[ALSA] hda-codec - Add auto-mute function to Sony VAIO with STAC9872
[pandora-kernel.git] / sound / pci / hda / patch_realtek.c
index 1950875..7466df4 100644 (file)
@@ -94,6 +94,7 @@ enum {
        ALC262_HP_BPC_D7000_WF,
        ALC262_BENQ_ED8,
        ALC262_SONY_ASSAMD,
+       ALC262_BENQ_T31,
        ALC262_AUTO,
        ALC262_MODEL_LAST /* last tag */
 };
@@ -101,6 +102,8 @@ enum {
 /* ALC268 models */
 enum {
        ALC268_3ST,
+       ALC268_TOSHIBA,
+       ALC268_ACER,
        ALC268_AUTO,
        ALC268_MODEL_LAST /* last tag */
 };
@@ -122,11 +125,13 @@ enum {
 /* ALC861-VD models */
 enum {
        ALC660VD_3ST,
+       ALC660VD_3ST_DIG,
        ALC861VD_3ST,
        ALC861VD_3ST_DIG,
        ALC861VD_6ST_DIG,
        ALC861VD_LENOVO,
        ALC861VD_DALLAS,
+       ALC861VD_HP,
        ALC861VD_AUTO,
        ALC861VD_MODEL_LAST,
 };
@@ -151,6 +156,7 @@ enum {
        ALC882_TARGA,
        ALC882_ASUS_A7J,
        ALC885_MACPRO,
+       ALC885_IMAC24,
        ALC882_AUTO,
        ALC882_MODEL_LAST,
 };
@@ -164,12 +170,15 @@ enum {
        ALC883_TARGA_DIG,
        ALC883_TARGA_2ch_DIG,
        ALC883_ACER,
+       ALC883_ACER_ASPIRE,
        ALC883_MEDION,
        ALC883_MEDION_MD2,      
        ALC883_LAPTOP_EAPD,
        ALC883_LENOVO_101E_2ch,
        ALC883_LENOVO_NB0763,
        ALC888_LENOVO_MS7195_DIG,               
+       ALC888_6ST_HP,
+       ALC888_3ST_HP,
        ALC883_AUTO,
        ALC883_MODEL_LAST,
 };
@@ -234,6 +243,10 @@ struct alc_spec {
        /* for pin sensing */
        unsigned int sense_updated: 1;
        unsigned int jack_present: 1;
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       struct hda_loopback_check loopback;
+#endif
 };
 
 /*
@@ -258,6 +271,9 @@ struct alc_config_preset {
        const struct hda_input_mux *input_mux;
        void (*unsol_event)(struct hda_codec *, unsigned int);
        void (*init_hook)(struct hda_codec *);
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       struct hda_amp_list *loopbacks;
+#endif
 };
 
 
@@ -436,8 +452,9 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
        change = pinctl != alc_pin_mode_values[val];
        if (change) {
                /* Set pin mode to that requested */
-               snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
-                                   alc_pin_mode_values[val]);
+               snd_hda_codec_write_cache(codec, nid, 0,
+                                         AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                         alc_pin_mode_values[val]);
 
                /* Also enable the retasking pin's input/output as required 
                 * for the requested pin mode.  Enum values of 2 or less are
@@ -450,19 +467,15 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
                 * this turns out to be necessary in the future.
                 */
                if (val <= 2) {
-                       snd_hda_codec_write(codec, nid, 0,
-                                           AC_VERB_SET_AMP_GAIN_MUTE,
-                                           AMP_OUT_MUTE);
-                       snd_hda_codec_write(codec, nid, 0,
-                                           AC_VERB_SET_AMP_GAIN_MUTE,
-                                           AMP_IN_UNMUTE(0));
+                       snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+                                                HDA_AMP_MUTE, HDA_AMP_MUTE);
+                       snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+                                                HDA_AMP_MUTE, 0);
                } else {
-                       snd_hda_codec_write(codec, nid, 0,
-                                           AC_VERB_SET_AMP_GAIN_MUTE,
-                                           AMP_IN_MUTE(0));
-                       snd_hda_codec_write(codec, nid, 0,
-                                           AC_VERB_SET_AMP_GAIN_MUTE,
-                                           AMP_OUT_UNMUTE);
+                       snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
+                                                HDA_AMP_MUTE, HDA_AMP_MUTE);
+                       snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+                                                HDA_AMP_MUTE, 0);
                }
        }
        return change;
@@ -481,15 +494,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
  * needed for any "production" models.
  */
 #ifdef CONFIG_SND_DEBUG
-static int alc_gpio_data_info(struct snd_kcontrol *kcontrol,
-                             struct snd_ctl_elem_info *uinfo)
-{
-       uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-       uinfo->count = 1;
-       uinfo->value.integer.min = 0;
-       uinfo->value.integer.max = 1;
-       return 0;
-}
+#define alc_gpio_data_info     snd_ctl_boolean_mono_info
 
 static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
                             struct snd_ctl_elem_value *ucontrol)
@@ -522,7 +527,8 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
                gpio_data &= ~mask;
        else
                gpio_data |= mask;
-       snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_GPIO_DATA, gpio_data);
+       snd_hda_codec_write_cache(codec, nid, 0,
+                                 AC_VERB_SET_GPIO_DATA, gpio_data);
 
        return change;
 }
@@ -542,15 +548,7 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
  * necessary.
  */
 #ifdef CONFIG_SND_DEBUG
-static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol,
-                              struct snd_ctl_elem_info *uinfo)
-{
-       uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
-       uinfo->count = 1;
-       uinfo->value.integer.min = 0;
-       uinfo->value.integer.max = 1;
-       return 0;
-}
+#define alc_spdif_ctrl_info    snd_ctl_boolean_mono_info
 
 static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
                              struct snd_ctl_elem_value *ucontrol)
@@ -583,8 +581,8 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
                ctrl_data &= ~mask;
        else
                ctrl_data |= mask;
-       snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
-                           ctrl_data);
+       snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+                                 ctrl_data);
 
        return change;
 }
@@ -633,6 +631,9 @@ static void setup_preset(struct alc_spec *spec,
 
        spec->unsol_event = preset->unsol_event;
        spec->init_hook = preset->init_hook;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       spec->loopback.amplist = preset->loopbacks;
+#endif
 }
 
 /* Enable GPIO mask and set output */
@@ -719,6 +720,38 @@ static void alc_subsystem_id(struct hda_codec *codec,
        }
 }
 
+/*
+ * Fix-up pin default configurations
+ */
+
+struct alc_pincfg {
+       hda_nid_t nid;
+       u32 val;
+};
+
+static void alc_fix_pincfg(struct hda_codec *codec,
+                          const struct snd_pci_quirk *quirk,
+                          const struct alc_pincfg **pinfix)
+{
+       const struct alc_pincfg *cfg;
+
+       quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk);
+       if (!quirk)
+               return;
+
+       cfg = pinfix[quirk->value];
+       for (; cfg->nid; cfg++) {
+               int i;
+               u32 val = cfg->val;
+               for (i = 0; i < 4; i++) {
+                       snd_hda_codec_write(codec, cfg->nid, 0,
+                                   AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
+                                   val & 0xff);
+                       val >>= 8;
+               }
+       }
+}
+
 /*
  * ALC880 3-stack model
  *
@@ -1267,11 +1300,13 @@ static struct hda_verb alc880_volume_init_verbs[] = {
         * panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
 
        /*
         * Set up output mixers (0x0c - 0x0f)
@@ -1531,15 +1566,11 @@ static void alc880_uniwill_hp_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x16, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x16, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
+       snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 /* auto-toggle front mic */
@@ -1550,11 +1581,8 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x18, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x0b, 0, HDA_INPUT, 1,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x0b, 1, HDA_INPUT, 1,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
 }
 
 static void alc880_uniwill_automute(struct hda_codec *codec)
@@ -1586,11 +1614,8 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_INPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_INPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
 }
 
 static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -1598,19 +1623,14 @@ static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
        unsigned int present;
        
        present = snd_hda_codec_read(codec, 0x21, 0,
-                                    AC_VERB_GET_VOLUME_KNOB_CONTROL, 0) & 0x7f;
-
-       snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0,
-                                0x7f, present);
-       snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
-                                0x7f,  present);
-
-       snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
-                                0x7f,  present);
-       snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
-                                0x7f, present);
-
+                                    AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+       present &= HDA_AMP_VOLMASK;
+       snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
+                                HDA_AMP_VOLMASK, present);
+       snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
+                                HDA_AMP_VOLMASK, present);
 }
+
 static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
                                           unsigned int res)
 {
@@ -1831,8 +1851,8 @@ static struct hda_verb alc880_lg_init_verbs[] = {
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
        /* mute all amp mixer inputs */
        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
        /* line-in to input */
        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
@@ -1863,11 +1883,9 @@ static void alc880_lg_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x1b, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1885,41 +1903,62 @@ static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res)
  * Pin assignment:
  *   Speaker-out: 0x14
  *   Mic-In: 0x18
- *   Built-in Mic-In: 0x19 (?)
- *   HP-Out: 0x1b
+ *   Built-in Mic-In: 0x19
+ *   Line-In: 0x1b
+ *   HP-Out: 0x1a
  *   SPDIF-Out: 0x1e
  */
 
-/* seems analog CD is not working */
 static struct hda_input_mux alc880_lg_lw_capture_source = {
-       .num_items = 2,
+       .num_items = 3,
        .items = {
                { "Mic", 0x0 },
                { "Internal Mic", 0x1 },
+               { "Line In", 0x2 },
        },
 };
 
+#define alc880_lg_lw_modes alc880_threestack_modes
+
 static struct snd_kcontrol_new alc880_lg_lw_mixer[] = {
-       HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-       HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+       HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
        HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+       {
+               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+               .name = "Channel Mode",
+               .info = alc_ch_mode_info,
+               .get = alc_ch_mode_get,
+               .put = alc_ch_mode_put,
+       },
        { } /* end */
 };
 
 static struct hda_verb alc880_lg_lw_init_verbs[] = {
+       {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
+       {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
+       {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
+
        /* set capture source to mic-in */
        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
        /* speaker-out */
        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
        /* HP-out */
-       {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
        {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
        /* mic-in to input */
@@ -1941,11 +1980,9 @@ static void alc880_lg_lw_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x1b, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1957,6 +1994,24 @@ static void alc880_lg_lw_unsol_event(struct hda_codec *codec, unsigned int res)
                alc880_lg_lw_automute(codec);
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc880_loopbacks[] = {
+       { 0x0b, HDA_INPUT, 0 },
+       { 0x0b, HDA_INPUT, 1 },
+       { 0x0b, HDA_INPUT, 2 },
+       { 0x0b, HDA_INPUT, 3 },
+       { 0x0b, HDA_INPUT, 4 },
+       { } /* end */
+};
+
+static struct hda_amp_list alc880_lg_loopbacks[] = {
+       { 0x0b, HDA_INPUT, 1 },
+       { 0x0b, HDA_INPUT, 6 },
+       { 0x0b, HDA_INPUT, 7 },
+       { } /* end */
+};
+#endif
+
 /*
  * Common callbacks
  */
@@ -1983,24 +2038,11 @@ static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
                spec->unsol_event(codec, res);
 }
 
-#ifdef CONFIG_PM
-/*
- * resume
- */
-static int alc_resume(struct hda_codec *codec)
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 {
        struct alc_spec *spec = codec->spec;
-       int i;
-
-       alc_init(codec);
-       for (i = 0; i < spec->num_mixers; i++)
-               snd_hda_resume_ctls(codec, spec->mixers[i]);
-       if (spec->multiout.dig_out_nid)
-               snd_hda_resume_spdif_out(codec);
-       if (spec->dig_in_nid)
-               snd_hda_resume_spdif_in(codec);
-
-       return 0;
+       return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
 }
 #endif
 
@@ -2235,8 +2277,8 @@ static struct hda_codec_ops alc_patch_ops = {
        .init = alc_init,
        .free = alc_free,
        .unsol_event = alc_unsol_event,
-#ifdef CONFIG_PM
-       .resume = alc_resume,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       .check_power_status = alc_check_power_status,
 #endif
 };
 
@@ -2334,11 +2376,14 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
                                     AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
        new_ctl = ctls[ucontrol->value.enumerated.item[0]];
        if (old_ctl != new_ctl) {
-               snd_hda_codec_write(codec, nid, 0,
-                                   AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                   (ucontrol->value.enumerated.item[0] >= 3 ?
-                                    0xb080 : 0xb000));
+               int val;
+               snd_hda_codec_write_cache(codec, nid, 0,
+                                         AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                         new_ctl);
+               val = ucontrol->value.enumerated.item[0] >= 3 ?
+                       HDA_AMP_MUTE : 0;
+               snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, val);
                return 1;
        }
        return 0;
@@ -2381,7 +2426,8 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
        sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
        if (ucontrol->value.enumerated.item[0] != sel) {
                sel = ucontrol->value.enumerated.item[0] & 3;
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, sel);
+               snd_hda_codec_write_cache(codec, nid, 0,
+                                         AC_VERB_SET_CONNECT_SEL, sel);
                return 1;
        }
        return 0;
@@ -2858,16 +2904,19 @@ static struct alc_config_preset alc880_presets[] = {
                .input_mux = &alc880_lg_capture_source,
                .unsol_event = alc880_lg_unsol_event,
                .init_hook = alc880_lg_automute,
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+               .loopbacks = alc880_lg_loopbacks,
+#endif
        },
        [ALC880_LG_LW] = {
                .mixers = { alc880_lg_lw_mixer },
                .init_verbs = { alc880_volume_init_verbs,
                                alc880_lg_lw_init_verbs },
-               .num_dacs = 1,
+               .num_dacs = ARRAY_SIZE(alc880_dac_nids),
                .dac_nids = alc880_dac_nids,
                .dig_out_nid = ALC880_DIGOUT_NID,
-               .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
-               .channel_mode = alc880_2_jack_modes,
+               .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes),
+               .channel_mode = alc880_lg_lw_modes,
                .input_mux = &alc880_lg_lw_capture_source,
                .unsol_event = alc880_lg_lw_unsol_event,
                .init_hook = alc880_lg_lw_automute,
@@ -3341,6 +3390,10 @@ static int patch_alc880(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC880_AUTO)
                spec->init_hook = alc880_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc880_loopbacks;
+#endif
 
        return 0;
 }
@@ -3689,12 +3742,12 @@ static struct hda_verb alc260_init_verbs[] = {
        /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
         * Line In 2 = 0x03
         */
-       /* mute CD */
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-       /* mute Line In */
-       {0x07,  AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       /* mute Mic */
-       {0x07,  AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+       /* mute analog inputs */
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
        /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
        /* mute Front out path */
        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -3739,12 +3792,12 @@ static struct hda_verb alc260_hp_init_verbs[] = {
        /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
         * Line In 2 = 0x03
         */
-       /* unmute CD */
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
-       /* unmute Line In */
-       {0x07,  AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
-       /* unmute Mic */
-       {0x07,  AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+       /* mute analog inputs */
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
        /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
        /* Unmute Front out path */
        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
@@ -3789,12 +3842,12 @@ static struct hda_verb alc260_hp_3013_init_verbs[] = {
        /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
         * Line In 2 = 0x03
         */
-       /* unmute CD */
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
-       /* unmute Line In */
-       {0x07,  AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
-       /* unmute Mic */
-       {0x07,  AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+       /* mute analog inputs */
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
        /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
        /* Unmute Front out path */
        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
@@ -4011,13 +4064,17 @@ static void alc260_replacer_672v_automute(struct hda_codec *codec)
         present = snd_hda_codec_read(codec, 0x0f, 0,
                                      AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
        if (present) {
-               snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 1);
-               snd_hda_codec_write(codec, 0x0f, 0,
-                                   AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+               snd_hda_codec_write_cache(codec, 0x01, 0,
+                                         AC_VERB_SET_GPIO_DATA, 1);
+               snd_hda_codec_write_cache(codec, 0x0f, 0,
+                                         AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                         PIN_HP);
        } else {
-               snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0);
-               snd_hda_codec_write(codec, 0x0f, 0,
-                                   AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+               snd_hda_codec_write_cache(codec, 0x01, 0,
+                                         AC_VERB_SET_GPIO_DATA, 0);
+               snd_hda_codec_write_cache(codec, 0x0f, 0,
+                                         AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                         PIN_OUT);
        }
 }
 
@@ -4412,11 +4469,12 @@ static struct hda_verb alc260_volume_init_verbs[] = {
         * front panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       /* mute analog inputs */
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /*
         * Set up output mixers (0x08 - 0x0a)
@@ -4493,6 +4551,17 @@ static void alc260_auto_init(struct hda_codec *codec)
        alc260_auto_init_analog_input(codec);
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc260_loopbacks[] = {
+       { 0x07, HDA_INPUT, 0 },
+       { 0x07, HDA_INPUT, 1 },
+       { 0x07, HDA_INPUT, 2 },
+       { 0x07, HDA_INPUT, 3 },
+       { 0x07, HDA_INPUT, 4 },
+       { } /* end */
+};
+#endif
+
 /*
  * ALC260 configurations
  */
@@ -4692,6 +4761,10 @@ static int patch_alc260(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC260_AUTO)
                spec->init_hook = alc260_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc260_loopbacks;
+#endif
 
        return 0;
 }
@@ -4754,12 +4827,13 @@ static int alc882_mux_enum_put(struct snd_kcontrol *kcontrol,
        idx = ucontrol->value.enumerated.item[0];
        if (idx >= imux->num_items)
                idx = imux->num_items - 1;
-       if (*cur_val == idx && !codec->in_resume)
+       if (*cur_val == idx)
                return 0;
        for (i = 0; i < imux->num_items; i++) {
-               unsigned int v = (i == idx) ? 0x7000 : 0x7080;
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                   v | (imux->items[i].index << 8));
+               unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+               snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+                                        imux->items[i].index,
+                                        HDA_AMP_MUTE, v);
        }
        *cur_val = idx;
        return 1;
@@ -5061,6 +5135,56 @@ static struct hda_verb alc882_macpro_init_verbs[] = {
        { }
 };
 
+/* iMac 24 mixer. */
+static struct snd_kcontrol_new alc885_imac24_mixer[] = {
+       HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+       HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT),
+       { } /* end */
+};
+
+/* iMac 24 init verbs. */
+static struct hda_verb alc885_imac24_init_verbs[] = {
+       /* Internal speakers: output 0 (0x0c) */
+       {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+       {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+       {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+       /* Internal speakers: output 0 (0x0c) */
+       {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+       {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+       {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+       /* Headphone: output 0 (0x0c) */
+       {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+       {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+       {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+       {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+       /* Front Mic: input vref at 80% */
+       {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+       {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+       { }
+};
+
+/* Toggle speaker-output according to the hp-jack state */
+static void alc885_imac24_automute(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_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+       snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+/* Processes unsolicited events. */
+static void alc885_imac24_unsol_event(struct hda_codec *codec,
+                                     unsigned int res)
+{
+       /* Headphone insertion or removal. */
+       if ((res >> 26) == ALC880_HP_EVENT)
+               alc885_imac24_automute(codec);
+}
+
 static struct hda_verb alc882_targa_verbs[] = {
        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -5086,11 +5210,10 @@ static void alc882_targa_automute(struct hda_codec *codec)
  
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA, present ? 1 : 3);
+       snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+       snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+                                 present ? 1 : 3);
 }
 
 static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -5153,6 +5276,20 @@ static void alc882_gpio_mute(struct hda_codec *codec, int pin, int muted)
                            AC_VERB_SET_GPIO_DATA, gpiostate);
 }
 
+/* set up GPIO at initialization */
+static void alc885_macpro_init_hook(struct hda_codec *codec)
+{
+       alc882_gpio_mute(codec, 0, 0);
+       alc882_gpio_mute(codec, 1, 0);
+}
+
+/* set up GPIO and update auto-muting at initialization */
+static void alc885_imac24_init_hook(struct hda_codec *codec)
+{
+       alc885_macpro_init_hook(codec);
+       alc885_imac24_automute(codec);
+}
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -5167,17 +5304,17 @@ static struct hda_verb alc882_auto_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-       /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+       /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
         * mixer widget
         * Note: PASD motherboards uses the Line In 2 as the input for
         * front panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /*
         * Set up output mixers (0x0c - 0x0f)
@@ -5266,6 +5403,10 @@ static struct snd_kcontrol_new alc882_capture_mixer[] = {
        { } /* end */
 };
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc882_loopbacks       alc880_loopbacks
+#endif
+
 /* pcm configuration: identiacal with ALC880 */
 #define alc882_pcm_analog_playback     alc880_pcm_analog_playback
 #define alc882_pcm_analog_capture      alc880_pcm_analog_capture
@@ -5281,6 +5422,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
        [ALC882_ARIMA]          = "arima",
        [ALC882_W2JC]           = "w2jc",
        [ALC885_MACPRO]         = "macpro",
+       [ALC885_IMAC24]         = "imac24",
        [ALC882_AUTO]           = "auto",
 };
 
@@ -5291,6 +5433,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
        SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
        SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+       SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
        SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
        SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
        {}
@@ -5351,6 +5494,20 @@ static struct alc_config_preset alc882_presets[] = {
                .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
                .channel_mode = alc882_ch_modes,
                .input_mux = &alc882_capture_source,
+               .init_hook = alc885_macpro_init_hook,
+       },
+       [ALC885_IMAC24] = {
+               .mixers = { alc885_imac24_mixer },
+               .init_verbs = { alc885_imac24_init_verbs },
+               .num_dacs = ARRAY_SIZE(alc882_dac_nids),
+               .dac_nids = alc882_dac_nids,
+               .dig_out_nid = ALC882_DIGOUT_NID,
+               .dig_in_nid = ALC882_DIGIN_NID,
+               .num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+               .channel_mode = alc882_ch_modes,
+               .input_mux = &alc882_capture_source,
+               .unsol_event = alc885_imac24_unsol_event,
+               .init_hook = alc885_imac24_init_hook,
        },
        [ALC882_TARGA] = {
                .mixers = { alc882_targa_mixer, alc882_chmode_mixer,
@@ -5385,6 +5542,29 @@ static struct alc_config_preset alc882_presets[] = {
 };
 
 
+/*
+ * Pin config fixes
+ */
+enum { 
+       PINFIX_ABIT_AW9D_MAX
+};
+
+static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
+       { 0x15, 0x01080104 }, /* side */
+       { 0x16, 0x01011012 }, /* rear */
+       { 0x17, 0x01016011 }, /* clfe */
+       { }
+};
+
+static const struct alc_pincfg *alc882_pin_fixes[] = {
+       [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+};
+
+static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+       SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
+       {}
+};
+
 /*
  * BIOS auto configuration
  */
@@ -5501,6 +5681,9 @@ static int patch_alc882(struct hda_codec *codec)
                case 0x106b0c00: /* Mac Pro */
                        board_config = ALC885_MACPRO;
                        break;
+               case 0x106b1000: /* iMac 24 */
+                       board_config = ALC885_IMAC24;
+                       break;
                default:
                        printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
                                         "trying auto-probe from BIOS...\n");
@@ -5508,6 +5691,8 @@ static int patch_alc882(struct hda_codec *codec)
                }
        }
 
+       alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+
        if (board_config == ALC882_AUTO) {
                /* automatic parse from the BIOS config */
                err = alc882_parse_auto_config(codec);
@@ -5525,11 +5710,6 @@ static int patch_alc882(struct hda_codec *codec)
        if (board_config != ALC882_AUTO)
                setup_preset(spec, &alc882_presets[board_config]);
 
-       if (board_config == ALC885_MACPRO) {
-               alc882_gpio_mute(codec, 0, 0);
-               alc882_gpio_mute(codec, 1, 0);
-       }
-
        spec->stream_name_analog = "ALC882 Analog";
        spec->stream_analog_playback = &alc882_pcm_analog_playback;
        spec->stream_analog_capture = &alc882_pcm_analog_capture;
@@ -5560,6 +5740,10 @@ static int patch_alc882(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC882_AUTO)
                spec->init_hook = alc882_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc882_loopbacks;
+#endif
 
        return 0;
 }
@@ -5637,12 +5821,13 @@ static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol,
        idx = ucontrol->value.enumerated.item[0];
        if (idx >= imux->num_items)
                idx = imux->num_items - 1;
-       if (*cur_val == idx && !codec->in_resume)
+       if (*cur_val == idx)
                return 0;
        for (i = 0; i < imux->num_items; i++) {
-               unsigned int v = (i == idx) ? 0x7000 : 0x7080;
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                   v | (imux->items[i].index << 8));
+               unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+               snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+                                        imux->items[i].index,
+                                        HDA_AMP_MUTE, v);
        }
        *cur_val = idx;
        return 1;
@@ -6002,46 +6187,150 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = {
        { } /* end */
 };     
 
-static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+static struct snd_kcontrol_new alc888_6st_hp_mixer[] = {
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+       HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+       HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+       HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+       HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+       HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+       HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+       HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+       HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+       HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
        {
                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-               .name = "Channel Mode",
-               .info = alc_ch_mode_info,
-               .get = alc_ch_mode_get,
-               .put = alc_ch_mode_put,
+               /* .name = "Capture Source", */
+               .name = "Input Source",
+               .count = 2,
+               .info = alc883_mux_enum_info,
+               .get = alc883_mux_enum_get,
+               .put = alc883_mux_enum_put,
        },
        { } /* end */
 };
 
-static struct hda_verb alc883_init_verbs[] = {
-       /* ADC1: mute amp left and right */
-       {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-       /* ADC2: mute amp left and right */
-       {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-       /* Front mixer: unmute input/output amp left and right (volume = 0) */
-       {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-       {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-       /* Rear mixer */
-       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-       /* CLFE mixer */
-       {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-       {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-       /* Side mixer */
-       {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-       {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-       {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+static struct snd_kcontrol_new alc888_3st_hp_mixer[] = {
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
+       HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
+       HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+       HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+       HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+       HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+       HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+       HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+       HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+       HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+       {
+               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+               /* .name = "Capture Source", */
+               .name = "Input Source",
+               .count = 2,
+               .info = alc883_mux_enum_info,
+               .get = alc883_mux_enum_get,
+               .put = alc883_mux_enum_put,
+       },
+       { } /* end */
+};
 
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+       HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+       HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+       {
+               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+               /* .name = "Capture Source", */
+               .name = "Input Source",
+               .count = 2,
+               .info = alc883_mux_enum_info,
+               .get = alc883_mux_enum_get,
+               .put = alc883_mux_enum_put,
+       },
+       { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+       {
+               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+               .name = "Channel Mode",
+               .info = alc_ch_mode_info,
+               .get = alc_ch_mode_get,
+               .put = alc_ch_mode_put,
+       },
+       { } /* end */
+};
+
+static struct hda_verb alc883_init_verbs[] = {
+       /* ADC1: mute amp left and right */
+       {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+       /* ADC2: mute amp left and right */
+       {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+       /* Front mixer: unmute input/output amp left and right (volume = 0) */
+       {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+       {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       /* Rear mixer */
+       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       /* CLFE mixer */
+       {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+       {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       /* Side mixer */
+       {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+       {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+       /* mute analog input loopbacks */
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /* Front Pin: output 0 (0x0c) */
        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
@@ -6133,6 +6422,42 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = {
        { } /* end */
 };
 
+static struct hda_verb alc888_6st_hp_verbs[] = {
+       {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},  /* Front: output 0 (0x0c) */
+       {0x15, AC_VERB_SET_CONNECT_SEL, 0x02},  /* Rear : output 2 (0x0e) */
+       {0x16, AC_VERB_SET_CONNECT_SEL, 0x01},  /* CLFE : output 1 (0x0d) */
+       {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},  /* Side : output 3 (0x0f) */
+       { }
+};
+
+static struct hda_verb alc888_3st_hp_verbs[] = {
+       {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},  /* Front: output 0 (0x0c) */
+       {0x18, AC_VERB_SET_CONNECT_SEL, 0x01},  /* Rear : output 1 (0x0d) */
+       {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},  /* CLFE : output 2 (0x0e) */
+       { }
+};
+
+static struct hda_verb alc888_3st_hp_2ch_init[] = {
+       { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+       { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+       { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+       { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+       { }
+};
+
+static struct hda_verb alc888_3st_hp_6ch_init[] = {
+       { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+       { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+       { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+       { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+       { }
+};
+
+static struct hda_channel_mode alc888_3st_hp_modes[2] = {
+       { 2, alc888_3st_hp_2ch_init },
+       { 6, alc888_3st_hp_6ch_init },
+};
+
 /* toggle front-jack and RCA according to the hp-jack state */
 static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
 {
@@ -6140,15 +6465,10 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec)
  
        present = snd_hda_codec_read(codec, 0x1b, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       
+       snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
 
 /* toggle RCA according to the front-jack state */
@@ -6158,12 +6478,10 @@ static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec)
  
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
+
 static void alc883_lenovo_ms7195_unsol_event(struct hda_codec *codec,
                                             unsigned int res)
 {
@@ -6190,10 +6508,8 @@ static void alc883_medion_md2_automute(struct hda_codec *codec)
  
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
 
 static void alc883_medion_md2_unsol_event(struct hda_codec *codec,
@@ -6211,13 +6527,11 @@ static void alc883_tagra_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_write(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
-                           present ? 1 : 3);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
+       snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA,
+                                 present ? 1 : 3);
 }
 
 static void alc883_tagra_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -6233,11 +6547,9 @@ static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
@@ -6247,15 +6559,11 @@ static void alc883_lenovo_101e_all_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x1b, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
+       snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
@@ -6267,6 +6575,44 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
                alc883_lenovo_101e_ispeaker_automute(codec);
 }
 
+/* toggle speaker-output according to the hp-jack state */
+static void alc883_acer_aspire_automute(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_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+       snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+}
+
+static void alc883_acer_aspire_unsol_event(struct hda_codec *codec,
+                                          unsigned int res)
+{
+       if ((res >> 26) == ALC880_HP_EVENT)
+               alc883_acer_aspire_automute(codec);
+}
+
+static struct hda_verb alc883_acer_eapd_verbs[] = {
+       /* HP Pin: output 0 (0x0c) */
+       {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+       {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+       {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+       /* Front Pin: output 0 (0x0c) */
+       {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+       {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+       {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+       {0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* eanable EAPD on medion laptop */
+       {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+       {0x20, AC_VERB_SET_PROC_COEF, 0x3050},
+       /* enable unsolicited event */
+       {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+       { }
+};
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -6279,17 +6625,17 @@ static struct hda_verb alc883_auto_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-       /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+       /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
         * mixer widget
         * Note: PASD motherboards uses the Line In 2 as the input for
         * front panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /*
         * Set up output mixers (0x0c - 0x0f)
@@ -6352,6 +6698,10 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = {
        { } /* end */
 };
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc883_loopbacks       alc880_loopbacks
+#endif
+
 /* pcm configuration: identiacal with ALC880 */
 #define alc883_pcm_analog_playback     alc880_pcm_analog_playback
 #define alc883_pcm_analog_capture      alc880_pcm_analog_capture
@@ -6369,12 +6719,15 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
        [ALC883_TARGA_DIG]      = "targa-dig",
        [ALC883_TARGA_2ch_DIG]  = "targa-2ch-dig",
        [ALC883_ACER]           = "acer",
+       [ALC883_ACER_ASPIRE]    = "acer-aspire",
        [ALC883_MEDION]         = "medion",
        [ALC883_MEDION_MD2]     = "medion-md2",
        [ALC883_LAPTOP_EAPD]    = "laptop-eapd",
        [ALC883_LENOVO_101E_2ch] = "lenovo-101e",
        [ALC883_LENOVO_NB0763]  = "lenovo-nb0763",
        [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig",
+       [ALC888_6ST_HP]         = "6stack-hp",
+       [ALC888_3ST_HP]         = "3stack-hp",
        [ALC883_AUTO]           = "auto",
 };
 
@@ -6389,6 +6742,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG),
        SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG),
        SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
+       SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
        SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
        SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
        SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
@@ -6397,10 +6751,13 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0x3fc1, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
+       SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
        SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+       SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
+       SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
        SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER),
        SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
        SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
@@ -6409,6 +6766,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
        SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch),
        SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763),
        SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763),
+       SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP),
+       SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
+       SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
        SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2),
        {}
 };
@@ -6496,8 +6856,7 @@ static struct alc_config_preset alc883_presets[] = {
                .init_hook = alc883_tagra_automute,
        },
        [ALC883_ACER] = {
-               .mixers = { alc883_base_mixer,
-                           alc883_chmode_mixer },
+               .mixers = { alc883_base_mixer },
                /* On TravelMate laptops, GPIO 0 enables the internal speaker
                 * and the headphone jack.  Turn this on and rely on the
                 * standard mute methods whenever the user wants to turn
@@ -6512,6 +6871,20 @@ static struct alc_config_preset alc883_presets[] = {
                .channel_mode = alc883_3ST_2ch_modes,
                .input_mux = &alc883_capture_source,
        },
+       [ALC883_ACER_ASPIRE] = {
+               .mixers = { alc883_acer_aspire_mixer },
+               .init_verbs = { alc883_init_verbs, alc883_acer_eapd_verbs },
+               .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+               .dac_nids = alc883_dac_nids,
+               .dig_out_nid = ALC883_DIGOUT_NID,
+               .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+               .adc_nids = alc883_adc_nids,
+               .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+               .channel_mode = alc883_3ST_2ch_modes,
+               .input_mux = &alc883_capture_source,
+               .unsol_event = alc883_acer_aspire_unsol_event,
+               .init_hook = alc883_acer_aspire_automute,
+       },
        [ALC883_MEDION] = {
                .mixers = { alc883_fivestack_mixer,
                            alc883_chmode_mixer },
@@ -6540,8 +6913,7 @@ static struct alc_config_preset alc883_presets[] = {
                .init_hook = alc883_medion_md2_automute,
        },      
        [ALC883_LAPTOP_EAPD] = {
-               .mixers = { alc883_base_mixer,
-                           alc883_chmode_mixer },
+               .mixers = { alc883_base_mixer },
                .init_verbs = { alc883_init_verbs, alc882_eapd_verbs },
                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
                .dac_nids = alc883_dac_nids,
@@ -6593,6 +6965,31 @@ static struct alc_config_preset alc883_presets[] = {
                .unsol_event = alc883_lenovo_ms7195_unsol_event,
                .init_hook = alc888_lenovo_ms7195_front_automute,
        },      
+       [ALC888_6ST_HP] = {
+               .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer },
+               .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs },
+               .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+               .dac_nids = alc883_dac_nids,
+               .dig_out_nid = ALC883_DIGOUT_NID,
+               .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+               .adc_nids = alc883_adc_nids,
+               .dig_in_nid = ALC883_DIGIN_NID,
+               .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+               .channel_mode = alc883_sixstack_modes,
+               .input_mux = &alc883_capture_source,
+       },
+       [ALC888_3ST_HP] = {
+               .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer },
+               .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs },
+               .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+               .dac_nids = alc883_dac_nids,
+               .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+               .adc_nids = alc883_adc_nids,
+               .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes),
+               .channel_mode = alc888_3st_hp_modes,
+               .need_dac_fix = 1,
+               .input_mux = &alc883_capture_source,
+       },
 };
 
 
@@ -6746,6 +7143,10 @@ static int patch_alc883(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC883_AUTO)
                spec->init_hook = alc883_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc883_loopbacks;
+#endif
 
        return 0;
 }
@@ -6856,9 +7257,18 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
        { } /* end */
 };
 
+static struct hda_bind_ctls alc262_sony_bind_sw = {
+       .ops = &snd_hda_bind_sw,
+       .values = {
+               HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT),
+               HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+               0,
+       },
+};
+
 static struct snd_kcontrol_new alc262_sony_mixer[] = {
        HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-       HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+       HDA_BIND_SW("Front Playback Switch", &alc262_sony_bind_sw),
        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
        HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
@@ -6866,7 +7276,16 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = {
        { } /* end */
 };
 
-
+static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+       HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+       HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+       HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+       HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+       { } /* end */
+};
 
 #define alc262_capture_mixer           alc882_capture_mixer
 #define alc262_capture_alt_mixer       alc882_capture_alt_mixer
@@ -6885,17 +7304,17 @@ static struct hda_verb alc262_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-       /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+       /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
         * mixer widget
         * Note: PASD motherboards uses the Line In 2 as the input for
         * front panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /*
         * Set up output mixers (0x0c - 0x0e)
@@ -6976,34 +7395,26 @@ static struct hda_verb alc262_sony_unsol_verbs[] = {
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hippo_automute(struct hda_codec *codec, int force)
+static void alc262_hippo_automute(struct hda_codec *codec)
 {
        struct alc_spec *spec = codec->spec;
        unsigned int mute;
+       unsigned int present;
 
-       if (force || !spec->sense_updated) {
-               unsigned int present;
-               /* need to execute and sync at first */
-               snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
-               present = snd_hda_codec_read(codec, 0x15, 0,
-                                        AC_VERB_GET_PIN_SENSE, 0);
-               spec->jack_present = (present & 0x80000000) != 0;
-               spec->sense_updated = 1;
-       }
+       /* need to execute and sync at first */
+       snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0);
+       present = snd_hda_codec_read(codec, 0x15, 0,
+                                    AC_VERB_GET_PIN_SENSE, 0);
+       spec->jack_present = (present & 0x80000000) != 0;
        if (spec->jack_present) {
                /* mute internal speaker */
-               snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                        0x80, 0x80);
-               snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                        0x80, 0x80);
+               snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, HDA_AMP_MUTE);
        } else {
                /* unmute internal speaker if necessary */
                mute = snd_hda_codec_amp_read(codec, 0x15, 0, HDA_OUTPUT, 0);
-               snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                        0x80, mute & 0x80);
-               mute = snd_hda_codec_amp_read(codec, 0x15, 1, HDA_OUTPUT, 0);
-               snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                        0x80, mute & 0x80);
+               snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, mute);
        }
 }
 
@@ -7013,37 +7424,27 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec,
 {
        if ((res >> 26) != ALC880_HP_EVENT)
                return;
-       alc262_hippo_automute(codec, 1);
+       alc262_hippo_automute(codec);
 }
 
-static void alc262_hippo1_automute(struct hda_codec *codec, int force)
+static void alc262_hippo1_automute(struct hda_codec *codec)
 {
-       struct alc_spec *spec = codec->spec;
        unsigned int mute;
+       unsigned int present;
 
-       if (force || !spec->sense_updated) {
-               unsigned int present;
-               /* need to execute and sync at first */
-               snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
-               present = snd_hda_codec_read(codec, 0x1b, 0,
-                                        AC_VERB_GET_PIN_SENSE, 0);
-               spec->jack_present = (present & 0x80000000) != 0;
-               spec->sense_updated = 1;
-       }
-       if (spec->jack_present) {
+       snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+       present = snd_hda_codec_read(codec, 0x1b, 0,
+                                    AC_VERB_GET_PIN_SENSE, 0);
+       present = (present & 0x80000000) != 0;
+       if (present) {
                /* mute internal speaker */
-               snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                        0x80, 0x80);
-               snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                        0x80, 0x80);
+               snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, HDA_AMP_MUTE);
        } else {
                /* unmute internal speaker if necessary */
                mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
-               snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                        0x80, mute & 0x80);
-               mute = snd_hda_codec_amp_read(codec, 0x1b, 1, HDA_OUTPUT, 0);
-               snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                        0x80, mute & 0x80);
+               snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, mute);
        }
 }
 
@@ -7053,7 +7454,7 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec,
 {
        if ((res >> 26) != ALC880_HP_EVENT)
                return;
-       alc262_hippo1_automute(codec, 1);
+       alc262_hippo1_automute(codec);
 }
 
 /*
@@ -7105,18 +7506,13 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
        }
        if (spec->jack_present) {
                /* mute internal speaker */
-               snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                        0x80, 0x80);
-               snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                        0x80, 0x80);
+               snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, HDA_AMP_MUTE);
        } else {
                /* unmute internal speaker if necessary */
                mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0);
-               snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                        0x80, mute & 0x80);
-               mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0);
-               snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                        0x80, mute & 0x80);
+               snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                        HDA_AMP_MUTE, mute);
        }
 }
 
@@ -7130,23 +7526,14 @@ static void alc262_fujitsu_unsol_event(struct hda_codec *codec,
 }
 
 /* bind volumes of both NID 0x0c and 0x0d */
-static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol,
-                                        struct snd_ctl_elem_value *ucontrol)
-{
-       struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-       long *valp = ucontrol->value.integer.value;
-       int change;
-
-       change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0,
-                                         0x7f, valp[0] & 0x7f);
-       change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0,
-                                          0x7f, valp[1] & 0x7f);
-       snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0,
-                                0x7f, valp[0] & 0x7f);
-       snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0,
-                                0x7f, valp[1] & 0x7f);
-       return change;
-}
+static struct hda_bind_ctls alc262_fujitsu_bind_master_vol = {
+       .ops = &snd_hda_bind_vol,
+       .values = {
+               HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
+               HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT),
+               0
+       },
+};
 
 /* bind hp and internal speaker mute (with plug check) */
 static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
@@ -7157,24 +7544,18 @@ static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol,
        int change;
 
        change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                         0x80, valp[0] ? 0 : 0x80);
+                                         HDA_AMP_MUTE,
+                                         valp[0] ? 0 : HDA_AMP_MUTE);
        change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                          0x80, valp[1] ? 0 : 0x80);
-       if (change || codec->in_resume)
-               alc262_fujitsu_automute(codec, codec->in_resume);
+                                          HDA_AMP_MUTE,
+                                          valp[1] ? 0 : HDA_AMP_MUTE);
+       if (change)
+               alc262_fujitsu_automute(codec, 0);
        return change;
 }
 
 static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
-       {
-               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-               .name = "Master Playback Volume",
-               .info = snd_hda_mixer_amp_volume_info,
-               .get = snd_hda_mixer_amp_volume_get,
-               .put = alc262_fujitsu_master_vol_put,
-               .tlv = { .c = snd_hda_mixer_amp_tlv },
-               .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT),
-       },
+       HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol),
        {
                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
                .name = "Master Playback Switch",
@@ -7198,6 +7579,15 @@ static struct hda_verb alc262_EAPD_verbs[] = {
        {}
 };
 
+static struct hda_verb alc262_benq_t31_EAPD_verbs[] = {
+       {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+       {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+
+       {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
+       {0x20, AC_VERB_SET_PROC_COEF,  0x3050},
+       {}
+};
+
 /* add playback controls from the parsed DAC table */
 static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
                                             const struct auto_pin_cfg *cfg)
@@ -7293,17 +7683,17 @@ static struct hda_verb alc262_volume_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-       /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+       /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
         * mixer widget
         * Note: PASD motherboards uses the Line In 2 as the input for
         * front panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /*
         * Set up output mixers (0x0c - 0x0f)
@@ -7354,19 +7744,19 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-       /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+       /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
         * mixer widget
         * Note: PASD motherboards uses the Line In 2 as the input for
         * front panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
        
        /*
         * Set up output mixers (0x0c - 0x0e)
@@ -7441,20 +7831,20 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-       /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+       /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
         * mixer widget
         * Note: PASD motherboards uses the Line In 2 as the input for front
         * panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
        /*
         * Set up output mixers (0x0c - 0x0e)
         */
@@ -7524,6 +7914,10 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = {
        { }
 };
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc262_loopbacks       alc880_loopbacks
+#endif
+
 /* pcm configuration: identiacal with ALC880 */
 #define alc262_pcm_analog_playback     alc880_pcm_analog_playback
 #define alc262_pcm_analog_capture      alc880_pcm_analog_capture
@@ -7593,7 +7987,8 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
        [ALC262_HP_BPC]         = "hp-bpc",
        [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000",
        [ALC262_BENQ_ED8]       = "benq",
-       [ALC262_BENQ_ED8]       = "sony-assamd",
+       [ALC262_BENQ_T31]       = "benq-t31",
+       [ALC262_SONY_ASSAMD]    = "sony-assamd",
        [ALC262_AUTO]           = "auto",
 };
 
@@ -7619,6 +8014,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
        SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
        SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1),
        SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
+       SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31),
+       SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
        SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
        SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
        SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
@@ -7647,6 +8044,7 @@ static struct alc_config_preset alc262_presets[] = {
                .channel_mode = alc262_modes,
                .input_mux = &alc262_capture_source,
                .unsol_event = alc262_hippo_unsol_event,
+               .init_hook = alc262_hippo_automute,
        },
        [ALC262_HIPPO_1] = {
                .mixers = { alc262_hippo1_mixer },
@@ -7659,6 +8057,7 @@ static struct alc_config_preset alc262_presets[] = {
                .channel_mode = alc262_modes,
                .input_mux = &alc262_capture_source,
                .unsol_event = alc262_hippo1_unsol_event,
+               .init_hook = alc262_hippo1_automute,
        },
        [ALC262_FUJITSU] = {
                .mixers = { alc262_fujitsu_mixer },
@@ -7723,6 +8122,19 @@ static struct alc_config_preset alc262_presets[] = {
                .channel_mode = alc262_modes,
                .input_mux = &alc262_capture_source,
                .unsol_event = alc262_hippo_unsol_event,
+               .init_hook = alc262_hippo_automute,
+       },
+       [ALC262_BENQ_T31] = {
+               .mixers = { alc262_benq_t31_mixer },
+               .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs },
+               .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+               .dac_nids = alc262_dac_nids,
+               .hp_nid = 0x03,
+               .num_channel_mode = ARRAY_SIZE(alc262_modes),
+               .channel_mode = alc262_modes,
+               .input_mux = &alc262_capture_source,
+               .unsol_event = alc262_hippo_unsol_event,
+               .init_hook = alc262_hippo_automute,
        },      
 };
 
@@ -7808,6 +8220,10 @@ static int patch_alc262(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC262_AUTO)
                spec->init_hook = alc262_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc262_loopbacks;
+#endif
                
        return 0;
 }
@@ -7842,6 +8258,64 @@ static struct snd_kcontrol_new alc268_base_mixer[] = {
        { }
 };
 
+static struct hda_verb alc268_eapd_verbs[] = {
+       {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+       {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+       { }
+};
+
+/* Toshiba specific */
+#define alc268_toshiba_automute        alc262_hippo_automute
+
+static struct hda_verb alc268_toshiba_verbs[] = {
+       {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+       { } /* end */
+};
+
+/* Acer specific */
+#define alc268_acer_bind_master_vol    alc262_fujitsu_bind_master_vol
+#define alc268_acer_master_sw_put      alc262_fujitsu_master_sw_put
+#define alc268_acer_automute   alc262_fujitsu_automute
+
+static struct snd_kcontrol_new alc268_acer_mixer[] = {
+       /* output mixer control */
+       HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol),
+       {
+               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+               .name = "Master Playback Switch",
+               .info = snd_hda_mixer_amp_switch_info,
+               .get = snd_hda_mixer_amp_switch_get,
+               .put = alc268_acer_master_sw_put,
+               .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+       },
+       { }
+};
+
+static struct hda_verb alc268_acer_verbs[] = {
+       {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+       {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+       {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+       { }
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc268_toshiba_unsol_event(struct hda_codec *codec,
+                                      unsigned int res)
+{
+       if ((res >> 28) != ALC880_HP_EVENT)
+               return;
+       alc268_toshiba_automute(codec);
+}
+
+static void alc268_acer_unsol_event(struct hda_codec *codec,
+                                      unsigned int res)
+{
+       if ((res >> 28) != ALC880_HP_EVENT)
+               return;
+       alc268_acer_automute(codec, 1);
+}
+
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
@@ -7951,14 +8425,16 @@ static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol,
        idx = ucontrol->value.enumerated.item[0];
        if (idx >= imux->num_items)
                idx = imux->num_items - 1;
-       if (*cur_val == idx && !codec->in_resume)
+       if (*cur_val == idx)
                return 0;
        for (i = 0; i < imux->num_items; i++) {
-               unsigned int v = (i == idx) ? 0x7000 : 0x7080;
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                   v | (imux->items[i].index << 8));
-                snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
-                                   idx );
+               unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+               snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+                                        imux->items[i].index,
+                                        HDA_AMP_MUTE, v);
+                snd_hda_codec_write_cache(codec, nid, 0,
+                                         AC_VERB_SET_CONNECT_SEL,
+                                         idx );
        }
        *cur_val = idx;
        return 1;
@@ -8220,11 +8696,16 @@ static void alc268_auto_init(struct hda_codec *codec)
  */
 static const char *alc268_models[ALC268_MODEL_LAST] = {
        [ALC268_3ST]            = "3stack",
+       [ALC268_TOSHIBA]        = "toshiba",
+       [ALC268_ACER]           = "acer",
        [ALC268_AUTO]           = "auto",
 };
 
 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(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA),
+       SND_PCI_QUIRK(0x1025, 0x0126, "Acer", ALC268_ACER),
        {}
 };
 
@@ -8242,6 +8723,36 @@ static struct alc_config_preset alc268_presets[] = {
                .channel_mode = alc268_modes,
                .input_mux = &alc268_capture_source,
        },
+       [ALC268_TOSHIBA] = {
+               .mixers = { alc268_base_mixer, alc268_capture_alt_mixer },
+               .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+                               alc268_toshiba_verbs },
+               .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+               .dac_nids = alc268_dac_nids,
+               .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+               .adc_nids = alc268_adc_nids_alt,
+               .hp_nid = 0x03,
+               .num_channel_mode = ARRAY_SIZE(alc268_modes),
+               .channel_mode = alc268_modes,
+               .input_mux = &alc268_capture_source,
+               .input_mux = &alc268_capture_source,
+               .unsol_event = alc268_toshiba_unsol_event,
+               .init_hook = alc268_toshiba_automute,
+       },
+       [ALC268_ACER] = {
+               .mixers = { alc268_acer_mixer, alc268_capture_alt_mixer },
+               .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
+                               alc268_acer_verbs },
+               .num_dacs = ARRAY_SIZE(alc268_dac_nids),
+               .dac_nids = alc268_dac_nids,
+               .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt),
+               .adc_nids = alc268_adc_nids_alt,
+               .hp_nid = 0x02,
+               .num_channel_mode = ARRAY_SIZE(alc268_modes),
+               .channel_mode = alc268_modes,
+               .input_mux = &alc268_capture_source,
+               .unsol_event = alc268_acer_unsol_event,
+       },
 };
 
 static int patch_alc268(struct hda_codec *codec)
@@ -8948,14 +9459,10 @@ static void alc861_toshiba_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x0f, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       snd_hda_codec_amp_update(codec, 0x16, 0, HDA_INPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x16, 1, HDA_INPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_INPUT, 3,
-                                0x80, present ? 0 : 0x80);
-       snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_INPUT, 3,
-                                0x80, present ? 0 : 0x80);
+       snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+       snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3,
+                                HDA_AMP_MUTE, present ? 0 : HDA_AMP_MUTE);
 }
 
 static void alc861_toshiba_unsol_event(struct hda_codec *codec,
@@ -9268,6 +9775,16 @@ static void alc861_auto_init(struct hda_codec *codec)
        alc861_auto_init_analog_input(codec);
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static struct hda_amp_list alc861_loopbacks[] = {
+       { 0x15, HDA_INPUT, 0 },
+       { 0x15, HDA_INPUT, 1 },
+       { 0x15, HDA_INPUT, 2 },
+       { 0x15, HDA_INPUT, 3 },
+       { } /* end */
+};
+#endif
+
 
 /*
  * configuration and preset
@@ -9289,14 +9806,21 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP),
        SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP),
        SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP),
+       SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP),
        SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS),
+       SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG),
        SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA),
-       SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA),
+       /* FIXME: the entry below breaks Toshiba A100 (model=auto works!)
+        *        Any other models that need this preset?
+        */
+       /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */
        SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31),
        SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31),
        SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31),
        SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST),
        SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST),
+       SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST),
+       SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST),
        {}
 };
 
@@ -9458,6 +9982,10 @@ static int patch_alc861(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC861_AUTO)
                spec->init_hook = alc861_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc861_loopbacks;
+#endif
                
        return 0;
 }
@@ -9514,6 +10042,14 @@ static struct hda_input_mux alc861vd_dallas_capture_source = {
        },
 };
 
+static struct hda_input_mux alc861vd_hp_capture_source = {
+       .num_items = 2,
+       .items = {
+               { "Front Mic", 0x0 },
+               { "ATAPI Mic", 0x1 },
+       },
+};
+
 #define alc861vd_mux_enum_info alc_mux_enum_info
 #define alc861vd_mux_enum_get alc_mux_enum_get
 
@@ -9532,12 +10068,13 @@ static int alc861vd_mux_enum_put(struct snd_kcontrol *kcontrol,
        idx = ucontrol->value.enumerated.item[0];
        if (idx >= imux->num_items)
                idx = imux->num_items - 1;
-       if (*cur_val == idx && !codec->in_resume)
+       if (*cur_val == idx)
                return 0;
        for (i = 0; i < imux->num_items; i++) {
-               unsigned int v = (i == idx) ? 0x7000 : 0x7080;
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                   v | (imux->items[i].index << 8));
+               unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+               snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+                                        imux->items[i].index,
+                                        HDA_AMP_MUTE, v);
        }
        *cur_val = idx;
        return 1;
@@ -9711,17 +10248,22 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
        HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x05, HDA_INPUT),
        HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x05, HDA_INPUT),
-       HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT),
-       HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT),
-       {
-               .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-               /* .name = "Capture Source", */
-               .name = "Input Source",
-               .count = 1,
-               .info = alc882_mux_enum_info,
-               .get = alc882_mux_enum_get,
-               .put = alc882_mux_enum_put,
-       },
+       { } /* end */
+};
+
+/* Pin assignment: Speaker=0x14, Line-out = 0x15,
+ *                 Front Mic=0x18, ATAPI Mic = 0x19,
+ */
+static struct snd_kcontrol_new alc861vd_hp_mixer[] = {
+       HDA_CODEC_VOLUME("Front Playback Volume", 0x02, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
+       HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+       HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+       HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+       HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+       
        { } /* end */
 };
 
@@ -9739,11 +10281,11 @@ static struct hda_verb alc861vd_volume_init_verbs[] = {
         * the analog-loopback mixer widget
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /* Capture mixer: unmute Mic, F-Mic, Line, CD inputs */
        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -9872,11 +10414,9 @@ static void alc861vd_lenovo_hp_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x1b, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
@@ -9886,11 +10426,9 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x18, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x0b, 0, HDA_INPUT, 1,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x0b, 1, HDA_INPUT, 1,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc861vd_lenovo_automute(struct hda_codec *codec)
@@ -9964,10 +10502,8 @@ static void alc861vd_dallas_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x15, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
-       snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                0x80, present ? 0x80 : 0);
+       snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
 
 static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -9976,6 +10512,10 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re
                alc861vd_dallas_automute(codec);
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc861vd_loopbacks     alc880_loopbacks
+#endif
+
 /* pcm configuration: identiacal with ALC880 */
 #define alc861vd_pcm_analog_playback   alc880_pcm_analog_playback
 #define alc861vd_pcm_analog_capture    alc880_pcm_analog_capture
@@ -9987,18 +10527,20 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re
  */
 static const char *alc861vd_models[ALC861VD_MODEL_LAST] = {
        [ALC660VD_3ST]          = "3stack-660",
+       [ALC660VD_3ST_DIG]      = "3stack-660-digout",
        [ALC861VD_3ST]          = "3stack",
        [ALC861VD_3ST_DIG]      = "3stack-digout",
        [ALC861VD_6ST_DIG]      = "6stack-digout",
        [ALC861VD_LENOVO]       = "lenovo",
        [ALC861VD_DALLAS]       = "dallas",
+       [ALC861VD_HP]           = "hp",
        [ALC861VD_AUTO]         = "auto",
 };
 
 static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST),
        SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),
-       SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST),
+       SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG),
        SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST),
        SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST),
 
@@ -10006,6 +10548,10 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
        SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS),
        SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
        SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
+       SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO),
+       SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
+       SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
+       SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP),
        {}
 };
 
@@ -10022,6 +10568,19 @@ static struct alc_config_preset alc861vd_presets[] = {
                .channel_mode = alc861vd_3stack_2ch_modes,
                .input_mux = &alc861vd_capture_source,
        },
+       [ALC660VD_3ST_DIG] = {
+               .mixers = { alc861vd_3st_mixer },
+               .init_verbs = { alc861vd_volume_init_verbs,
+                                alc861vd_3stack_init_verbs },
+               .num_dacs = ARRAY_SIZE(alc660vd_dac_nids),
+               .dac_nids = alc660vd_dac_nids,
+               .dig_out_nid = ALC861VD_DIGOUT_NID,
+               .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
+               .adc_nids = alc861vd_adc_nids,
+               .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+               .channel_mode = alc861vd_3stack_2ch_modes,
+               .input_mux = &alc861vd_capture_source,
+       },
        [ALC861VD_3ST] = {
                .mixers = { alc861vd_3st_mixer },
                .init_verbs = { alc861vd_volume_init_verbs,
@@ -10082,7 +10641,21 @@ static struct alc_config_preset alc861vd_presets[] = {
                .input_mux = &alc861vd_dallas_capture_source,
                .unsol_event = alc861vd_dallas_unsol_event,
                .init_hook = alc861vd_dallas_automute,
-       },      
+       },
+       [ALC861VD_HP] = {
+               .mixers = { alc861vd_hp_mixer },
+               .init_verbs = { alc861vd_dallas_verbs, alc861vd_eapd_verbs },
+               .num_dacs = ARRAY_SIZE(alc861vd_dac_nids),
+               .dac_nids = alc861vd_dac_nids,
+               .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids),
+               .dig_out_nid = ALC861VD_DIGOUT_NID,
+               .adc_nids = alc861vd_adc_nids,
+               .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes),
+               .channel_mode = alc861vd_3stack_2ch_modes,
+               .input_mux = &alc861vd_hp_capture_source,
+               .unsol_event = alc861vd_dallas_unsol_event,
+               .init_hook = alc861vd_dallas_automute,
+       },              
 };
 
 /*
@@ -10382,6 +10955,10 @@ static int patch_alc861vd(struct hda_codec *codec)
 
        if (board_config == ALC861VD_AUTO)
                spec->init_hook = alc861vd_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc861vd_loopbacks;
+#endif
 
        return 0;
 }
@@ -10439,7 +11016,7 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol,
        struct alc_spec *spec = codec->spec;
        const struct hda_input_mux *imux = spec->input_mux;
        unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-       static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
+       static hda_nid_t capture_mixers[2] = { 0x23, 0x22 };
        hda_nid_t nid = capture_mixers[adc_idx];
        unsigned int *cur_val = &spec->cur_mux[adc_idx];
        unsigned int i, idx;
@@ -10447,12 +11024,13 @@ static int alc662_mux_enum_put(struct snd_kcontrol *kcontrol,
        idx = ucontrol->value.enumerated.item[0];
        if (idx >= imux->num_items)
                idx = imux->num_items - 1;
-       if (*cur_val == idx && !codec->in_resume)
+       if (*cur_val == idx)
                return 0;
        for (i = 0; i < imux->num_items; i++) {
-               unsigned int v = (i == idx) ? 0x7000 : 0x7080;
-               snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                   v | (imux->items[i].index << 8));
+               unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE;
+               snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT,
+                                        imux->items[i].index,
+                                        HDA_AMP_MUTE, v);
        }
        *cur_val = idx;
        return 1;
@@ -10661,11 +11239,11 @@ static struct hda_verb alc662_init_verbs[] = {
        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
        /* Front mixer: unmute input/output amp left and right (volume = 0) */
 
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
        {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -10734,11 +11312,11 @@ static struct hda_verb alc662_auto_init_verbs[] = {
         * panel mic (mic 2)
         */
        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
-       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+       {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 
        /*
         * Set up output mixers (0x0c - 0x0f)
@@ -10762,11 +11340,7 @@ static struct hda_verb alc662_auto_init_verbs[] = {
        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
        /* Input mixer */
        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-       {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-       {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-       /*{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},*/
-       {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
+       {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
        { }
 };
 
@@ -10797,11 +11371,9 @@ static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x14, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
@@ -10811,15 +11383,11 @@ static void alc662_lenovo_101e_all_automute(struct hda_codec *codec)
 
        present = snd_hda_codec_read(codec, 0x1b, 0,
                                     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-       bits = present ? 0x80 : 0;
-       snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0,
-                                0x80, bits);
-       snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0,
-                                0x80, bits);
+       bits = present ? HDA_AMP_MUTE : 0;
+       snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
+       snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+                                HDA_AMP_MUTE, bits);
 }
 
 static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
@@ -10831,6 +11399,10 @@ static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
                alc662_lenovo_101e_ispeaker_automute(codec);
 }
 
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+#define alc662_loopbacks       alc880_loopbacks
+#endif
+
 
 /* pcm configuration: identiacal with ALC880 */
 #define alc662_pcm_analog_playback     alc880_pcm_analog_playback
@@ -10943,7 +11515,7 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
        for (i = 0; i < cfg->line_outs; i++) {
                if (!spec->multiout.dac_nids[i])
                        continue;
-               nid = alc880_idx_to_dac(i);
+               nid = alc880_idx_to_mixer(i);
                if (i == 2) {
                        /* Center/LFE */
                        err = add_control(spec, ALC_CTL_WIDGET_VOL,
@@ -11166,14 +11738,10 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
        spec->num_mux_defs = 1;
        spec->input_mux = &spec->private_imux;
        
-       if (err < 0)
-               return err;
-       else if (err > 0)
-               /* hack - override the init verbs */
-               spec->init_verbs[0] = alc662_auto_init_verbs;
+       spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs;
        spec->mixers[spec->num_mixers] = alc662_capture_mixer;
        spec->num_mixers++;
-       return err;
+       return 1;
 }
 
 /* additional initialization for auto-configuration model */
@@ -11210,7 +11778,7 @@ static int patch_alc662(struct hda_codec *codec)
                if (err < 0) {
                        alc_free(codec);
                        return err;
-               } else if (err) {
+               } else if (!err) {
                        printk(KERN_INFO
                               "hda_codec: Cannot set up configuration "
                               "from BIOS.  Using base mode...\n");
@@ -11237,6 +11805,10 @@ static int patch_alc662(struct hda_codec *codec)
        codec->patch_ops = alc_patch_ops;
        if (board_config == ALC662_AUTO)
                spec->init_hook = alc662_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+       if (!spec->loopback.amplist)
+               spec->loopback.amplist = alc662_loopbacks;
+#endif
 
        return 0;
 }