9 files changed, 107 insertions, 52 deletions

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 2da89810ca10..1292dcee072d 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1971,6 +1971,9 @@ static int snd_ac97_ad1888_lohpsel_get(struct snd_kcontrol *kcontrol, struct snd
 
         val = ac97->regs[AC97_AD_MISC];
         ucontrol->value.integer.value[0] = !(val & AC97_AD198X_LOSEL);
+        if (ac97->spec.ad18xx.lo_as_master)
+                ucontrol->value.integer.value[0] =
+                        !ucontrol->value.integer.value[0];
         return 0;
 }
 
@@ -1979,8 +1982,10 @@ static int snd_ac97_ad1888_lohpsel_put(struct snd_kcontrol *kcontrol, struct snd
         struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol);
         unsigned short val;
 
-        val = !ucontrol->value.integer.value[0]
-                ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
+        val = !ucontrol->value.integer.value[0];
+        if (ac97->spec.ad18xx.lo_as_master)
+                val = !val;
+        val = val ? (AC97_AD198X_LOSEL | AC97_AD198X_HPSEL) : 0;
         return snd_ac97_update_bits(ac97, AC97_AD_MISC,
                                     AC97_AD198X_LOSEL | AC97_AD198X_HPSEL, val);
 }
@@ -2031,7 +2036,7 @@ static void ad1888_update_jacks(struct snd_ac97 *ac97)
 {
         unsigned short val = 0;
         /* clear LODIS if shared jack is to be used for Surround out */
-        if (is_shared_linein(ac97))
+        if (!ac97->spec.ad18xx.lo_as_master && is_shared_linein(ac97))
                 val |= (1 << 12);
         /* clear CLDIS if shared jack is to be used for C/LFE out */
         if (is_shared_micin(ac97))
@@ -2067,9 +2072,13 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
 
 static int patch_ad1888_specific(struct snd_ac97 *ac97)
 {
-        /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
-        snd_ac97_rename_vol_ctl(ac97, "Master Playback", "Master Surround Playback");
-        snd_ac97_rename_vol_ctl(ac97, "Headphone Playback", "Master Playback");
+        if (!ac97->spec.ad18xx.lo_as_master) {
+                /* rename 0x04 as "Master" and 0x02 as "Master Surround" */
+                snd_ac97_rename_vol_ctl(ac97, "Master Playback",
+                                        "Master Surround Playback");
+                snd_ac97_rename_vol_ctl(ac97, "Headphone Playback",
+                                        "Master Playback");
+        }
         return patch_build_controls(ac97, snd_ac97_ad1888_controls, ARRAY_SIZE(snd_ac97_ad1888_controls));
 }
 
@@ -2088,16 +2097,27 @@ static int patch_ad1888(struct snd_ac97 * ac97)
         
         patch_ad1881(ac97);
         ac97->build_ops = &patch_ad1888_build_ops;
-        /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
-        /* it seems that most vendors connect line-out connector to headphone out of AC'97 */
+
+        /*
+         * LO can be used as a real line-out on some devices,
+         * and we need to revert the front/surround mixer switches
+         */
+        if (ac97->subsystem_vendor == 0x1043 &&
+            ac97->subsystem_device == 0x1193) /* ASUS A9T laptop */
+                ac97->spec.ad18xx.lo_as_master = 1;
+
+        misc = snd_ac97_read(ac97, AC97_AD_MISC);
         /* AD-compatible mode */
         /* Stereo mutes enabled */
-        misc = snd_ac97_read(ac97, AC97_AD_MISC);
-        snd_ac97_write_cache(ac97, AC97_AD_MISC, misc |
-                             AC97_AD198X_LOSEL |
-                             AC97_AD198X_HPSEL |
-                             AC97_AD198X_MSPLT |
-                             AC97_AD198X_AC97NC);
+        misc |= AC97_AD198X_MSPLT | AC97_AD198X_AC97NC;
+        if (!ac97->spec.ad18xx.lo_as_master)
+                /* Switch FRONT/SURROUND LINE-OUT/HP-OUT default connection */
+                /* it seems that most vendors connect line-out connector to
+                 * headphone out of AC'97
+                 */
+                misc |= AC97_AD198X_LOSEL | AC97_AD198X_HPSEL;
+
+        snd_ac97_write_cache(ac97, AC97_AD_MISC, misc);
         ac97->flags |= AC97_STEREO_MUTES;
         return 0;
 }
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 56f87cd33c19..3f00ddf450f8 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -316,6 +316,8 @@ static int __devinit snd_aw2_create(struct snd_card *card,
                 return -ENOMEM;
         }
 
+        /* (2) initialization of the chip hardware */
+        snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
 
         if (request_irq(pci->irq, snd_aw2_saa7146_interrupt,
                         IRQF_SHARED, "Audiowerk2", chip)) {
@@ -329,8 +331,6 @@ static int __devinit snd_aw2_create(struct snd_card *card,
         }
         chip->irq = pci->irq;
 
-        /* (2) initialization of the chip hardware */
-        snd_aw2_saa7146_setup(&chip->saa7146, chip->iobase_virt);
         err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
         if (err < 0) {
                 free_irq(chip->irq, (void *)chip);
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index abde5b901884..548c9cc81af5 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -1818,13 +1818,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
         }
         emu->port = pci_resource_start(pci, 0);
 
-        if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
-                        "EMU10K1", emu)) {
-                err = -EBUSY;
-                goto error;
-        }
-        emu->irq = pci->irq;
-
         emu->max_cache_pages = max_cache_bytes >> PAGE_SHIFT;
         if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci),
                                 32 * 1024, &emu->ptb_pages) < 0) {
@@ -1887,6 +1880,14 @@ int __devinit snd_emu10k1_create(struct snd_card *card,
         emu->fx8010.etram_pages.area = NULL;
         emu->fx8010.etram_pages.bytes = 0;
 
+        /* irq handler must be registered after I/O ports are activated */
+        if (request_irq(pci->irq, snd_emu10k1_interrupt, IRQF_SHARED,
+                        "EMU10K1", emu)) {
+                err = -EBUSY;
+                goto error;
+        }
+        emu->irq = pci->irq;
+
         /*
          *  Init to 0x02109204 :
          *  Clock accuracy    = 0     (1000ppm)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e0a605adde42..a99e86d74278 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2858,6 +2858,7 @@ static const char *ad1988_models[AD1988_MODEL_LAST] = {
 static struct snd_pci_quirk ad1988_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
         SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
+        SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
         {}
 };
 
@@ -3643,33 +3644,17 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
         { } /* end */
 };
 
-static struct hda_input_mux ad1884a_mobile_capture_source = {
-        .num_items = 2,
-        .items = {
-                { "Mic", 0x1 }, /* port-C */
-                { "Mix", 0x3 },
-        },
-};
-
 static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
         HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
         HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
         HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
         HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-        HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
         HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
         HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
         HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
         HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-        {
-                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-                .name = "Capture Source",
-                .info = ad198x_mux_enum_info,
-                .get = ad198x_mux_enum_get,
-                .put = ad198x_mux_enum_put,
-        },
         { } /* end */
 };
 
@@ -3686,14 +3671,31 @@ static void ad1884a_hp_automute(struct hda_codec *codec)
                             present ? 0x00 : 0x02);
 }
 
+/* switch to external mic if plugged */
+static void ad1884a_hp_automic(struct hda_codec *codec)
+{
+        unsigned int present;
+
+        present = snd_hda_codec_read(codec, 0x14, 0,
+                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+        snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
+                            present ? 0 : 1);
+}
+
 #define AD1884A_HP_EVENT                0x37
+#define AD1884A_MIC_EVENT               0x36
 
 /* unsolicited event for HP jack sensing */
 static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-        if ((res >> 26) != AD1884A_HP_EVENT)
-                return;
-        ad1884a_hp_automute(codec);
+        switch (res >> 26) {
+        case AD1884A_HP_EVENT:
+                ad1884a_hp_automute(codec);
+                break;
+        case AD1884A_MIC_EVENT:
+                ad1884a_hp_automic(codec);
+                break;
+        }
 }
 
 /* initialize jack-sensing, too */
@@ -3701,6 +3703,7 @@ static int ad1884a_hp_init(struct hda_codec *codec)
 {
         ad198x_init(codec);
         ad1884a_hp_automute(codec);
+        ad1884a_hp_automic(codec);
         return 0;
 }
 
@@ -3714,10 +3717,15 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
         /* Port-F pin */
         {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
         {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Port-C pin - internal mic-in */
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+        {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
         /* analog mix */
         {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
         /* unsolicited event for pin-sense */
         {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
         { } /* end */
 };
 
@@ -3877,7 +3885,6 @@ static int patch_ad1884a(struct hda_codec *codec)
                 spec->mixers[0] = ad1884a_mobile_mixers;
                 spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
                 spec->multiout.dig_out_nid = 0;
-                spec->input_mux = &ad1884a_mobile_capture_source;
                 codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
                 codec->patch_ops.init = ad1884a_hp_init;
                 break;
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index c73ce074a6ea..6ef57fbfb6eb 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -611,6 +611,7 @@ static const char *cmi9880_models[CMI_MODELS] = {
 
 static struct snd_pci_quirk cmi9880_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", CMI_FULL_DIG),
+        SND_PCI_QUIRK(0x1854, 0x002b, "LG LS75", CMI_MINIMAL),
         SND_PCI_QUIRK(0x1854, 0x0032, "LG", CMI_FULL_DIG),
         {} /* terminator */
 };
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 864b2f598c38..b0a2a262ece2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -853,6 +853,7 @@ do_sku:
                 case 0x10ec0269:
                 case 0x10ec0862:
                 case 0x10ec0662:        
+                case 0x10ec0889:
                         snd_hda_codec_write(codec, 0x14, 0,
                                             AC_VERB_SET_EAPD_BTLENABLE, 2);
                         snd_hda_codec_write(codec, 0x15, 0,
@@ -877,6 +878,7 @@ do_sku:
                 case 0x10ec0883:
                 case 0x10ec0885:
                 case 0x10ec0888:
+                case 0x10ec0889:
                         snd_hda_codec_write(codec, 0x20, 0,
                                             AC_VERB_SET_COEF_INDEX, 7);
                         tmp = snd_hda_codec_read(codec, 0x20, 0,
@@ -940,7 +942,6 @@ do_sku:
                             AC_VERB_SET_UNSOLICITED_ENABLE,
                             AC_USRSP_EN | ALC880_HP_EVENT);
         spec->unsol_event = alc_sku_unsol_event;
-        spec->init_hook = alc_sku_automute;     
 }
 
 /*
@@ -7743,6 +7744,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
         SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
         SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
+        SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
         SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC),
         SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD),
@@ -8640,6 +8642,7 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
 
         {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
         {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        {}
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
@@ -10510,6 +10513,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
         SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA),
         SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA),
+        SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
         SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER),
         SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1),
         SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 393f7fd2b1be..a4f44a00bae8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -840,7 +840,7 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
         STAC_INPUT_SOURCE(1),
         HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
-        HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
         HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT),
         { } /* end */
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 52b1d81a26f7..e7e43524f8c7 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -447,6 +447,23 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = {
         },
 };
 
+static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = {
+        .substreams = 1,
+        .channels_min = 2,
+        .channels_max = 8,
+        .nid = 0x10, /* NID to query formats and rates */
+        /* We got noisy outputs on the right channel on VT1708 when
+         * 24bit samples are used.  Until any workaround is found,
+         * disable the 24bit format, so far.
+         */
+        .formats = SNDRV_PCM_FMTBIT_S16_LE,
+        .ops = {
+                .open = via_playback_pcm_open,
+                .prepare = via_playback_pcm_prepare,
+                .cleanup = via_playback_pcm_cleanup
+        },
+};
+
 static struct hda_pcm_stream vt1708_pcm_analog_capture = {
         .substreams = 2,
         .channels_min = 2,
@@ -899,6 +916,9 @@ static int patch_vt1708(struct hda_codec *codec)
         
         spec->stream_name_analog = "VT1708 Analog";
         spec->stream_analog_playback = &vt1708_pcm_analog_playback;
+        /* disable 32bit format on VT1708 */
+        if (codec->vendor_id == 0x11061708)
+                spec->stream_analog_playback = &vt1708_pcm_analog_s16_playback;
         spec->stream_analog_capture = &vt1708_pcm_analog_capture;
 
         spec->stream_name_digital = "VT1708 Digital";
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index cc0cddadd589..6facac5aed90 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -936,11 +936,13 @@ static int add_controls(struct oxygen *chip,
 
         for (i = 0; i < count; ++i) {
                 template = controls[i];
-                err = chip->model->control_filter(&template);
-                if (err < 0)
-                        return err;
-                if (err == 1)
-                        continue;
+                if (chip->model->control_filter) {
+                        err = chip->model->control_filter(&template);
+                        if (err < 0)
+                                return err;
+                        if (err == 1)
+                                continue;
+                }
                 if (!strcmp(template.name, "Master Playback Volume") &&
                     chip->model->dac_tlv) {
                         template.tlv.p = chip->model->dac_tlv;