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-rw-r--r--Documentation/sound/alsa/HD-Audio.txt16
-rw-r--r--sound/pci/hda/hda_intel.c1
-rw-r--r--sound/pci/hda/patch_analog.c8
-rw-r--r--sound/pci/hda/patch_realtek.c164
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);
}
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-02 08:58:51 +0000