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/hda/patch_realtek.c') 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/hda/patch_realtek.c') 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 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/hda/patch_realtek.c') 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 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/hda/patch_realtek.c') 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/hda/patch_realtek.c') 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 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/hda/patch_realtek.c') 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 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/hda/patch_realtek.c') 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 c14c95f62ecb8710af14ae0d48e01991b70bb6f4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 16:38:07 +0100 Subject: ALSA: hda/realtek - Fix overflow of vol/sw check bitmap The bitmap introduced in the commit [527e4d73: ALSA: hda/realtek - Fix missing volume controls with ALC260] is too narrow for some codecs, which may have more NIDs than 0x20, thus it may overflow the bitmap array on them. Just double the number to cover all and also add a sanity-check code to be safer. Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1358987c49d8..389a28a21fa9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -80,6 +80,8 @@ enum { ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; +#define MAX_VOL_NIDS 0x40 + struct alc_spec { /* codec parameterization */ const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ @@ -118,8 +120,8 @@ struct alc_spec { const hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ hda_nid_t mixer_nid; /* analog-mixer NID */ - DECLARE_BITMAP(vol_ctls, 0x20 << 1); - DECLARE_BITMAP(sw_ctls, 0x20 << 1); + DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1); + DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1); /* capture setup for dynamic dual-adc switch */ hda_nid_t cur_adc; @@ -3149,7 +3151,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) static inline unsigned int get_ctl_pos(unsigned int data) { hda_nid_t nid = get_amp_nid_(data); - unsigned int dir = get_amp_direction_(data); + unsigned int dir; + if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) + return 0; + dir = get_amp_direction_(data); return (nid << 1) | dir; } -- cgit v1.2.3 From ef8d60fb79614a86a82720dc2402631dbcafb315 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 10:12:38 +0100 Subject: ALSA: hda/realtek - Fix surround output regression on Acer Aspire 5935 The previous fix for the speaker on Acer Aspire 59135 introduced another problem for surround outputs. It changed the connections on the line-in/mic pins for limiting the routes, but it left the modified connections. Thus wrong connection indices were written when set to 4ch or 6ch mode. This patch fixes it by restoring the right connections just after parsing the tree but before the initialization. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740 Cc: [v3.2+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 389a28a21fa9..3647baa9bfed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4441,12 +4441,20 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { + /* fake the connections during parsing the tree */ 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); + } else if (action == ALC_FIXUP_ACT_PROBE) { + /* restore the connections */ + hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 }; + snd_hda_override_conn_list(codec, 0x14, 5, conn); + snd_hda_override_conn_list(codec, 0x15, 5, conn); + snd_hda_override_conn_list(codec, 0x18, 5, conn); + snd_hda_override_conn_list(codec, 0x1a, 5, conn); } } -- cgit v1.2.3 From 068b939431486f524438330b0848a8222e33d421 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 25 Feb 2012 11:13:16 +0100 Subject: ALSA: hda/realtek - Fix resume of multiple input sources When there are multiple input sources, the driver wrongly overwrites with the value of the last input source on other slots at resume. Thus the primary input source may be shown wrongly. Reported-and-tested-by: Julian Sikorski Cc: [v3.1+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3647baa9bfed..4fe2d5960a04 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3797,7 +3797,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec) else nums = spec->num_adc_nids; for (c = 0; c < nums; c++) - alc_mux_select(codec, 0, spec->cur_mux[c], true); + alc_mux_select(codec, c, spec->cur_mux[c], true); } /* add mic boosts if needed */ -- cgit v1.2.3 From e49a3434f1bc64dc49ff3a56e416bb5894868dde Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Mar 2012 18:14:41 +0100 Subject: ALSA: hda - Kill hyphenated names Kill hyphens from "Line-Out" name strings, as suggested by Mark Brown. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_cirrus.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 6 +++--- 4 files changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/pci/hda/patch_realtek.c') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0ae6eb20b13b..684307372d73 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5118,7 +5118,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid, const char *pfx = "", *sfx = ""; /* handle as a speaker if it's a fixed line-out */ - if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT) + if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT) name = "Speaker"; /* check the location */ switch (attr) { @@ -5177,7 +5177,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid, switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: - return fill_audio_out_name(codec, nid, cfg, "Line-Out", + return fill_audio_out_name(codec, nid, cfg, "Line Out", label, maxlen, indexp); case AC_JACK_SPEAKER: return fill_audio_out_name(codec, nid, cfg, "Speaker", diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index bc5a993d1146..c83ccdba1e5a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, "Front Speaker", "Surround Speaker", "Bass Speaker" }; static const char * const line_outs[] = { - "Front Line-Out", "Surround Line-Out", "Bass Line-Out" + "Front Line Out", "Surround Line Out", "Bass Line Out" }; fix_volume_caps(codec, dac); @@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, if (num_ctls > 1) name = line_outs[idx]; else - name = "Line-Out"; + name = "Line Out"; break; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index ca117cf5132f..d29d6d377904 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3482,7 +3482,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4fe2d5960a04..f286bb8fda13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -802,7 +802,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, "Disabled", "Enabled" }; static const char * const texts3[] = { - "Disabled", "Speaker Only", "Line-Out+Speaker" + "Disabled", "Speaker Only", "Line Out+Speaker" }; const char * const *texts; @@ -1856,7 +1856,7 @@ static const char * const alc_slave_vols[] = { "Headphone Playback Volume", "Speaker Playback Volume", "Mono Playback Volume", - "Line-Out Playback Volume", + "Line Out Playback Volume", "CLFE Playback Volume", "Bass Speaker Playback Volume", "PCM Playback Volume", @@ -1873,7 +1873,7 @@ static const char * const alc_slave_sws[] = { "Speaker Playback Switch", "Mono Playback Switch", "IEC958 Playback Switch", - "Line-Out Playback Switch", + "Line Out Playback Switch", "CLFE Playback Switch", "Bass Speaker Playback Switch", "PCM Playback Switch", -- cgit v1.2.3