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); } }