Merge branch 'fix/hda' into for-linus

diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index f4dd3bf99d126e01e2150b0623f8264baede76f3..98d14cb8a85daf825af4b5e84f3396aa32b6fab6 100644 (file)
--- 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 4bb90675f70fced63e22fe1f9b70283b2094e673..f8fd586ae024616f8fe674add102f954957d9e91 100644 (file)
--- 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 e6d1bdff1b6e56f53778ace479d682eaf385d915..af34606c30c34b8fd854406d476b15faa5635058 100644 (file)
--- 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 9a23444e9e7ac2ee609678c6fb8a8f9a91ee1e1d..c7730dbb9ddbeee2ee802568e88c851e49c8dfb2 100644 (file)
--- 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);
        }
 }