Merge branch 'fix/hda' into for-linus

4 files changed, 138 insertions, 51 deletions

diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index f4dd3bf99d12..98d14cb8a85d 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -119,10 +119,18 @@ the codec slots 0 and 1 no matter what the hardware reports.
 
 Interrupt Handling
 ~~~~~~~~~~~~~~~~~~
-In rare but some cases, the interrupt isn't properly handled as
-default.  You would notice this by the DMA transfer error reported by
-ALSA PCM core, for example.  Using MSI might help in such a case.
-Pass `enable_msi=1` option for enabling MSI.
+HD-audio driver uses MSI as default (if available) since 2.6.33
+kernel as MSI works better on some machines, and in general, it's
+better for performance.  However, Nvidia controllers showed bad
+regressions with MSI (especially in a combination with AMD chipset),
+thus we disabled MSI for them.
+
+There seem also still other devices that don't work with MSI.  If you
+see a regression wrt the sound quality (stuttering, etc) or a lock-up
+in the recent kernel, try to pass `enable_msi=0` option to disable
+MSI.  If it works, you can add the known bad device to the blacklist
+defined in hda_intel.c.  In such a case, please report and give the
+patch back to the upstream developer. 
 
 
 HD-AUDIO CODEC
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4bb90675f70f..f8fd586ae024 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2362,6 +2362,7 @@ static struct snd_pci_quirk msi_black_list[] __devinitdata = {
         SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */
         SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */
         SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */
+        SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */
         {}
 };
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e6d1bdff1b6e..af34606c30c3 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1896,6 +1896,14 @@ static int patch_ad1981(struct hda_codec *codec)
         case AD1981_THINKPAD:
                 spec->mixers[0] = ad1981_thinkpad_mixers;
                 spec->input_mux = &ad1981_thinkpad_capture_source;
+                /* set the upper-limit for mixer amp to 0dB for avoiding the
+                 * possible damage by overloading
+                 */
+                snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
+                                          (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+                                          (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+                                          (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+                                          (1 << AC_AMPCAP_MUTE_SHIFT));
                 break;
         case AD1981_TOSHIBA:
                 spec->mixers[0] = ad1981_hp_mixers;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9a23444e9e7a..c7730dbb9ddb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1621,6 +1621,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
  */
 
 static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Route to built-in subwoofer as well as speakers */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 /* Bias voltage on for external mic port */
         {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
 /* Front Mic: set to PIN_IN (empty by default) */
@@ -1632,10 +1637,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
 /* Enable speaker output */
         {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
         {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
 /* Enable headphone output */
         {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
         {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
         {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
         { }
 };
 
@@ -4984,6 +4991,70 @@ static void set_capture_mixer(struct hda_codec *codec)
         }
 }
 
+/* fill adc_nids (and capsrc_nids) containing all active input pins */
+static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
+                                 int num_nids)
+{
+        struct alc_spec *spec = codec->spec;
+        int n;
+        hda_nid_t fallback_adc = 0, fallback_cap = 0;
+
+        for (n = 0; n < num_nids; n++) {
+                hda_nid_t adc, cap;
+                hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+                int nconns, i, j;
+
+                adc = nids[n];
+                if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN)
+                        continue;
+                cap = adc;
+                nconns = snd_hda_get_connections(codec, cap, conn,
+                                                 ARRAY_SIZE(conn));
+                if (nconns == 1) {
+                        cap = conn[0];
+                        nconns = snd_hda_get_connections(codec, cap, conn,
+                                                         ARRAY_SIZE(conn));
+                }
+                if (nconns <= 0)
+                        continue;
+                if (!fallback_adc) {
+                        fallback_adc = adc;
+                        fallback_cap = cap;
+                }
+                for (i = 0; i < AUTO_PIN_LAST; i++) {
+                        hda_nid_t nid = spec->autocfg.input_pins[i];
+                        if (!nid)
+                                continue;
+                        for (j = 0; j < nconns; j++) {
+                                if (conn[j] == nid)
+                                        break;
+                        }
+                        if (j >= nconns)
+                                break;
+                }
+                if (i >= AUTO_PIN_LAST) {
+                        int num_adcs = spec->num_adc_nids;
+                        spec->private_adc_nids[num_adcs] = adc;
+                        spec->private_capsrc_nids[num_adcs] = cap;
+                        spec->num_adc_nids++;
+                        spec->adc_nids = spec->private_adc_nids;
+                        if (adc != cap)
+                                spec->capsrc_nids = spec->private_capsrc_nids;
+                }
+        }
+        if (!spec->num_adc_nids) {
+                printk(KERN_WARNING "hda_codec: %s: no valid ADC found;"
+                       " using fallback 0x%x\n",
+                       codec->chip_name, fallback_adc);
+                spec->private_adc_nids[0] = fallback_adc;
+                spec->adc_nids = spec->private_adc_nids;
+                if (fallback_adc != fallback_cap) {
+                        spec->private_capsrc_nids[0] = fallback_cap;
+                        spec->capsrc_nids = spec->private_adc_nids;
+                }
+        }
+}
+
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 #define set_beep_amp(spec, nid, idx, dir) \
         ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
@@ -8398,9 +8469,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
 
 static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
         HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
-        HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
         HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
-        HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
@@ -10041,13 +10110,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
         int idx;
 
         alc_set_pin_output(codec, nid, pin_type);
+        if (dac_idx >= spec->multiout.num_dacs)
+                return;
         if (spec->multiout.dac_nids[dac_idx] == 0x25)
                 idx = 4;
-        else {
-                if (spec->multiout.num_dacs >= dac_idx)
-                        return;
+        else
                 idx = spec->multiout.dac_nids[dac_idx] - 2;
-        }
         snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
 
 }
@@ -12459,11 +12527,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
         unsigned char bits;
 
         present = snd_hda_jack_detect(codec, 0x15);
-        bits = present ? AMP_IN_MUTE(0) : 0;
+        bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 }
 
 static void alc268_acer_lc_unsol_event(struct hda_codec *codec,
@@ -13333,9 +13401,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = {
         0x22,
 };
 
-/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24),
- *       not a mux!
- */
+static hda_nid_t alc269_adc_candidates[] = {
+        0x08, 0x09, 0x07,
+};
 
 #define alc269_modes            alc260_modes
 #define alc269_capture_source   alc880_lg_lw_capture_source
@@ -13482,11 +13550,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec)
         unsigned char bits;
 
         present = snd_hda_jack_detect(codec, 0x15);
-        bits = present ? AMP_IN_MUTE(0) : 0;
+        bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                        AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                        AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 
         snd_hda_codec_write(codec, 0x20, 0,
                         AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13511,11 +13579,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
         /* Check port replicator headphone socket */
         present |= snd_hda_jack_detect(codec, 0x1a);
 
-        bits = present ? AMP_IN_MUTE(0) : 0;
+        bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                        AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                        AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 
         snd_hda_codec_write(codec, 0x20, 0,
                         AC_VERB_SET_COEF_INDEX, 0x0c);
@@ -13646,11 +13714,11 @@ static void alc269_speaker_automute(struct hda_codec *codec)
         unsigned char bits;
 
         present = snd_hda_jack_detect(codec, nid);
-        bits = present ? AMP_IN_MUTE(0) : 0;
+        bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 }
 
 /* unsolicited event for HP jack sensing */
@@ -13842,7 +13910,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
         struct alc_spec *spec = codec->spec;
         int err;
         static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
-        hda_nid_t real_capsrc_nids;
 
         err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
                                            alc269_ignore);
@@ -13866,18 +13933,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
         if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) {
                 add_verb(spec, alc269vb_init_verbs);
-                real_capsrc_nids = alc269vb_capsrc_nids[0];
                 alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21);
         } else {
                 add_verb(spec, alc269_init_verbs);
-                real_capsrc_nids = alc269_capsrc_nids[0];
                 alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0);
         }
 
         spec->num_mux_defs = 1;
         spec->input_mux = &spec->private_imux[0];
+        fillup_priv_adc_nids(codec, alc269_adc_candidates,
+                             sizeof(alc269_adc_candidates));
+
         /* set default input source */
-        snd_hda_codec_write_cache(codec, real_capsrc_nids,
+        snd_hda_codec_write_cache(codec, spec->capsrc_nids[0],
                                   0, AC_VERB_SET_CONNECT_SEL,
                                   spec->input_mux->items[0].index);
 
@@ -14156,14 +14224,16 @@ static int patch_alc269(struct hda_codec *codec)
         spec->stream_digital_playback = &alc269_pcm_digital_playback;
         spec->stream_digital_capture = &alc269_pcm_digital_capture;
 
-        if (!is_alc269vb) {
-                spec->adc_nids = alc269_adc_nids;
-                spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
-                spec->capsrc_nids = alc269_capsrc_nids;
-        } else {
-                spec->adc_nids = alc269vb_adc_nids;
-                spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
-                spec->capsrc_nids = alc269vb_capsrc_nids;
+        if (!spec->adc_nids) { /* wasn't filled automatically? use default */
+                if (!is_alc269vb) {
+                        spec->adc_nids = alc269_adc_nids;
+                        spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
+                        spec->capsrc_nids = alc269_capsrc_nids;
+                } else {
+                        spec->adc_nids = alc269vb_adc_nids;
+                        spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids);
+                        spec->capsrc_nids = alc269vb_capsrc_nids;
+                }
         }
 
         if (!spec->cap_mixer)
@@ -17115,9 +17185,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec)
         present = snd_hda_jack_detect(codec, 0x21);
         bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 }
 
 static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
@@ -17128,13 +17198,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec)
         present = snd_hda_jack_detect(codec, 0x21);
         bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 }
 
 static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
@@ -17145,13 +17215,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec)
         present = snd_hda_jack_detect(codec, 0x15);
         bits = present ? HDA_AMP_MUTE : 0;
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
         snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), bits);
+                                 HDA_AMP_MUTE, bits);
 }
 
 static void alc662_f5z_speaker_automute(struct hda_codec *codec)
@@ -17190,14 +17260,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
 
         if (present1 || present2) {
                 snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+                                         HDA_AMP_MUTE, HDA_AMP_MUTE);
                 snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), AMP_IN_MUTE(0));
+                                         HDA_AMP_MUTE, HDA_AMP_MUTE);
         } else {
                 snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0,
-                                AMP_IN_MUTE(0), 0);
+                                         HDA_AMP_MUTE, 0);
                 snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1,
-                                AMP_IN_MUTE(0), 0);
+                                         HDA_AMP_MUTE, 0);
         }
 }