Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This batch contains a few USB audio fixes, a couple of HD-audio
  quirks, various small ASoC driver fixes in addition to an ASoC core
  fix that may lead to memory corruption.

  Unfortunately slightly more volume than the previous pull request, but
  all are reasonable regression fixes"

* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Add a fixup for Gateway LT27
  ASoC: tegra: fix Tegra30 I2S capture parameter setup
  ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
  ALSA: hda - Fix missing mute controls for CX5051
  ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection
  ALSA: 6fire: make buffers DMA-able (midi)
  ALSA: 6fire: make buffers DMA-able (pcm)
  ALSA: hda - Add pinfix for LG LW25 laptop
  ASoC: cs42l52: Add new TLV for Beep Volume
  ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume
  ASoC: dapm: Fix empty list check in dapm_new_mux()
  ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control
  ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture

12 files changed, 98 insertions, 23 deletions

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 8e77cbbad871..e3c7ba8d7582 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1,
 }
 
 #define nid_has_mute(codec, nid, dir) \
-        check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE)
+        check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE))
 #define nid_has_volume(codec, nid, dir) \
         check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS)
 
@@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid,
                 if (enable)
                         val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
         }
-        if (caps & AC_AMPCAP_MUTE) {
+        if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
                 if (!enable)
                         val |= HDA_AMP_MUTE;
         }
@@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec,
 {
         unsigned int mask = 0xff;
 
-        if (caps & AC_AMPCAP_MUTE) {
+        if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) {
                 if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL))
                         mask &= ~0x80;
         }
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8bd226149868..f303cd898515 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1031,6 +1031,7 @@ enum {
         ALC880_FIXUP_GPIO2,
         ALC880_FIXUP_MEDION_RIM,
         ALC880_FIXUP_LG,
+        ALC880_FIXUP_LG_LW25,
         ALC880_FIXUP_W810,
         ALC880_FIXUP_EAPD_COEF,
         ALC880_FIXUP_TCL_S700,
@@ -1089,6 +1090,14 @@ static const struct hda_fixup alc880_fixups[] = {
                         { }
                 }
         },
+        [ALC880_FIXUP_LG_LW25] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = (const struct hda_pintbl[]) {
+                        { 0x1a, 0x0181344f }, /* line-in */
+                        { 0x1b, 0x0321403f }, /* headphone */
+                        { }
+                }
+        },
         [ALC880_FIXUP_W810] = {
                 .type = HDA_FIXUP_PINS,
                 .v.pins = (const struct hda_pintbl[]) {
@@ -1341,6 +1350,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG),
         SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG),
         SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG),
+        SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25),
         SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700),
 
         /* Below is the copied entries from alc880_quirks.c.
@@ -4329,6 +4339,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
         SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
         SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
+        SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC),
         SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
         SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
         SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index 987f728718c5..be2ba1b6fe4a 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0);
 
 static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0);
 
+static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0);
+
 static const unsigned int limiter_tlv[] = {
         TLV_DB_RANGE_HEAD(2),
         0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0),
@@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = {
         SOC_ENUM("Beep Pitch", beep_pitch_enum),
         SOC_ENUM("Beep on Time", beep_ontime_enum),
         SOC_ENUM("Beep off Time", beep_offtime_enum),
-        SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv),
+        SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL,
+                        0, 0x07, 0x1f, beep_tlv),
         SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1),
         SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum),
         SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum),
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 6c8a9e7bee25..760e8bfeacaa 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 static int power_vag_event(struct snd_soc_dapm_widget *w,
         struct snd_kcontrol *kcontrol, int event)
 {
+        const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP;
+
         switch (event) {
         case SND_SOC_DAPM_POST_PMU:
                 snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
@@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w,
                 break;
 
         case SND_SOC_DAPM_PRE_PMD:
-                snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
-                        SGTL5000_VAG_POWERUP, 0);
-                msleep(400);
+                /*
+                 * Don't clear VAG_POWERUP, when both DAC and ADC are
+                 * operational to prevent inadvertently starving the
+                 * other one of them.
+                 */
+                if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) &
+                                mask) != mask) {
+                        snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+                                SGTL5000_VAG_POWERUP, 0);
+                        msleep(400);
+                }
                 break;
         default:
                 break;
@@ -388,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
         SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
         SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
                         SGTL5000_CHIP_ANA_ADC_CTRL,
-                        8, 2, 0, capture_6db_attenuate),
+                        8, 1, 0, capture_6db_attenuate),
         SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
 
         SOC_DOUBLE_TLV("Headphone Playback Volume",
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index bd16010441cc..4375c9f2b791 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -679,13 +679,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w)
                 return -EINVAL;
         }
 
-        path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
-                                list_sink);
-        if (!path) {
+        if (list_empty(&w->sources)) {
                 dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name);
                 return -EINVAL;
         }
 
+        path = list_first_entry(&w->sources, struct snd_soc_dapm_path,
+                                list_sink);
+
         ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path);
         if (ret < 0)
                 return ret;
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index d04146cad61f..47565fd04505 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream,
                 reg = TEGRA30_I2S_CIF_RX_CTRL;
         } else {
                 val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
-                reg = TEGRA30_I2S_CIF_RX_CTRL;
+                reg = TEGRA30_I2S_CIF_TX_CTRL;
         }
 
         regmap_write(i2s->regmap, reg, val);
diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c
index 26722423330d..f3dd7266c391 100644
--- a/sound/usb/6fire/midi.c
+++ b/sound/usb/6fire/midi.c
@@ -19,6 +19,10 @@
 #include "chip.h"
 #include "comm.h"
 
+enum {
+        MIDI_BUFSIZE = 64
+};
+
 static void usb6fire_midi_out_handler(struct urb *urb)
 {
         struct midi_runtime *rt = urb->context;
@@ -156,6 +160,12 @@ int usb6fire_midi_init(struct sfire_chip *chip)
         if (!rt)
                 return -ENOMEM;
 
+        rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL);
+        if (!rt->out_buffer) {
+                kfree(rt);
+                return -ENOMEM;
+        }
+
         rt->chip = chip;
         rt->in_received = usb6fire_midi_in_received;
         rt->out_buffer[0] = 0x80; /* 'send midi' command */
@@ -169,6 +179,7 @@ int usb6fire_midi_init(struct sfire_chip *chip)
 
         ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance);
         if (ret < 0) {
+                kfree(rt->out_buffer);
                 kfree(rt);
                 snd_printk(KERN_ERR PREFIX "unable to create midi.\n");
                 return ret;
@@ -197,6 +208,9 @@ void usb6fire_midi_abort(struct sfire_chip *chip)
 
 void usb6fire_midi_destroy(struct sfire_chip *chip)
 {
-        kfree(chip->midi);
+        struct midi_runtime *rt = chip->midi;
+
+        kfree(rt->out_buffer);
+        kfree(rt);
         chip->midi = NULL;
 }
diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h
index c321006e5430..84851b9f5559 100644
--- a/sound/usb/6fire/midi.h
+++ b/sound/usb/6fire/midi.h
@@ -16,10 +16,6 @@
 
 #include "common.h"
 
-enum {
-        MIDI_BUFSIZE = 64
-};
-
 struct midi_runtime {
         struct sfire_chip *chip;
         struct snd_rawmidi *instance;
@@ -32,7 +28,7 @@ struct midi_runtime {
         struct snd_rawmidi_substream *out;
         struct urb out_urb;
         u8 out_serial; /* serial number of out packet */
-        u8 out_buffer[MIDI_BUFSIZE];
+        u8 *out_buffer;
         int buffer_offset;
 
         void (*in_received)(struct midi_runtime *rt, u8 *data, int length);
diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c
index 3d2551cc10f2..b5eb97fdc842 100644
--- a/sound/usb/6fire/pcm.c
+++ b/sound/usb/6fire/pcm.c
@@ -582,6 +582,33 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb,
         urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB;
 }
 
+static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt)
+{
+        int i;
+
+        for (i = 0; i < PCM_N_URBS; i++) {
+                rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB
+                                * PCM_MAX_PACKET_SIZE, GFP_KERNEL);
+                if (!rt->out_urbs[i].buffer)
+                        return -ENOMEM;
+                rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB
+                                * PCM_MAX_PACKET_SIZE, GFP_KERNEL);
+                if (!rt->in_urbs[i].buffer)
+                        return -ENOMEM;
+        }
+        return 0;
+}
+
+static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt)
+{
+        int i;
+
+        for (i = 0; i < PCM_N_URBS; i++) {
+                kfree(rt->out_urbs[i].buffer);
+                kfree(rt->in_urbs[i].buffer);
+        }
+}
+
 int usb6fire_pcm_init(struct sfire_chip *chip)
 {
         int i;
@@ -593,6 +620,13 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
         if (!rt)
                 return -ENOMEM;
 
+        ret = usb6fire_pcm_buffers_init(rt);
+        if (ret) {
+                usb6fire_pcm_buffers_destroy(rt);
+                kfree(rt);
+                return ret;
+        }
+
         rt->chip = chip;
         rt->stream_state = STREAM_DISABLED;
         rt->rate = ARRAY_SIZE(rates);
@@ -614,6 +648,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
 
         ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm);
         if (ret < 0) {
+                usb6fire_pcm_buffers_destroy(rt);
                 kfree(rt);
                 snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n");
                 return ret;
@@ -625,6 +660,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip)
         snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops);
 
         if (ret) {
+                usb6fire_pcm_buffers_destroy(rt);
                 kfree(rt);
                 snd_printk(KERN_ERR PREFIX
                                 "error preallocating pcm buffers.\n");
@@ -669,6 +705,9 @@ void usb6fire_pcm_abort(struct sfire_chip *chip)
 
 void usb6fire_pcm_destroy(struct sfire_chip *chip)
 {
-        kfree(chip->pcm);
+        struct pcm_runtime *rt = chip->pcm;
+
+        usb6fire_pcm_buffers_destroy(rt);
+        kfree(rt);
         chip->pcm = NULL;
 }
diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h
index 9b01133ee3fe..f5779d6182c6 100644
--- a/sound/usb/6fire/pcm.h
+++ b/sound/usb/6fire/pcm.h
@@ -32,7 +32,7 @@ struct pcm_urb {
         struct urb instance;
         struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB];
         /* END DO NOT SEPARATE */
-        u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE];
+        u8 *buffer;
 
         struct pcm_urb *peer;
 };
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index d5438083fd6a..95558ef4a7a0 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -888,6 +888,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval,
         case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */
         case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */
         case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */
+        case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */
         case USB_ID(0x046d, 0x0991):
         /* Most audio usb devices lie about volume resolution.
          * Most Logitech webcams have res = 384.
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 1bc45e71f1fe..0df9ede99dfd 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -319,19 +319,19 @@ static int create_auto_midi_quirk(struct snd_usb_audio *chip,
         if (altsd->bNumEndpoints < 1)
                 return -ENODEV;
         epd = get_endpoint(alts, 0);
-        if (!usb_endpoint_xfer_bulk(epd) ||
+        if (!usb_endpoint_xfer_bulk(epd) &&
             !usb_endpoint_xfer_int(epd))
                 return -ENODEV;
 
         switch (USB_ID_VENDOR(chip->usb_id)) {
         case 0x0499: /* Yamaha */
                 err = create_yamaha_midi_quirk(chip, iface, driver, alts);
-                if (err < 0 && err != -ENODEV)
+                if (err != -ENODEV)
                         return err;
                 break;
         case 0x0582: /* Roland */
                 err = create_roland_midi_quirk(chip, iface, driver, alts);
-                if (err < 0 && err != -ENODEV)
+                if (err != -ENODEV)
                         return err;
                 break;
         }