Merge git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-fix-2.6

* git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-fix-2.6:
  [ALSA] intel8x0 - Add quirk for Compaq Deskpro EN
  [ALSA] hda-codec - Adapt eeepc p701 mixer for virtual master control
  [ALSA] sound: ice1712: unused structs
  [ALSA] ASoC: Fix WM9712 mixer_event DAPM widget function type
  [ALSA] ASoC: Fix DAPM widget function types in pxa machine drivers
  [ALSA] hda-codec - Fix mixer names of realtek codecs to adapt mater controls
  [ALSA] intel8x0 - Add quirk for Acer Travelmate 2310
  [ALSA] hda_intel - Add model quirk for Albatron KI690-AM2 motherboard
  [ALSA] oxygen: add owner field
  [ALSA] hda-codec - Add Fujitsu Lifebook E8410 to quirk table
  [ALSA] hda-codec - Fix AD1988 capture elements
  [ALSA] sb8: fix SB 1.0 capture DMA programming
  [ALSA] usb-audio: add workaround for broken E-Mu frequency feedback
  [ALSA] ASoC: Fix TLV320AIC3X PLL divider table for 64 kHz rate
  [ALSA] hda-codec - Fix Master volume on HP dv8000

16 files changed, 88 insertions, 27 deletions

diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c
index 6304c3a89ba0..fe03bb820532 100644
--- a/sound/isa/sb/sb8_main.c
+++ b/sound/isa/sb/sb8_main.c
@@ -277,7 +277,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream)
         } else {
                 snd_sbdsp_command(chip, 256 - runtime->rate_den);
         }
-        if (chip->capture_format != SB_DSP_OUTPUT) {
+        if (chip->capture_format != SB_DSP_INPUT) {
                 count--;
                 snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE);
                 snd_sbdsp_command(chip, count & 0xff);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 19f08846d6fc..c8649282c2cf 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1778,9 +1778,9 @@ static hda_nid_t ad1988_capsrc_nids[3] = {
 static struct hda_input_mux ad1988_6stack_capture_source = {
         .num_items = 5,
         .items = {
-                { "Front Mic", 0x0 },
-                { "Line", 0x1 },
-                { "Mic", 0x4 },
+                { "Front Mic", 0x1 },   /* port-B */
+                { "Line", 0x2 },        /* port-C */
+                { "Mic", 0x4 },         /* port-E */
                 { "CD", 0x5 },
                 { "Mix", 0x9 },
         },
@@ -1789,7 +1789,7 @@ static struct hda_input_mux ad1988_6stack_capture_source = {
 static struct hda_input_mux ad1988_laptop_capture_source = {
         .num_items = 3,
         .items = {
-                { "Mic/Line", 0x0 },
+                { "Mic/Line", 0x1 },    /* port-B */
                 { "CD", 0x5 },
                 { "Mix", 0x9 },
         },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index f7cd3a804b11..7206b30cbf94 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1230,6 +1230,11 @@ static struct hda_verb cxt5047_toshiba_init_verbs[] = {
 static struct hda_verb cxt5047_hp_init_verbs[] = {
         /* pin sensing on HP jack */
         {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+        /* 0x13 is actually shared by both HP and speaker;
+         * setting the connection to 0 (=0x19) makes the master volume control
+         * working mysteriouslly...
+         */
+        {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
         /* Record selector: Ext Mic */
         {0x12, AC_VERB_SET_CONNECT_SEL,0x03},
         {0x19, AC_VERB_SET_AMP_GAIN_MUTE,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 777f8c01ca7a..33282f9c01c7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3973,8 +3973,8 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
         ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
         HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
         HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
-        HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
-        HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
         { } /* end */
 };
 
@@ -4005,9 +4005,9 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = {
         HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
         HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
         ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
-        HDA_CODEC_VOLUME_MONO("Mono Speaker Playback Volume", 0x0a, 1, 0x0,
+        HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
                               HDA_OUTPUT),
-        HDA_BIND_MUTE_MONO("Mono Speaker Playback Switch", 0x0a, 1, 2,
+        HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2,
                            HDA_INPUT),
         HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
         HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
@@ -7639,6 +7639,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
         SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763),
         SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
         SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
+        SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
         SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
         {}
@@ -8102,7 +8103,7 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
         HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
         HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
         /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-           HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+           HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
         HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
         HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
         HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
@@ -8124,7 +8125,7 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
         HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
         HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
         /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-           HDA_CODEC_MUTE("PC Beelp Playback Switch", 0x0b, 0x05, HDA_INPUT), */
+           HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
         /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
         HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
         { } /* end */
@@ -9238,6 +9239,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
         SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
         SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
         SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
+        SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
         SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
         SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
         SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
@@ -12993,8 +12995,8 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = {
 static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
         HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT),
 
-        HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-        HDA_CODEC_MUTE("LineOut Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Line-Out Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 
         HDA_CODEC_VOLUME("e-Mic Boost", 0x18, 0, HDA_INPUT),
         HDA_CODEC_VOLUME("e-Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
@@ -13007,8 +13009,8 @@ static struct snd_kcontrol_new alc662_eeepc_p701_mixer[] = {
 };
 
 static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = {
-        HDA_CODEC_VOLUME("LineOut Playback Volume", 0x02, 0x0, HDA_OUTPUT),
-        HDA_CODEC_MUTE("LineOut Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Line-Out Playback Switch", 0x14, 0x0, HDA_OUTPUT),
         HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
         HDA_BIND_MUTE("Surround Playback Switch", 0x03, 2, HDA_INPUT),
         HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x04, 1, 0x0, HDA_OUTPUT),
diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c
index 9ab4a9f383cb..5a158b73dcaa 100644
--- a/sound/pci/ice1712/phase.c
+++ b/sound/pci/ice1712/phase.c
@@ -51,7 +51,7 @@
 struct phase28_spec {
         unsigned short master[2];
         unsigned short vol[8];
-} phase28;
+};
 
 /* WM8770 registers */
 #define WM_DAC_ATTEN            0x00    /* DAC1-8 analog attenuation */
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index ddd5fc8d4fe1..301bf929acd9 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -36,7 +36,7 @@
 struct revo51_spec {
         struct snd_i2c_device *dev;
         struct snd_pt2258 *pt2258;
-} revo51;
+};
 
 static void revo_i2s_mclk_changed(struct snd_ice1712 *ice)
 {
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 061072c7db03..c52abd0bf22e 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1708,6 +1708,12 @@ static struct ac97_pcm ac97_pcm_defs[] __devinitdata = {
 };
 
 static struct ac97_quirk ac97_quirks[] __devinitdata = {
+        {
+                .subvendor = 0x0e11,
+                .subdevice = 0x000e,
+                .name = "Compaq Deskpro EN",    /* AD1885 */
+                .type = AC97_TUNE_HP_ONLY
+        },
         {
                 .subvendor = 0x0e11,
                 .subdevice = 0x008a,
@@ -1740,6 +1746,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
         },
         {
                 .subvendor = 0x1025,
+                .subdevice = 0x0082,
+                .name = "Acer Travelmate 2310",
+                .type = AC97_TUNE_HP_ONLY
+        },
+        {
+                .subvendor = 0x1025,
                 .subdevice = 0x0083,
                 .name = "Acer Aspire 3003LCi",
                 .type = AC97_TUNE_HP_ONLY
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 3ea1f05228a1..666f69a3312e 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -150,6 +150,7 @@ static const struct oxygen_model model_hifier = {
         .shortname = "C-Media CMI8787",
         .longname = "C-Media Oxygen HD Audio",
         .chip = "CMI8788",
+        .owner = THIS_MODULE,
         .init = hifier_init,
         .control_filter = hifier_control_filter,
         .mixer_init = hifier_mixer_init,
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 40e92f5cd69c..d163397b85cc 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -389,6 +389,7 @@ static const struct oxygen_model model_xonar = {
         .shortname = "Asus AV200",
         .longname = "Asus Virtuoso 200",
         .chip = "AV200",
+        .owner = THIS_MODULE,
         .init = xonar_init,
         .control_filter = xonar_control_filter,
         .mixer_init = xonar_mixer_init,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 710e0287ef8c..569ecaca0e8b 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -681,8 +681,8 @@ static const struct aic3x_rate_divs aic3x_divs[] = {
         {22579200, 48000, 48000, 0x0, 8, 7075},
         {33868800, 48000, 48000, 0x0, 5, 8049},
         /* 64k */
-        {22579200, 96000, 96000, 0x1, 8, 7075},
-        {33868800, 96000, 96000, 0x1, 5, 8049},
+        {22579200, 64000, 96000, 0x1, 8, 7075},
+        {33868800, 64000, 96000, 0x1, 5, 8049},
         /* 88.2k */
         {22579200, 88200, 88200, 0x0, 8, 0},
         {33868800, 88200, 88200, 0x0, 5, 3333},
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 590baea3c4c3..524f7450804f 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -176,7 +176,8 @@ static int wm9712_add_controls(struct snd_soc_codec *codec)
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path.
  */
-static int mixer_event (struct snd_soc_dapm_widget *w, int event)
+static int mixer_event(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *k, int event)
 {
         u16 l, r, beep, line, phone, mic, pcm, aux;
 
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 3f34e531bebf..1a70a6ac98ce 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -215,7 +215,8 @@ static int corgi_set_spk(struct snd_kcontrol *kcontrol,
         return 1;
 }
 
-static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
+static int corgi_amp_event(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *k, int event)
 {
         if (SND_SOC_DAPM_EVENT_ON(event))
                 set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON);
@@ -225,7 +226,8 @@ static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event)
         return 0;
 }
 
-static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event)
+static int corgi_mic_event(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *k, int event)
 {
         if (SND_SOC_DAPM_EVENT_ON(event))
                 set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 5ae59bd309a3..4fbf8bba9627 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -196,7 +196,8 @@ static int poodle_set_spk(struct snd_kcontrol *kcontrol,
         return 1;
 }
 
-static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event)
+static int poodle_amp_event(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *k, int event)
 {
         if (SND_SOC_DAPM_EVENT_ON(event))
                 locomo_gpio_write(&poodle_locomo_device.dev,
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index d56709e15435..ecca39033fcc 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -215,7 +215,8 @@ static int spitz_set_spk(struct snd_kcontrol *kcontrol,
         return 1;
 }
 
-static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event)
+static int spitz_mic_bias(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *k, int event)
 {
         if (machine_is_borzoi() || machine_is_spitz()) {
                 if (SND_SOC_DAPM_EVENT_ON(event))
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index e4d40b528ca4..7346d7e5d066 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -135,7 +135,8 @@ static int tosa_set_spk(struct snd_kcontrol *kcontrol,
 }
 
 /* tosa dapm event handlers */
-static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event)
+static int tosa_hp_event(struct snd_soc_dapm_widget *w,
+        struct snd_kcontrol *k, int event)
 {
         if (SND_SOC_DAPM_EVENT_ON(event))
                 set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE);
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 8fa935665702..675672f313be 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -479,6 +479,33 @@ static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs,
         return 0;
 }
 
+/*
+ * process after E-Mu 0202/0404 high speed playback sync complete
+ *
+ * These devices return the number of samples per packet instead of the number
+ * of samples per microframe.
+ */
+static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs,
+                                           struct snd_pcm_runtime *runtime,
+                                           struct urb *urb)
+{
+        unsigned int f;
+        unsigned long flags;
+
+        if (urb->iso_frame_desc[0].status == 0 &&
+            urb->iso_frame_desc[0].actual_length == 4) {
+                f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff;
+                f >>= subs->datainterval;
+                if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) {
+                        spin_lock_irqsave(&subs->lock, flags);
+                        subs->freqm = f;
+                        spin_unlock_irqrestore(&subs->lock, flags);
+                }
+        }
+
+        return 0;
+}
+
 /* determine the number of frames in the next packet */
 static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
 {
@@ -2219,10 +2246,17 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo
         subs->stream = as;
         subs->direction = stream;
         subs->dev = as->chip->dev;
-        if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
+        if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) {
                 subs->ops = audio_urb_ops[stream];
-        else
+        } else {
                 subs->ops = audio_urb_ops_high_speed[stream];
+                switch (as->chip->usb_id) {
+                case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */
+                case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */
+                        subs->ops.retire_sync = retire_playback_sync_urb_hs_emu;
+                        break;
+                }
+        }
         snd_pcm_set_ops(as->pcm, stream,
                         stream == SNDRV_PCM_STREAM_PLAYBACK ?
                         &snd_usb_playback_ops : &snd_usb_capture_ops);