From a389d67cf9849aff1722ed73186a584e2196a873 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 27 Jan 2012 14:31:19 +0100 Subject: ALSA: HDA: Remove quirk for Asus N53Jq The user reports that he needs to add model=auto for audio to work properly. In fact, since node 0x15 is not even a pin node, the existing fixup is definitely wrong. Relevant information can be found in the buglink below. Cc: stable@kernel.org (3.2+) BugLink: https://bugs.launchpad.net/bugs/918254 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0db1dc49382b..a7f17becbd7c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5377,7 +5377,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC), -- cgit v1.2.3 From 31150f2327cbb66363f38e13ca1be973d2f9203a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 30 Jan 2012 10:54:08 +0100 Subject: ALSA: hda - Apply 0x0f-VREF fix to all ASUS laptops with ALC861/660 It turned out that other ASUS laptops require the similar fix to enable the VREF on the pin 0x0f for the secret output amp, not only ASUS A6Rp. Moreover, it's required even when the pin is being used as the output. Thus, writing a fixed value doesn't work always. This patch applies the VREF-fix for all ASUS laptops with ALC861/660 in a fixup function that checks the current value and turns on only the VREF value no matter whether input or output direction is set. The automute function is modified as well to keep the pin VREF upon muting/unmuting via pin-control; otherwise the pin VREF is reset at plugging/unplugging a jack. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 35 insertions(+), 8 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7f17becbd7c..42b6a01e17db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -177,6 +177,7 @@ struct alc_spec { unsigned int detect_lo:1; /* Line-out detection enabled */ unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ unsigned int automute_lo_possible:1; /* there are line outs and HP */ + unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -495,13 +496,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; + unsigned int val; if (!nid) break; switch (spec->automute_mode) { case ALC_AUTOMUTE_PIN: + /* don't reset VREF value in case it's controlling + * the amp (see alc861_fixup_asus_amp_vref_0f()) + */ + if (spec->keep_vref_in_automute) { + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val &= ~PIN_HP; + } else + val = 0; + val |= pin_bits; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_bits); + val); break; case ALC_AUTOMUTE_AMP: snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, @@ -5588,6 +5600,25 @@ enum { PINFIX_ASUS_A6RP, }; +/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ +static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + if (action != ALC_FIXUP_ACT_INIT) + return; + val = snd_hda_codec_read(codec, 0x0f, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))) + val |= AC_PINCTL_IN_EN; + val |= AC_PINCTL_VREF_50; + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + spec->keep_vref_in_automute = 1; +} + static const struct alc_fixup alc861_fixups[] = { [PINFIX_FSC_AMILO_PI1505] = { .type = ALC_FIXUP_PINS, @@ -5598,17 +5629,13 @@ static const struct alc_fixup alc861_fixups[] = { } }, [PINFIX_ASUS_A6RP] = { - .type = ALC_FIXUP_VERBS, - .v.verbs = (const struct hda_verb[]) { - /* node 0x0f VREF seems controlling the master output */ - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50 }, - { } - }, + .type = ALC_FIXUP_FUNC, + .v.func = alc861_fixup_asus_amp_vref_0f, }, }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} -- cgit v1.2.3 From 05c3b36e539627b7aed67d038381d0d9fa9d61e7 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 31 Jan 2012 09:04:15 +0100 Subject: ALSA: HDA: Fix jack creation for codecs with front and rear Line In If a codec has both a front and a rear Line In, two controls both named "Line Jack" will be created, which causes parsing to fail. While a long term solution might be to name the jacks differently, this extra check is consistent with what is already being done in many auto-parsers, and will also protect against other cases when two inputs have the same label. BugLink: https://bugs.launchpad.net/bugs/923409 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d8a35da0803f..9d819c4b4923 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) + const struct auto_pin_cfg *cfg, + char *lastname, int *lastidx) { unsigned int def_conf, conn; char name[44]; @@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (!strcmp(name, lastname) && idx == *lastidx) + idx++; + strncpy(lastname, name, 44); + *lastidx = idx; err = snd_hda_jack_add_kctl(codec, nid, name, idx); if (err < 0) return err; @@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err; + int i, err, lastidx = 0; + char lastname[44] = ""; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); if (err < 0) return err; return 0; -- cgit v1.2.3 From 3422a47041b8cb8f14ac1e3926bcf711121df6dc Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 31 Jan 2012 10:31:49 +0100 Subject: ALSA: HDA: Remove quirk for Toshiba Qosmio G50 The user reports that model=auto works better than current handling on a 3.2 based kernel (with jack detection patches backported). Since model=auto is what we prefer these days anyway, the quirk should be removed. Alsa-info for the relevant machine: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/923316/+attachment/2702812/+files/alsa-info.txt.Pbfno2x7bp BugLink: https://bugs.launchpad.net/bugs/923316 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 ------------- 1 file changed, 13 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42b6a01e17db..a8e82be3d2fc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4747,7 +4747,6 @@ enum { ALC262_FIXUP_FSC_H270, ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, - ALC262_FIXUP_TOSHIBA_RX1, ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, @@ -4777,16 +4776,6 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [ALC262_FIXUP_TOSHIBA_RX1] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x90170110 }, /* speaker */ - { 0x15, 0x0421101f }, /* HP */ - { 0x1a, 0x40f000f0 }, /* N/A */ - { 0x1b, 0x40f000f0 }, /* N/A */ - { 0x1e, 0x40f000f0 }, /* N/A */ - } - }, [ALC262_FIXUP_LENOVO_3000] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -4819,8 +4808,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), -- cgit v1.2.3 From f70eecde3bca92630d3886496e73316ff353f185 Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Tue, 31 Jan 2012 13:04:41 -0800 Subject: ALSA: hda - Fix calling cs_automic twice for Cirrus codecs. If cs_automic is called twice (like it is during init) while the mic is present, it will over-write the last_input with the new one, causing it to switch back to the automic input when the mic is unplugged. This leaves the driver in a state (cur_input, last_input, and automix_idx the same) where the internal mic can not be selected until it is rebooted without the mic attached. Check that the mic hasn't already been switched to before setting last_input. Signed-off-by: Dylan Reid Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e99357e822c..bc5a993d1146 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec) change_cur_input(codec, !spec->automic_idx, 0); } else { if (present) { - spec->last_input = spec->cur_input; - spec->cur_input = spec->automic_idx; + if (spec->cur_input != spec->automic_idx) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } } else { spec->cur_input = spec->last_input; } -- cgit v1.2.3 From 54c2a89f60fd71b924d0f848ac892442951401a6 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 1 Feb 2012 12:05:41 +0100 Subject: ALSA: HDA: Fix duplicated output to more than one codec This typo caused the wrong codec's nid to be checked for wcaps type. As a result, sometimes speakers would duplicate the output sent to HDMI output. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/924320 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4df72c0e8c37..c2c65f63bf06 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); if (!p->active && p->stream_tag == stream_tag && - get_wcaps_type(get_wcaps(codec, p->nid)) == type) + get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; } } -- cgit v1.2.3 From 054d867e032daf55c3343fc6d36c5c5f1e7954db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jan 2012 12:25:50 +0100 Subject: ALSA: hda - Check power-state before changing in patch_via.c Instead of always writing AC_VERB_SET_POWER_STATE, check the current power-state and don't write again if the value is already set. This may reduce the click noise upon the dynamic power-state change (e.g. in analog-input mixer). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 256 +++++++++++++++++++--------------------------- 1 file changed, 107 insertions(+), 149 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 03e63fed9caf..fb1f0ffc556b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -687,6 +687,15 @@ static void via_auto_init_analog_input(struct hda_codec *codec) } } +static void update_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int parm) +{ + if (snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0) == parm) + return; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, unsigned int *affected_parm) { @@ -709,7 +718,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, } else parm = AC_PWRST_D3; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, nid, parm); } static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol, @@ -2295,10 +2304,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, if (mux) { /* switch to D0 beofre change index */ - if (snd_hda_codec_read(codec, mux, 0, - AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) - snd_hda_codec_write(codec, mux, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, mux, AC_PWRST_D0); snd_hda_codec_write(codec, mux, 0, AC_VERB_SET_CONNECT_SEL, spec->inputs[cur].mux_idx); @@ -2922,9 +2928,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (17h), AIW 0/1 (13h/14h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x13, parm); + update_power_state(codec, 0x14, parm); /* outputs */ /* PW0 (19h), SW1 (18h), AOW1 (11h) */ @@ -2932,8 +2938,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x19, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x11, parm); /* PW6 (22h), SW2 (26h), AOW2 (24h) */ if (is_8ch) { @@ -2941,20 +2947,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x22, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x26, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x24, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x26, parm); + update_power_state(codec, 0x24, parm); } else if (codec->vendor_id == 0x11064397) { /* PW7(23h), SW2(27h), AOW2(25h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x23, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x27, parm); + update_power_state(codec, 0x25, parm); } /* PW 3/4/7 (1ch/1dh/23h) */ @@ -2966,17 +2968,13 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec) set_pin_power_state(codec, 0x23, &parm); /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, parm); if (is_8ch) { - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x27, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); + update_power_state(codec, 0x27, parm); } else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); } static int patch_vt1708S(struct hda_codec *codec); @@ -3149,10 +3147,10 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x13, parm); + update_power_state(codec, 0x12, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x20, parm); /* outputs */ /* PW 3/4 (16h/17h) */ @@ -3160,10 +3158,9 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec) set_pin_power_state(codec, 0x17, &parm); set_pin_power_state(codec, 0x16, &parm); /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x1d, parm); } static int patch_vt1702(struct hda_codec *codec) @@ -3228,52 +3225,48 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x27, &parm); - snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1a, parm); + update_power_state(codec, 0xb, parm); /* PW2 (26h), AOW2 (ah) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x26, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x2b, &parm); - snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0xa, parm); /* PW0 (24h), AOW0 (8h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (!spec->hp_independent_mode) /* check for redirected HP */ set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x8, parm); /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); /* PW1 (25h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); if (spec->smart51_enabled) set_pin_power_state(codec, 0x2a, &parm); - snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x9, parm); if (spec->hp_independent_mode) { /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x1b, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0xc, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1b, parm); + update_power_state(codec, 0x34, parm); + update_power_state(codec, 0xc, parm); } } @@ -3433,8 +3426,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) if (imux_is_smixer) parm = AC_PWRST_D0; /* SW0 (17h), AIW0(13h) */ - snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x13, parm); parm = AC_PWRST_D3; set_pin_power_state(codec, 0x1e, &parm); @@ -3442,12 +3435,11 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) if (spec->dmic_enabled) set_pin_power_state(codec, 0x22, &parm); else - snd_hda_codec_write(codec, 0x22, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x22, AC_PWRST_D3); /* SW2(26h), AIW1(14h) */ - snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x26, parm); + update_power_state(codec, 0x14, parm); /* outputs */ /* PW0 (19h), SW1 (18h), AOW1 (11h) */ @@ -3456,8 +3448,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) /* Smart 5.1 PW2(1bh) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1b, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x11, parm); /* PW7 (23h), SW3 (27h), AOW3 (25h) */ parm = AC_PWRST_D3; @@ -3465,12 +3457,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) /* Smart 5.1 PW1(1ah) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1a, &parm); - snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x27, parm); /* Smart 5.1 PW5(1eh) */ if (spec->smart51_enabled) set_pin_power_state(codec, 0x1e, &parm); - snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); /* Mono out */ /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ @@ -3486,9 +3478,9 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) mono_out = 1; } parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x28, parm); + update_power_state(codec, 0x29, parm); + update_power_state(codec, 0x2a, parm); /* PW 3/4 (1ch/1dh) */ parm = AC_PWRST_D3; @@ -3496,15 +3488,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec) set_pin_power_state(codec, 0x1d, &parm); /* HP Independent Mode, power on AOW3 */ if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x25, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x25, parm); /* force to D0 for internal Speaker */ /* MW0 (16h), AOW0 (10h) */ - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, - imux_is_smixer ? AC_PWRST_D0 : parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, - mono_out ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); + update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm); } static int patch_vt1716S(struct hda_codec *codec) @@ -3580,54 +3569,45 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) set_pin_power_state(codec, 0x2b, &parm); parm = AC_PWRST_D0; /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x8, parm); if (spec->codec_type == VT1802) { /* PW4 (28h), MW4 (18h), MUX4(38h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x38, parm); } else { /* PW4 (26h), MW4 (1ch), MUX4(37h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x26, &parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x37, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1c, parm); + update_power_state(codec, 0x37, parm); } if (spec->codec_type == VT1802) { /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x15, parm); + update_power_state(codec, 0x35, parm); } else { /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x19, parm); + update_power_state(codec, 0x35, parm); } if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x9, AC_PWRST_D0); /* Class-D */ /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ @@ -3637,12 +3617,10 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) set_pin_power_state(codec, 0x24, &parm); parm = present ? AC_PWRST_D3 : AC_PWRST_D0; if (spec->codec_type == VT1802) - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x14, parm); else - snd_hda_codec_write(codec, 0x18, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x34, parm); /* Mono Out */ present = snd_hda_jack_detect(codec, 0x26); @@ -3650,28 +3628,20 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec) parm = present ? AC_PWRST_D3 : AC_PWRST_D0; if (spec->codec_type == VT1802) { /* PW15 (33h), MW8(1ch), MUX8(3ch) */ - snd_hda_codec_write(codec, 0x33, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x33, parm); + update_power_state(codec, 0x1c, parm); + update_power_state(codec, 0x3c, parm); } else { /* PW15 (31h), MW8(17h), MUX8(3bh) */ - snd_hda_codec_write(codec, 0x31, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x17, 0, - AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3b, 0, - AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x31, parm); + update_power_state(codec, 0x17, parm); + update_power_state(codec, 0x3b, parm); } /* MW9 (21h) */ if (imux_is_smixer || !is_aa_path_mute(codec)) - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x21, AC_PWRST_D0); else - snd_hda_codec_write(codec, 0x21, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x21, AC_PWRST_D3); } /* patch for vt2002P */ @@ -3731,30 +3701,28 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) set_pin_power_state(codec, 0x2b, &parm); parm = AC_PWRST_D0; /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1e, parm); + update_power_state(codec, 0x1f, parm); + update_power_state(codec, 0x10, parm); + update_power_state(codec, 0x11, parm); /* outputs */ /* AOW0 (8h)*/ - snd_hda_codec_write(codec, 0x8, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x8, AC_PWRST_D0); /* PW4 (28h), MW4 (18h), MUX4(38h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x28, &parm); - snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x18, parm); + update_power_state(codec, 0x38, parm); /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x25, &parm); - snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x15, parm); + update_power_state(codec, 0x35, parm); if (spec->hp_independent_mode) - snd_hda_codec_write(codec, 0x9, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x9, AC_PWRST_D0); /* Internal Speaker */ /* PW0 (24h), MW0(14h), MUX0(34h) */ @@ -3763,15 +3731,11 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x24, &parm); if (present) { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x14, AC_PWRST_D3); + update_power_state(codec, 0x34, AC_PWRST_D3); } else { - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x34, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x14, AC_PWRST_D0); + update_power_state(codec, 0x34, AC_PWRST_D0); } @@ -3782,26 +3746,20 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec) parm = AC_PWRST_D3; set_pin_power_state(codec, 0x31, &parm); if (present) { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + update_power_state(codec, 0x1c, AC_PWRST_D3); + update_power_state(codec, 0x3c, AC_PWRST_D3); + update_power_state(codec, 0x3e, AC_PWRST_D3); } else { - snd_hda_codec_write(codec, 0x1c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3c, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - snd_hda_codec_write(codec, 0x3e, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + update_power_state(codec, 0x1c, AC_PWRST_D0); + update_power_state(codec, 0x3c, AC_PWRST_D0); + update_power_state(codec, 0x3e, AC_PWRST_D0); } /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ parm = AC_PWRST_D3; set_pin_power_state(codec, 0x33, &parm); - snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm); - snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm); + update_power_state(codec, 0x1d, parm); + update_power_state(codec, 0x3d, parm); } -- cgit v1.2.3 From 924339239fd5ba3e505f9420d41f0939196f3530 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Jan 2012 13:58:36 +0100 Subject: ALSA: hda - Fix the logic to detect VIA analog low-current mode The analog low-current mode must be enabled when the no stream is running but the current detection checks it in a wrong way. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index fb1f0ffc556b..de43cd92b0a5 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1051,7 +1051,7 @@ static void analog_low_current_mode(struct hda_codec *codec) bool enable; unsigned int verb, parm; - enable = is_aa_path_mute(codec) && (spec->opened_streams != 0); + enable = is_aa_path_mute(codec) && !spec->opened_streams; /* decide low current mode's verb & parameter */ switch (spec->codec_type) { -- cgit v1.2.3 From e9d010c2e8f03952e67a6fd8aed0f0dc92084ccc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Feb 2012 10:33:23 +0100 Subject: ALSA: hda - Allow analog low-current mode when dynamic power-control is on VIA codecs have several different power-saving features, and one of them is the analog low-current mode. But it turned out that the ALC mode causes pop-noises at each on/off time on some machines. As a quick workaround, disable the ALC when another power-saving feature, the dynamic pin power-control, is turned off, too, since the dynamic power-control is already exposed as a mixer enum element so that user can turn it on/off freely. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de43cd92b0a5..79166fb8b074 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -199,6 +199,9 @@ struct via_spec { unsigned int no_pin_power_ctl; enum VIA_HDA_CODEC codec_type; + /* analog low-power control */ + bool alc_mode; + /* smart51 setup */ unsigned int smart51_nums; hda_nid_t smart51_pins[2]; @@ -758,6 +761,7 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, return 0; spec->no_pin_power_ctl = val; set_widgets_power_state(codec); + analog_low_current_mode(codec); return 1; } @@ -1045,13 +1049,19 @@ static bool is_aa_path_mute(struct hda_codec *codec) } /* enter/exit analog low-current mode */ -static void analog_low_current_mode(struct hda_codec *codec) +static void __analog_low_current_mode(struct hda_codec *codec, bool force) { struct via_spec *spec = codec->spec; bool enable; unsigned int verb, parm; - enable = is_aa_path_mute(codec) && !spec->opened_streams; + if (spec->no_pin_power_ctl) + enable = false; + else + enable = is_aa_path_mute(codec) && !spec->opened_streams; + if (enable == spec->alc_mode && !force) + return; + spec->alc_mode = enable; /* decide low current mode's verb & parameter */ switch (spec->codec_type) { @@ -1083,6 +1093,11 @@ static void analog_low_current_mode(struct hda_codec *codec) snd_hda_codec_write(codec, codec->afg, 0, verb, parm); } +static void analog_low_current_mode(struct hda_codec *codec) +{ + return __analog_low_current_mode(codec, false); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -1508,10 +1523,6 @@ static int via_build_controls(struct hda_codec *codec) return err; } - /* init power states */ - set_widgets_power_state(codec); - analog_low_current_mode(codec); - via_free_kctls(codec); /* no longer needed */ err = snd_hda_jack_add_kctls(codec, &spec->autocfg); @@ -2782,6 +2793,10 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + /* init power states */ + set_widgets_power_state(codec); + __analog_low_current_mode(codec, true); + via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_speaker_out(codec); -- cgit v1.2.3 From b5bcc189401c815988b7dd37611fc56f40c9139d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Feb 2012 10:30:17 +0100 Subject: ALSA: hda - Disable dynamic-power control for VIA as default Since the dynamic pin power-control and the analog low-current mode may lead to pop-noise, it's safer to set it off as default. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128 Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 79166fb8b074..284e311040fe 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1470,6 +1470,7 @@ static int via_build_controls(struct hda_codec *codec) struct snd_kcontrol *kctl; int err, i; + spec->no_pin_power_ctl = 1; if (spec->set_widgets_power_state) if (!via_clone_control(spec, &via_pin_power_ctl_enum)) return -ENOMEM; -- cgit v1.2.3 From b544d1e0e233f83a2e6d20ee96b54ea272d5d5ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Feb 2012 11:56:35 +0100 Subject: ALSA: hda/realtek - Add missing Bass and CLFE as vmaster slaves The recent changes in Realtek auto-parser added the new "Bass Speaker" and "CLFE" mixer elements which should be tracked as vmaster slaves, too. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42720 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8e82be3d2fc..33b6077fcdb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1855,6 +1855,8 @@ static const char * const alc_slave_vols[] = { "Speaker Playback Volume", "Mono Playback Volume", "Line-Out Playback Volume", + "CLFE Playback Volume", + "Bass Speaker Playback Volume", "PCM Playback Volume", NULL, }; @@ -1870,6 +1872,8 @@ static const char * const alc_slave_sws[] = { "Mono Playback Switch", "IEC958 Playback Switch", "Line-Out Playback Switch", + "CLFE Playback Switch", + "Bass Speaker Playback Switch", "PCM Playback Switch", NULL, }; -- cgit v1.2.3 From eedec3d3854a390fc14008f265930f8c22b0373f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Feb 2012 10:24:04 +0100 Subject: ALSA: hda/realtek - Fix a wrong condition sparse complains that "spec->multiout.dac_nids" is a pointer. sound/pci/hda/patch_realtek.c:2321:37: error: incompatible types for operation (>) sound/pci/hda/patch_realtek.c:2321:37: left side has type unsigned short const [usertype] *dac_nids sound/pci/hda/patch_realtek.c:2321:37: right side has type int It was meant to be num_dacs instead of dac_nids. Although the current code still works as expected (when num_dacs is zero, dac_nids should be NULL, too), better to fix now, of course. Reported-by: Dan Carpenter Cc: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33b6077fcdb8..485a83746e5a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2322,7 +2322,7 @@ static int alc_build_pcms(struct hda_codec *codec) "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - if (spec->multiout.dac_nids > 0) { + if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; if (!p) p = &alc_pcm_analog_playback; -- cgit v1.2.3 From b97f6bfdd1af95681de5a9f652da644a6525e376 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Feb 2012 11:00:53 +0100 Subject: ALSA: hda - Fix error handling in patch_ca0132.c In patch_ca0132.c, the error returned from chipio_write() isn't checked always. Also, the power-up/down sequence isn't tracked properly in some error paths. Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 33 +++++++++++++++++++-------------- 1 file changed, 19 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c62908..21d91d580da8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_hp_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol, @@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_MUTE, &data); if (err < 0) - return err; + goto exit; /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); - chipio_write(codec, REG_CODEC_MUTE, data); + err = chipio_write(codec, REG_CODEC_MUTE, data); if (err < 0) - return err; + goto exit; spec->curr_speaker_switch = *valp; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol, @@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data); if (err < 0) - return err; + goto exit; val = 31 - left_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_L, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_L, data); if (err < 0) - return err; + goto exit; val = 31 - right_vol; data = (data & 0xe0) | val; - chipio_write(codec, REG_CODEC_HP_VOL_R, data); + err = chipio_write(codec, REG_CODEC_HP_VOL_R, data); if (err < 0) - return err; + goto exit; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol; + exit: snd_hda_power_down(codec); - return 1; + return err < 0 ? err : 1; } static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid) @@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; err = add_in_volume(codec, spec->dig_in, "IEC958"); + if (err < 0) + return err; } return 0; } -- cgit v1.2.3 From 416846d2b31fc740ed9d5a5ec116964fb43c4358 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 7 Feb 2012 14:18:14 +0100 Subject: ALSA: hda - add support for Uniwill ECS M31EI notebook This hardware requires same fixup for the node 0x0f like Asus A6Rp. More information: https://bugzilla.redhat.com/show_bug.cgi?id=785417 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 485a83746e5a..9350f3c3bdf8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5627,6 +5627,7 @@ static const struct alc_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), {} -- cgit v1.2.3 From 2492250e4412c6411324c14ab289629360640b0a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sat, 4 Feb 2012 20:56:47 +0100 Subject: ALSA: oxygen, virtuoso: fix exchanged L/R volumes of aux and CD inputs The driver accidentally exchanged the left/right fields for stereo AC'97 mixer registers. This affected only the aux and CD inputs because the line input bypasses the AC'97 codec and the mic input is mono; cards without AC'97 (Xonar DS/DG/HDAV Slim, HG2PCI, HiFier) were not affected. Reported-and-tested-by: Abby Cedar Signed-off-by: Clemens Ladisch Cc: 2.6.31+ Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 25 ++++++++++++++----------- 1 file changed, 14 insertions(+), 11 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 26c7e8bcb229..c0dbb52d45be 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); - value->value.integer.value[0] = 31 - (reg & 0x1f); - if (stereo) - value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + if (!stereo) { + value->value.integer.value[0] = 31 - (reg & 0x1f); + } else { + value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f); + value->value.integer.value[1] = 31 - (reg & 0x1f); + } return 0; } @@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); oldreg = oxygen_read_ac97(chip, codec, index); - newreg = oldreg; - newreg = (newreg & ~0x1f) | - (31 - (value->value.integer.value[0] & 0x1f)); - if (stereo) - newreg = (newreg & ~0x1f00) | - ((31 - (value->value.integer.value[1] & 0x1f)) << 8); - else - newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8); + if (!stereo) { + newreg = oldreg & ~0x1f; + newreg |= 31 - (value->value.integer.value[0] & 0x1f); + } else { + newreg = oldreg & ~0x1f1f; + newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8; + newreg |= 31 - (value->value.integer.value[1] & 0x1f); + } change = newreg != oldreg; if (change) oxygen_write_ac97(chip, codec, index, newreg); -- cgit v1.2.3 From a1e0c3cf7fb07227fe1f26161d969101dba78287 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Feb 2012 09:32:19 +0100 Subject: ALSA: hda - Fix mute-LED VREF value for new HP laptops The new HP laptops turns off the mute LED with VREF50 or VREF80, but not in HIZ unlike the previous models. Since VREF50 (also 80) works with the previous models, let's use VREF50 for all. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 948f0be2f4f3..6345df131a00 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5078,9 +5078,9 @@ static int stac92xx_update_led_status(struct hda_codec *codec) spec->gpio_dir, spec->gpio_data); } else { notmtd_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD; + AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; muted_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ; + AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50; spec->vref_led = muted ? muted_lvl : notmtd_lvl; stac_vrefout_set(codec, spec->vref_mute_led_nid, spec->vref_led); -- cgit v1.2.3 From fc1156c0b0f7ad45ec03d919866349eeca2bf18c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 15:04:06 +0100 Subject: ALSA: hda - Fix initialization of secondary capture source on VT1705 VT1705 codec has two ADCs where the secondary ADC has no MUX but only a fixed connection to the mic pin. This confused the driver and it tries always overriding the input-source selection by assumption of the existing MUX for the secondary ADC, resulted in resetting the input-source at each time PM (including power-saving) occurs. The fix is simply to check the existence of MUX for secondary ADCs in the initialization code. Tested-by: Anisse Astier Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 284e311040fe..dff9a00ee8fb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -666,6 +666,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init input-src */ for (i = 0; i < spec->num_adc_nids; i++) { int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx; + /* secondary ADCs must have the unique MUX */ + if (i > 0 && !spec->mux_nids[i]) + break; if (spec->mux_nids[adc_idx]) { int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx; snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, -- cgit v1.2.3 From 02a237b24d57e2e2d5402c92549e9e792aa24359 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 15:25:07 +0100 Subject: ALSA: hda - Fix silent speaker output on Acer Aspire 6935 Since 3.2 kernel, the driver starts trying to assign the multi-io DACs before the speaker, thus it assigns DAC2/3 for multi-io and DAC4 for the speaker for a standard laptop setup like a HP, a speaker, a mic-in and a line-in. However, on Acer Aspire 6935, it seems that the speaker pin 0x14 must be connected with either DAC1 or 2; otherwise it results in silence by some reason, although the codec itself allows the routing to DAC3/4. As a workaround, the connection list of each pin is reduced to be mapped to either only DAC1/2 or DAC3/4, so that the compatible assignment as in kernel 3.1 is achieved. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1d07e8fa2433..c4bde7108328 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4201,8 +4201,26 @@ enum { PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, PINFIX_ASUS_W90V, + ALC889_FIXUP_DAC_ROUTE, }; +/* Fix the connection of some pins for ALC889: + * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't + * work correctly (bko#42740) + */ +static void alc889_fixup_dac_route(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn1[2] = { 0x0c, 0x0d }; + hda_nid_t conn2[2] = { 0x0e, 0x0f }; + snd_hda_override_conn_list(codec, 0x14, 2, conn1); + snd_hda_override_conn_list(codec, 0x15, 2, conn1); + snd_hda_override_conn_list(codec, 0x18, 2, conn2); + snd_hda_override_conn_list(codec, 0x1a, 2, conn2); + } +} + static const struct alc_fixup alc882_fixups[] = { [PINFIX_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -4239,10 +4257,15 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC889_FIXUP_DAC_ROUTE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_dac_route, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), + SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", PINFIX_ASUS_W90V), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), -- cgit v1.2.3 From 27c3afe6e1cf129faac90405121203962da08ff4 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 13 Feb 2012 23:44:22 -0500 Subject: ALSA: intel8x0: Fix default inaudible sound on Gateway M520 BugLink: https://bugs.launchpad.net/bugs/930842 The reporter states that audio is inaudible by default without muting 'External Amplifier'. Add a quirk to handle his SSID so that changing the control is not necessary. Reported-and-tested-by: Benjamin Carlson Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 9f3b01bb72c8..e0a4263baa20 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2100,6 +2100,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .name = "MSI P4 ATX 645 Ultra", .type = AC97_TUNE_HP_ONLY }, + { + .subvendor = 0x161f, + .subdevice = 0x202f, + .name = "Gateway M520", + .type = AC97_TUNE_INV_EAPD + }, { .subvendor = 0x161f, .subdevice = 0x203a, -- cgit v1.2.3