
         NULL,
 };
@@ -1870,6 +1872,8 @@ static const char * const alc_slave_sws[] = {
         "Mono Playback Switch",
         "IEC958 Playback Switch",
         "Line-Out Playback Switch",
+        "CLFE Playback Switch",
+        "Bass Speaker Playback Switch",
         "PCM Playback Switch",
         NULL,
 };
@@ -2318,7 +2322,7 @@ static int alc_build_pcms(struct hda_codec *codec)
                  "%s Analog", codec->chip_name);
         info->name = spec->stream_name_analog;
 
-        if (spec->multiout.dac_nids > 0) {
+        if (spec->multiout.num_dacs > 0) {
                 p = spec->stream_analog_playback;
                 if (!p)
                         p = &alc_pcm_analog_playback;
@@ -5623,6 +5627,7 @@ static const struct alc_fixup alc861_fixups[] = {
 
 static const struct snd_pci_quirk alc861_fixup_tbl[] = {
         SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP),
+        SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP),   
         SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP),
         SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
         {}
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 26c7e8bcb229..c0dbb52d45be 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl,
         mutex_lock(&chip->mutex);
         reg = oxygen_read_ac97(chip, codec, index);
         mutex_unlock(&chip->mutex);
-        value->value.integer.value[0] = 31 - (reg & 0x1f);
-        if (stereo)
-                value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f);
+        if (!stereo) {
+                value->value.integer.value[0] = 31 - (reg & 0x1f);
+        } else {
+                value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f);
+                value->value.integer.value[1] = 31 - (reg & 0x1f);
+        }
         return 0;
 }
 
@@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl,
 
         mutex_lock(&chip->mutex);
         oldreg = oxygen_read_ac97(chip, codec, index);
-        newreg = oldreg;
-        newreg = (newreg & ~0x1f) |
-                (31 - (value->value.integer.value[0] & 0x1f));
-        if (stereo)
-                newreg = (newreg & ~0x1f00) |
-                        ((31 - (value->value.integer.value[1] & 0x1f)) << 8);
-        else
-                newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8);
+        if (!stereo) {
+                newreg = oldreg & ~0x1f;
+                newreg |= 31 - (value->value.integer.value[0] & 0x1f);
+        } else {
+                newreg = oldreg & ~0x1f1f;
+                newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8;
+                newreg |= 31 - (value->value.integer.value[1] & 0x1f);
+        }
         change = newreg != oldreg;
         if (change)
                 oxygen_write_ac97(chip, codec, index, newreg);
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 9d38db8f1919..78979b3e0e95 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
                 priv->config[id].mmcc &= 0xC0;
                 priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
                 priv->config[id].spc &= 0xFC;
-                priv->config[id].spc &= MCK_SCLK_64FS;
+                priv->config[id].spc |= MCK_SCLK_MCLK;
         } else {
                 /* CS42L73 Slave */
                 priv->config[id].spc &= 0xFC;
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index bda3da887d7e..29c4b02c4790 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3159,13 +3159,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream,
         case SNDRV_PCM_FORMAT_S16_LE:
                 break;
         case SNDRV_PCM_FORMAT_S20_3LE:
-                aif0 |= 0x40;
+                aif0 |= 0x4;
                 break;
         case SNDRV_PCM_FORMAT_S24_LE:
-                aif0 |= 0x80;
+                aif0 |= 0x8;
                 break;
         case SNDRV_PCM_FORMAT_S32_LE:
-                aif0 |= 0xc0;
+                aif0 |= 0xc;
                 break;
         default:
                 return -EINVAL;
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 93d27b660257..ec69a6c152fe 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec)
 {
         struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
+        pm_runtime_get_sync(codec->dev);
+
         wm8994->vmid_refcount++;
 
         dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n",
@@ -783,7 +785,12 @@ static void vmid_reference(struct snd_soc_codec *codec)
                                     WM8994_VMID_RAMP_MASK,
                                     WM8994_STARTUP_BIAS_ENA |
                                     WM8994_VMID_BUF_ENA |
-                                    (0x11 << WM8994_VMID_RAMP_SHIFT));
+                                    (0x3 << WM8994_VMID_RAMP_SHIFT));
+
+                /* Remove discharge for line out */
+                snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
+                                    WM8994_LINEOUT1_DISCH |
+                                    WM8994_LINEOUT2_DISCH, 0);
 
                 /* Main bias enable, VMID=2x40k */
                 snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
@@ -837,6 +844,8 @@ static void vmid_dereference(struct snd_soc_codec *codec)
                                     WM8994_VMID_BUF_ENA |
                                     WM8994_VMID_RAMP_MASK, 0);
         }
+
+        pm_runtime_put(codec->dev);
 }
 
 static int vmid_event(struct snd_soc_dapm_widget *w,
@@ -2753,11 +2762,6 @@ static int wm8994_resume(struct snd_soc_codec *codec)
                 codec->cache_only = 0;
         }
 
-        /* Restore the registers */
-        ret = snd_soc_cache_sync(codec);
-        if (ret != 0)
-                dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
-
         wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
         for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index ea2672455d07..8a68cea4a3ee 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
 };
 
 static const struct snd_kcontrol_new line2_mix[] = {
-SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0),
-SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0),
+SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0),
+SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0),
 SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
 };
 
@@ -848,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
 };
 
 static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
-        { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
-        { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
+        { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" },
+        { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" },
         { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
 
         { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b5ecf6d23214..92cee24ed2dc 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev)
                 if (!codec->suspended && codec->driver->suspend) {
                         switch (codec->dapm.bias_level) {
                         case SND_SOC_BIAS_STANDBY:
+                                /*
+                                 * If the CODEC is capable of idle
+                                 * bias off then being in STANDBY
+                                 * means it's doing something,
+                                 * otherwise fall through.
+                                 */
+                                if (codec->dapm.idle_bias_off) {
+                                        dev_dbg(codec->dev,
+                                                "idle_bias_off CODEC on over suspend\n");
+                                        break;
+                                }
                         case SND_SOC_BIAS_OFF:
                                 codec->driver->suspend(codec);
                                 codec->suspended = 1;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 8edc5035fc8f..d89ab4c7d44b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1618,6 +1618,14 @@ YAMAHA_DEVICE(0x7010, "UB99"),
         }
 },
 {
+        /* Edirol UM-3G */
+        USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108),
+        .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+                .ifnum = 0,
+                .type = QUIRK_MIDI_STANDARD_INTERFACE
+        }
+},
+{
         /* Boss JS-8 Jam Station  */
         USB_DEVICE(0x0582, 0x0109),
         .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {