From 0a1896b27b030529ec770aefd790544a1bdb7d5a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 6 Jun 2011 18:55:34 -0400 Subject: ALSA: hda: Fix quirk for Dell Inspiron 910 BugLink: https://launchpad.net/bugs/792712 The original reporter states that sound from the internal speakers is inaudible until using the model=auto quirk. This symptom is due to an existing quirk mask for 0x102802b* that uses the model=dell quirk. To limit the possible regressions, leave the existing quirk mask but add a higher priority specific mask for the reporter's PCI SSID. Reported-and-tested-by: rodni hipp Cc: [2.6.38+] Signed-off-by: Daniel T Chen 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 7a4e10002f5..d7007896772 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13860,6 +13860,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), + SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO), SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), /* almost compatible with toshiba but with optional digital outs; -- cgit v1.2.3-18-g5258 From 20f5e0b36d968326fab3b720035f226113e34ae9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 09:31:54 +0200 Subject: ALSA: hda - Fix invalid unsol tag for some alc262 model quirks The tag number was forgotten to be fixed after cleaning up the model quirks for ALC262 fujitsu and lenovo-3000 models. Tested-by: Michal Hocko 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 d7007896772..ca211c1cba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11924,7 +11924,7 @@ static const struct hda_verb alc262_nec_verbs[] = { * 0x1b = port replicator headphone out */ -#define ALC_HP_EVENT 0x37 +#define ALC_HP_EVENT ALC880_HP_EVENT static const struct hda_verb alc262_fujitsu_unsol_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, -- cgit v1.2.3-18-g5258 From c0a20263dbe1fc5f394913d71063c9cd8282c5db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 15:28:15 +0200 Subject: ALSA: hda - Fix initialization of hp pins with master_mute in Realtek Some Reatlek model quirks use master_mute bool switch for controlling the master-mute of outputs. For these cases, the initialization of HP pins/amps were forgotten during the transition to the common automute helper function in 3.0 development time, and resulted in the muted HP output as default. This patch fixes the issue by adjusting the HP output explicitly with master_mute switch. Tested-by: Michal Hocko Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ca211c1cba0..43fcfbd3284 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int on; + /* Control HP pins/amps depending on master_mute state; + * in general, HP pins/amps control should be enabled in all cases, + * but currently set only for master_mute, just to be safe + */ + do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + spec->autocfg.hp_pins, spec->master_mute, true); + if (!spec->automute) on = 0; else @@ -6201,11 +6208,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = { /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; - - /* change HP pins */ - do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), - spec->autocfg.hp_pins, spec->master_mute, true); update_speakers(codec); } -- cgit v1.2.3-18-g5258 From 7ab1fc0af3464d231e17eb729a03495d93d0cc5c Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 10 Jun 2011 10:14:01 -0400 Subject: ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop BugLink: https://launchpad.net/bugs/761171 The original reporter needs the model=auto quirk for his internal speakers to be audible in the latest daily snapshot, so add an entry in the quirk table for his PCI SSID. A trivially different version of this patch using the model=asus quirk should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much improved. Reported-and-tested-by: tomdeering7 Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3e6b9a8539c..694b9daf691 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3102,6 +3102,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ + SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; -- cgit v1.2.3-18-g5258 From c0da00145f9a32ef33b14508e6fd90fc130afbdc Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Sun, 12 Jun 2011 17:26:17 +0200 Subject: ALSA: hdspm - Fix locking in snd_hdspm_midi_input_read For the MIDI part, we need to acquire (and release) the hmidi->lock, access to the global hdspm structure is serialized through hmidi->hdspm->lock instead. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 949691a876d..32d80af012c 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1639,12 +1639,14 @@ static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi) } } hmidi->pending = 0; + spin_unlock_irqrestore(&hmidi->lock, flags); + spin_lock_irqsave(&hmidi->hdspm->lock, flags); hmidi->hdspm->control_register |= hmidi->ie; hdspm_write(hmidi->hdspm, HDSPM_controlRegister, hmidi->hdspm->control_register); + spin_unlock_irqrestore(&hmidi->hdspm->lock, flags); - spin_unlock_irqrestore (&hmidi->lock, flags); return snd_hdspm_midi_output_write (hmidi); } -- cgit v1.2.3-18-g5258 From fedf1535ab5ee02acbbc235c2272d84bb9334758 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Sun, 12 Jun 2011 17:26:18 +0200 Subject: ALSA: hdspm - Fix jumping external wordclock frequency in AutoSync mode When using Word Clock on RME MADI cards, AutoSync mode was alternating betweeen MADI and WC due to a typo: AutoSync is indicated in the second status register (status2), not the first one (status). While the proc output was always correct, the reported WC frequency to ALSA was unstable as mentioned in http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 32d80af012c..d03ef94d570 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1143,7 +1143,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) /* if wordclock has synced freq and wordclock is valid */ if ((status2 & HDSPM_wcLock) != 0 && - (status & HDSPM_SelSyncRef0) == 0) { + (status2 & HDSPM_SelSyncRef0) == 0) { rate_bits = status2 & HDSPM_wcFreqMask; -- cgit v1.2.3-18-g5258 From efef054e8c4bc4fd48a0b4deb5491116d9f557c7 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Sun, 12 Jun 2011 17:26:19 +0200 Subject: ALSA: hdspm - Add firmware revision ID for RME MADI PCI version The PCI version of the RME HDSP MADI card uses 0xcf as revision ID. Just add this to the list of supported cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d03ef94d570..3f08afc0f0d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -521,6 +521,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024) /* revisions >= 230 indicate AES32 card */ +#define HDSPM_MADI_OLD_REV 207 #define HDSPM_MADI_REV 210 #define HDSPM_RAYDAT_REV 211 #define HDSPM_AIO_REV 212 @@ -6379,6 +6380,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, switch (hdspm->firmware_rev) { case HDSPM_MADI_REV: + case HDSPM_MADI_OLD_REV: hdspm->io_type = MADI; hdspm->card_name = "RME MADI"; hdspm->midiPorts = 3; -- cgit v1.2.3-18-g5258 From ac5d4b404e78bd7eb67fc70c2acb437a25497e98 Mon Sep 17 00:00:00 2001 From: Florian Zeitz Date: Sun, 12 Jun 2011 01:15:42 +0200 Subject: ALSA: emu10k1: Add details for E-mu 0404 PCIe version This patch adds the necessary details to support the PCIe version of E-MU's 0404 card. From comparing the PCBs it seems the PCIe version just added a PCIe chipset and left all other components pretty much in place. For anyone intrigued to take a look at the PCB there are pictures I took at . Signed-off-by: Florian Zeitz Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 5e619a84da0..15f0161ce4a 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ + /* EMU0404 PCIe */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102, + .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */ /* Note that all E-mu cards require kernel 2.6 or newer. */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]", -- cgit v1.2.3-18-g5258 From 54463a66b91cf491a7c9af612b0e310babc5fa24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Jun 2011 08:32:06 +0200 Subject: ALSA: hda - Fix wrong auto-mute type for Acer Aspire-one The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly during the clean-up of auto-mute function. Fixed now. Tested-by: Borislav Petkov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 43fcfbd3284..61a774b3d3c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13316,9 +13316,8 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0f; spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; + spec->automute_mode = ALC_AUTOMUTE_AMP; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; -- cgit v1.2.3-18-g5258 From 2308f4add3de9f6c9c9f02e49461e94d84bb200a Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sun, 12 Jun 2011 13:02:43 -0700 Subject: ALSA: hda - Fix beep_device compilation warnings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Using static inline functions can reduce compilation messages and macro misuse. sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’: sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.h | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index f1de1bac042..4967eabe774 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -50,7 +50,12 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) 0 -#define snd_hda_detach_beep_device(...) +static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +{ + return 0; +} +void snd_hda_detach_beep_device(struct hda_codec *codec) +{ +} #endif #endif -- cgit v1.2.3-18-g5258 From ca2585afa013ec2cf99a48e46d6b82df2e240493 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Jun 2011 08:14:32 +0200 Subject: ALSA: hda - Fix missing static inline to beep dummy function The commit 2308f4add3de9f6c9c9f02e49461e94d84bb200a missed static inline thus it resulted in multiple-definitions error at linking. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 4967eabe774..55f0647458c 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -54,7 +54,7 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { return 0; } -void snd_hda_detach_beep_device(struct hda_codec *codec) +static inline void snd_hda_detach_beep_device(struct hda_codec *codec) { } #endif -- cgit v1.2.3-18-g5258 From e72888e91cc902ccdc089f237b6eed7587e2b4df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Jun 2011 15:14:49 +0200 Subject: ALSA: lola - Fix section mismatch Add missing __devinit. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 34b24286d27..2692e5ae5f2 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip) lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */ } -static int lola_parse_tree(struct lola *chip) +static int __devinit lola_parse_tree(struct lola *chip) { unsigned int val; int nid, err; -- cgit v1.2.3-18-g5258 From ad2409413d09fca763be1ac5161f2a9d82367903 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2011 14:23:46 +0200 Subject: ALSA: hda - Fix no NID error with VIA codecs The via driver spews warnigs like hda-codec: no NID for mapping control Independent HP:0:0 with some codecs because snd_hda_add_nid() is called with nid=0. This patch fixes it by skipping the call when no corresponding widget is found. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 605c99e1e52..c952582fb21 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -832,10 +832,13 @@ static int via_hp_build(struct hda_codec *codec) knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; - knew = via_clone_control(spec, &via_hp_mixer[1]); - if (knew == NULL) - return -ENOMEM; - knew->subdevice = side_mute_channel(spec); + nid = side_mute_channel(spec); + if (nid) { + knew = via_clone_control(spec, &via_hp_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = nid; + } return 0; } -- cgit v1.2.3-18-g5258 From 6f2e810ad5d162c2bfa063c1811087277b299e4e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Jun 2011 10:27:07 +0200 Subject: ALSA: HDA: Remove quirk for an HP device The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. BugLink: http://bugs.launchpad.net/bugs/761022 Cc: stable@kernel.org (v2.6.38+) Reported-by: Jo 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 61a774b3d3c..c923b2cc9e5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4883,7 +4883,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST), SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), -- cgit v1.2.3-18-g5258 From c933790614529c06b221f73ff36e2456aecee30d Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 20 Jun 2011 22:11:11 +0100 Subject: ALSA: hda - Remove ALC268 model override for CPR2000 The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER. This keeps headphone automute and microphone input from operating on at least one laptop from Opti Systems. Without the override, the BIOS parser does a fine job setting the card up and everything works. Tested-By: Peter Schneider Signed-off-by: Tony Vroon 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 c923b2cc9e5..475ed1e8ffc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13871,7 +13871,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), {} }; -- cgit v1.2.3-18-g5258 From 42467b32ce4f1ba933673b396f807110e3618ff5 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Mon, 20 Jun 2011 14:14:37 +0800 Subject: ALSA: VIA HDA: Modify initial verbs list for VT1718S. Remove some invalid initial verbs and correct some wrong initial verbs for VT1718S codec. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c952582fb21..abee9ac1590 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4283,9 +4283,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = { {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - - /* Setup default input of Front HP to MW9 */ - {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 PW10 Output enable */ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, @@ -4294,10 +4291,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = { /* Enable Boost Volume backdoor */ {0x1, 0xf88, 0x8}, /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -4307,8 +4304,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = { /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Unmute MW4's index 0 */ - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } }; -- cgit v1.2.3-18-g5258 From ba31a60d0fd8a3976d44d32f2b82491c62646b2a Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Mon, 20 Jun 2011 14:16:33 +0800 Subject: ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S. When switch HP independent mode, mute/unmute connctions of mixer which is connected to headphone for VT1718S. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index abee9ac1590..f1a80cd6afe 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; + unsigned int parm0, parm1; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - if (spec->codec_type == VT1718S) + if (spec->codec_type == VT1718S) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + /* Set correct mute switch for MW3 */ + parm0 = spec->hp_independent_mode ? + AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0); + parm1 = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm0); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm1); + } else snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); -- cgit v1.2.3-18-g5258 From e905a83acd7bf8989c3d5ba3099b72675f5d7d29 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Mon, 20 Jun 2011 14:17:56 +0800 Subject: ALSA: VIA HDA: Create a master amplifier control for VT1718S. Create a master volume and mute control of playback for VT1718S. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index f1a80cd6afe..f43bb0eaed8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4462,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; /* Front */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control( -- cgit v1.2.3-18-g5258 From d2a19da79d3ea5b7859248b0f132c479ed4505e2 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 22 Jun 2011 09:58:37 +0200 Subject: ALSA: HDA: Pinfix quirk for HP Z200 Workstation BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. BugLink: http://bugs.launchpad.net/bugs/754964 Reported-by: Marc Legris Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 475ed1e8ffc..d21191dcfe8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12599,6 +12599,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { */ enum { PINFIX_FSC_H270, + PINFIX_HP_Z200, }; static const struct alc_fixup alc262_fixups[] = { @@ -12611,9 +12612,17 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, + [PINFIX_HP_Z200] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x16, 0x99130120 }, /* internal speaker */ + { } + } + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270), {} }; @@ -12730,6 +12739,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { ALC262_HP_BPC), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", + ALC262_AUTO), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), -- cgit v1.2.3-18-g5258 From 16866741bda5d16f3d30d1656ce941faf5dad34c Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 23 Jun 2011 23:54:40 +0200 Subject: ALSA: Remove unneeded version.h includes from sound/ In the sound/ directory there are two files (flagged by 'make versioncheck'); sound/pci/asihpi/asihpi.c and sound/soc/codecs/wm8991.c that include linux/version.h although they don't need it. This patch removes the unneeded includes. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 2ca6f4f85b4..e3569bdd3b6 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -27,7 +27,6 @@ #include "hpioctl.h" #include -#include #include #include #include -- cgit v1.2.3-18-g5258 From f0ca89b031d327b80b612a0608d31b8e13e6dc33 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 21 Jun 2011 20:51:34 +0200 Subject: ALSA: HDA: Add a new Conexant codec ID (506c) Conexant ID 506c was found on Acer Aspire 3830TG. As users report no playback, sending to stable should be safe. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/783582 Reported-by: andROOM Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 694b9daf691..4158949ea07 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4389,6 +4389,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f1506c, .name = "CX20588", + .patch = patch_cxt5066 }, { .id = 0x14f1506e, .name = "CX20590", .patch = patch_cxt5066 }, { .id = 0x14f15097, .name = "CX20631", @@ -4417,6 +4419,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f1506c"); MODULE_ALIAS("snd-hda-codec-id:14f1506e"); MODULE_ALIAS("snd-hda-codec-id:14f15097"); MODULE_ALIAS("snd-hda-codec-id:14f15098"); -- cgit v1.2.3-18-g5258 From 9966db22caf8f74c0e6d84a569e6d7d56332e127 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 21 Jun 2011 21:01:52 +0200 Subject: ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TG Since we're not using the new auto parser as a fallback yet, add it manually as a quirk. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4158949ea07..7bbc5f237a5 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3074,6 +3074,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), -- cgit v1.2.3-18-g5258 From f5b2d0ef631bb0647ae8ed1752d2127b8fb6da70 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 29 Jun 2011 14:26:07 +0800 Subject: ALSA: HDMI - fix ELD monitor name length I noticed that the last character of the ELD monitor name is lost, this fixes the issue. This fix should be confirming to the HDA spec, and works together with the DRM part of the ELD patch. The HDA spec does not mention that Monitor_Name_String is an '\0' ending string, and it allows NML to be 1, which is only valid when MNL does not count the possible ending '\0'. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index b05f7be9dc1..e3e853153d1 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_eld *e, snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else - strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl); + strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { -- cgit v1.2.3-18-g5258 From 71276410e17653cfacfa238a363475cde9e18fb3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2011 12:31:23 +0200 Subject: ALSA: cs5535 - Fix invalid big-endian conversions Fix the wrongly converted short values: sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index f16bc8aad6e..e083122ca55 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -149,7 +149,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i]; desc->addr = cpu_to_le32(addr); desc->size = cpu_to_le32(period_bytes); - desc->ctlreserved = cpu_to_le32(PRD_EOP); + desc->ctlreserved = cpu_to_le16(PRD_EOP); desc_addr += sizeof(struct cs5535audio_dma_desc); addr += period_bytes; } @@ -157,7 +157,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods]; lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr); lastdesc->size = 0; - lastdesc->ctlreserved = cpu_to_le32(PRD_JMP); + lastdesc->ctlreserved = cpu_to_le16(PRD_JMP); jmpprd_addr = cpu_to_le32(lastdesc->addr + (sizeof(struct cs5535audio_dma_desc)*periods)); -- cgit v1.2.3-18-g5258 From 286bed0f0c447b6660e72093d7e778784fdd9ee6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2011 12:45:36 +0200 Subject: ALSA: hdspm - Fix compile warnings with PPC The char can be unsigned on some architectures. Since the code checks the negative values, they should be declared as signed char explicitly. sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 3f08afc0f0d..c8e402fc378 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -896,11 +896,11 @@ struct hdspm { unsigned char max_channels_in; unsigned char max_channels_out; - char *channel_map_in; - char *channel_map_out; + signed char *channel_map_in; + signed char *channel_map_out; - char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs; - char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs; + signed char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs; + signed char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs; char **port_names_in; char **port_names_out; -- cgit v1.2.3-18-g5258 From 9c7a083d94656ad6d6f2e03ba90194f2cc5bced5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Jul 2011 09:25:54 +0200 Subject: ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek When the dual-adc switching mode is active in Realtek auto-parser, we need to couple all ADCs as a single capture-volume. Currently, the volume control changes only the first ADC, thus others may remain silent. This patch fixes the problem. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++---------- 1 file changed, 23 insertions(+), 10 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d21191dcfe8..7d492713c1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol, static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - getput_call_t func) + getput_call_t func, bool check_adc_switch) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - int err; + int i, err; mutex_lock(&codec->control_mutex); - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], - 3, 0, HDA_INPUT); - err = func(kcontrol, ucontrol); + if (check_adc_switch && spec->dual_adc_switch) { + for (i = 0; i < spec->num_adc_nids; i++) { + kcontrol->private_value = + HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + if (err < 0) + goto error; + } + } else { + i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + kcontrol->private_value = + HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + } + error: mutex_unlock(&codec->control_mutex); return err; } @@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_volume_get); + snd_hda_mixer_amp_volume_get, false); } static int alc_cap_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_volume_put); + snd_hda_mixer_amp_volume_put, true); } /* capture mixer elements */ @@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_switch_get); + snd_hda_mixer_amp_switch_get, false); } static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_switch_put); + snd_hda_mixer_amp_switch_put, true); } #define _DEFINE_CAPMIX(num) \ -- cgit v1.2.3-18-g5258 From abaead6ac55dbda8b4bae40251d69dc3f0c49b1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 9 Jul 2011 11:55:28 +0200 Subject: ALSA: hda - Fix a copmile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It's harmless but annyoing. sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’: sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function 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 7d492713c1c..b48fb43b544 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2719,7 +2719,7 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int i, err; + int i, err = 0; mutex_lock(&codec->control_mutex); if (check_adc_switch && spec->dual_adc_switch) { -- cgit v1.2.3-18-g5258