From 150fe14c1a1f08cb430d8382bf5554c2a168b79b Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 19 Aug 2009 16:58:59 +0800 Subject: ALSA: hda: enable speaker output for Compaq 6530s/6531s HP Compaq 6530s and 6531s internal speaker is silence or becomes silence within 1 minute after fresh boot. It is found that pin 0x1c must be set to PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and speaker pin 0x16 seem to be unrelated. The codec differences before/after patch are: @@ Node 0x17 [Pin Complex] wcaps 0x40020b: Pin Default 0x41a6e130: [N/A] Mic at Ext Rear Conn = Digital, Color = White DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE - Pin-ctls: 0x24: IN + Pin-ctls: 0x40: OUT @@ Node 0x1c [Pin Complex] wcaps 0x40018d: Pin Default 0x41813021: [N/A] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x2, Sequence = 0x1 - Pin-ctls: 0x24: IN VREF_80 + Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x24 Tests show that it won't impact (external) Mic recording. Reported-by: "Lin, Ming M" Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e8e6a43865c..f2bb4803417 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3823,9 +3823,11 @@ static struct hda_verb ad1884a_laptop_verbs[] = { /* Port-F (int speaker) mixer - route only from analog mixer */ {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Port-F (int speaker) pin */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* required for compaq 6530s/6531s speaker output */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-C pin - internal mic-in */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ -- cgit v1.2.3 From ae709440edb2d36f51f5ea51cfab931f45c03e02 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 19 Aug 2009 17:05:11 +0800 Subject: ALSA: hda: add model for Intel DG45ID/DG45FC boards The BIOS pin configs are in fact correct and shall not be overwritten. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 456ef6ac12e..6990cfcb6a3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -76,6 +76,7 @@ enum { STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, + STAC_92HD73XX_INTEL, STAC_DELL_M6_AMIC, STAC_DELL_M6_DMIC, STAC_DELL_M6_BOTH, @@ -1777,6 +1778,7 @@ static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", + [STAC_92HD73XX_INTEL] = "intel", [STAC_DELL_M6_AMIC] = "dell-m6-amic", [STAC_DELL_M6_DMIC] = "dell-m6-dmic", [STAC_DELL_M6_BOTH] = "dell-m6", @@ -1789,6 +1791,10 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5002, + "Intel DG45ID", STAC_92HD73XX_INTEL), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5003, + "Intel DG45FC", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, -- cgit v1.2.3 From 3abf2f3639959e4f53f209f93cd4d93fe9356de1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 19 Aug 2009 20:05:02 +0200 Subject: ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec There are many variants of Toshiba laptops with ALC268 codec, and it seems that a few of them don't work with model=toshiba preset since they have the secondary ALC268 codec just for HDMI output. This is a regression due to the previous clean-up work to merge all Toshiba quirk entries into a single check. This patch adds the identification of such laptops to apply the standard BIOS-probing method. Unfortunately, Toshiba laptops have all the same PCI SSID, so we need to check the codec SSID to identify each device. Tested-by: Alexey Dobriyan Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fea976793ae..6f683e451f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12521,8 +12521,6 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), - SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", - ALC268_TOSHIBA), 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), @@ -12530,6 +12528,15 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { {} }; +/* Toshiba laptops have no unique PCI SSID but only codec SSID */ +static struct snd_pci_quirk alc268_ssid_cfg_tbl[] = { + SND_PCI_QUIRK(0x1179, 0xff0a, "TOSHIBA X-200", ALC268_AUTO), + SND_PCI_QUIRK(0x1179, 0xff0e, "TOSHIBA X-200 HDMI", ALC268_AUTO), + SND_PCI_QUIRK_MASK(0x1179, 0xff00, 0xff00, "TOSHIBA A/Lx05", + ALC268_TOSHIBA), + {} +}; + static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, @@ -12696,6 +12703,10 @@ static int patch_alc268(struct hda_codec *codec) alc268_models, alc268_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) + board_config = snd_hda_check_board_codec_sid_config(codec, + ALC882_MODEL_LAST, alc268_models, alc268_ssid_cfg_tbl); + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { printk(KERN_INFO "hda_codec: Unknown model for %s, " "trying auto-probe from BIOS...\n", codec->chip_name); -- cgit v1.2.3 From 70bdbd3d1ae9c4ca3e84a43df34262face26575d Mon Sep 17 00:00:00 2001 From: Bartlomiej Zolnierkiewicz Date: Sun, 23 Aug 2009 15:27:25 +0200 Subject: ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready() Modify loops in such way that the register value is checked also after the timeout condition, just in case the heavy interrupt load etc. caused the thread to sleep for the time period exceeding the timeout value. While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready(). Reported-by: Jack Byer Signed-off-by: Bartlomiej Zolnierkiewicz Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index c551006e292..76d76c08339 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -310,12 +310,16 @@ static int snd_ali_codec_ready(struct snd_ali *codec, unsigned int res; end_time = jiffies + msecs_to_jiffies(250); - do { + + for (;;) { res = snd_ali_5451_peek(codec,port); if (!(res & 0x8000)) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } + snd_ali_5451_poke(codec, port, res & ~0x8000); snd_printdd("ali_codec_ready: codec is not ready.\n "); return -EIO; @@ -327,15 +331,17 @@ static int snd_ali_stimer_ready(struct snd_ali *codec) unsigned long dwChk1,dwChk2; dwChk1 = snd_ali_5451_peek(codec, ALI_STIMER); - dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); - end_time = jiffies + msecs_to_jiffies(250); - do { + + for (;;) { dwChk2 = snd_ali_5451_peek(codec, ALI_STIMER); if (dwChk2 != dwChk1) return 0; + if (!time_after_eq(end_time, jiffies)) + break; schedule_timeout_uninterruptible(1); - } while (time_after_eq(end_time, jiffies)); + } + snd_printk(KERN_ERR "ali_stimer_read: stimer is not ready.\n"); return -EIO; } -- cgit v1.2.3 From edd1365e90eb32625041d09de427d7b03461bc5c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 24 Aug 2009 09:11:58 +0200 Subject: sound: vx222: fix input level control range check Fix a logic error in the range check of the input level control that would prevent setting any volume less than the maximum. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/vx222/vx222_ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 6416d3f0c7b..a69e774d0b1 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -885,10 +885,10 @@ static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vx222 *chip = (struct snd_vx222 *)_chip; if (ucontrol->value.integer.value[0] < 0 || - ucontrol->value.integer.value[0] < MIC_LEVEL_MAX) + ucontrol->value.integer.value[0] > MIC_LEVEL_MAX) return -EINVAL; if (ucontrol->value.integer.value[1] < 0 || - ucontrol->value.integer.value[1] < MIC_LEVEL_MAX) + ucontrol->value.integer.value[1] > MIC_LEVEL_MAX) return -EINVAL; mutex_lock(&_chip->mixer_mutex); if (chip->input_level[0] != ucontrol->value.integer.value[0] || -- cgit v1.2.3 From 0f67a611629f84dd0afacd23d422b4b9c2558285 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 31 Aug 2009 08:12:29 +0200 Subject: ALSA: hda - Add missing mux check for VT1708 In patch_vt1708(), the check of MUX nids is missing and this results in the -EINVAL error in accessing Input Source mixer element. Simpliy adding the call of get_mux_nids() fixes the problem. Reference: Novell bnc#534904 https://bugzilla.novell.com/show_bug.cgi?id=534904 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 9008b4b013a..e8f10b10cce 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1395,6 +1395,7 @@ static int patch_vt1708(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708_capture_mixer; spec->num_mixers++; } -- cgit v1.2.3 From a3f730af7e33cea10ea66f05b2565fde1f9512df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 31 Aug 2009 08:15:26 +0200 Subject: ALSA: hda - Fix MacBookPro 3,1/4,1 quirk with ALC889A This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1 quirk with ALC889A codec, which caused the silent headphone output. Also, this gives the individual Headphone and Speaker volume controls. Reference: kernel bug#14078 http://bugzilla.kernel.org/show_bug.cgi?id=14078 Signed-off-by: Takashi Iwai Cc: --- sound/pci/hda/patch_realtek.c | 34 ++++++++++++++++++++-------------- 1 file changed, 20 insertions(+), 14 deletions(-) (limited to 'sound/pci') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6f683e451f2..30eeb304351 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6423,9 +6423,9 @@ static struct hda_verb alc885_mbp_ch2_init[] = { }; /* - * 6ch mode + * 4ch mode */ -static struct hda_verb alc885_mbp_ch6_init[] = { +static struct hda_verb alc885_mbp_ch4_init[] = { { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, @@ -6434,9 +6434,9 @@ static struct hda_verb alc885_mbp_ch6_init[] = { { } /* end */ }; -static struct hda_channel_mode alc885_mbp_6ch_modes[2] = { +static struct hda_channel_mode alc885_mbp_4ch_modes[2] = { { 2, alc885_mbp_ch2_init }, - { 6, alc885_mbp_ch6_init }, + { 4, alc885_mbp_ch4_init }, }; /* @@ -6497,10 +6497,11 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { }; static struct snd_kcontrol_new alc885_mbp3_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), @@ -6814,14 +6815,18 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Front Pin: output 0 (0x0c) */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: output 0 (0x0d) */ + /* HP Pin: output 0 (0x0e) */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Mic (rear) pin: input vref at 80% */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, @@ -7195,10 +7200,11 @@ static struct alc_config_preset alc882_presets[] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs, alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .num_dacs = 2, .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mbp_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes), + .hp_nid = 0x04, + .channel_mode = alc885_mbp_4ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, -- cgit v1.2.3 From b91ab72b830e1494c2c7f8de05ccb2ab2c9cfb26 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 1 Sep 2009 08:23:58 +0200 Subject: sound: oxygen: fix MCLK rate for 192 kHz playback Do not forget to program the MCLK ratio for the I2S output. Otherwise, the master clock frequency can be too high for the DACs at sample frequencies above 96 kHz. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index 3b5ca70c9d4..ef2345d82b8 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -469,9 +469,11 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, oxygen_rate(hw_params) | chip->model.dac_i2s_format | + oxygen_i2s_mclk(hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | + OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); oxygen_update_dac_routing(chip); oxygen_update_spdif_source(chip); -- cgit v1.2.3 From 92653453c3015c083b9fe0ad48261c6b2267d482 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 2 Sep 2009 18:25:39 +0200 Subject: sound: oxygen: handle cards with missing EEPROM MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The card model detection code introduced in 2.6.30 that tries to work around partially broken EEPROM contents by reading the EEPROM directly does not handle cards where the EEPROM has been omitted. In this case, we have to use the default ID to allow the driver to load. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Ozan Çağlayan Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_lib.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/pci') diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 312251d3969..9a8936e2074 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -260,6 +260,9 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) * chip didn't if the first EEPROM word was overwritten. */ subdevice = oxygen_read_eeprom(chip, 2); + /* use default ID if EEPROM is missing */ + if (subdevice == 0xffff) + subdevice = 0x8788; /* * We use only the subsystem device ID for searching because it is * unique even without the subsystem vendor ID, which may have been -- cgit v1.2.3