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Merge branch 'fix/hda' into for-linus

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This commit is contained in:
Takashi Iwai 2010-04-07 09:54:46 +02:00
commit 55b371d4ac
4 changed files with 138 additions and 51 deletions
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16
Documentation/sound/alsa/HD-Audio.txt
View File

@ -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

1
sound/pci/hda/hda_intel.c
View File

@ -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 */
{}
};

8
sound/pci/hda/patch_analog.c
View File

@ -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;

164
sound/pci/hda/patch_realtek.c
View File

@ -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);
}
}