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2008-05-05[ALSA] ac97 - Add a workaround for broken quirk for VT1617A codecTakashi Iwai1-1/+8
On boards with VT1617A codec, the sound disappears suddenly. This looks like a problem with HPE-bit control that is supposed to be set in patch_vt1617a(). However, on such problematic hardwares, the bit is actually reset mysteriously. The patch adds a workaround for the wrong quirk. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-05-05[ALSA] Revert migration to alc_set_pin_output() in ↵Jacek Luczak1-1/+4
alc861_auto_set_output_and_unmute() Change done by: commit f6c7e5461e9046445d50c5c7a9a4587824239623 [ALSA] hda-codec - Fix auto-configuration of Realtek codecs broke sound on ALC861 Analog. Signed-off-by: Jacek Luczak <luczak.jacek@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-05-05[ALSA] fm801 - Fix kconfig dependency mess of fm801-tea575xTakashi Iwai1-4/+1
FM801-tea575x tuner has a reverse selection to V4L1 and this causes nasty dependency problems. The patch simplifies the dependency with a normal "depends on VIDEO_V4L1". This decreases the usability but fixes bugs, yeah. If any better feature like "requires" is introduced to kbuild in future, we'll be able to switch it... Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-05-05[ALSA] hda - Support IDT 92HD206 codecTakashi Iwai1-0/+2
Added the support for IDT 92HD206 codec chip. It's compatible with STAC927x. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-29[ALSA] hda - Add support of Medion RIM 2150Takashi Iwai1-0/+86
Added the support of Medion RIM 2150 laptop with ALC880 codec. ALSA bug#3708: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3708 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-29[ALSA] ice1724 - Enable watermarksTakashi Iwai1-2/+0
Enable watermarks settings (previously commented out) for MPU RX/TX. Otherwise irqs aren't issued properly. Tested-by: Pavel Hofman <pavel.hofman@insite.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-29[ALSA] Add MPU401_INFO_NO_ACK bitflagTakashi Iwai1-0/+1
Added MPU401_INFO_NO_ACK bitflag to ignore the ACK check for UART commands. VT172x doesn't handle ACK commands, for example. Tested-by: Pavel Hofman <pavel.hofman@insite.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] ice1724 - Fix IRQ lock-up with MPU accessTakashi Iwai2-18/+81
The sound boards with VT1724 and compatible chips may lock up when MPU401 is accessed together with the PCM streaming. This patch fixes the problem. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Fix Thinkpad X300 digital micTakashi Iwai1-0/+3
TP X300 digital mic requires additional init verbs with magic COEFs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Fix model for Acer Aspire 5720zTakashi Iwai1-0/+1
Set the proper model=acer for Acer Aspire 5720z with ALC268 codec. ALSA bug#3550: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3550 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] Fix possible races at free_irq in PCI driversTakashi Iwai9-32/+26
The irq handler of PCI drivers must be released before releasing other resources since the handler for a shared irq can be still called and may access the freed resource again. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] ice1712 - Add Terrasoniq TS88 supportPeter Lienig2-1/+18
Added the support of Terrasonq TS88. Signed-off-by: Peter Lienig <lienig@rheinmetall-de.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] Fix synchronize_irq() bugs, redundanciesJeff Garzik20-40/+21
free_irq() calls synchronize_irq() for you, so there is no need for drivers to manually do the same thing (again). Thus, calls where sync-irq immediately precedes free-irq can be simplified. However, during this audit several bugs were noticed, where free-irq is preceded by a "irq >= 0" check... but the sync-irq call is not covered by the same check. So, where sync-irq could not be eliminated completely, the missing check was added. Signed-off-by: Jeff Garzik <jgarzik@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] Audiophile 192: Fix ad converter initializationKarsten Wiese1-2/+2
Correct some arguments in calls to snd_ice1712_gpio_write_bits() from ap192_set_rate_val(). Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] Don't set gpio mask register in snd_ice1712_gpio_write_bits()Karsten Wiese1-2/+6
Some calls to snd_ice1712_gpio_write() go wrong, if snd_ice1712_gpio_write_bits() ran before and changed the gpio mask register. Read the actual gpio value and combine it with the to be set bits in the cpu instead. Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] ice1724.c: toggle "chip reset" and "eeprom based setup" sequenceKarsten Wiese1-3/+8
Let "chip reset" become first. Increasement of the "chip reset" related timeout leads to correctly read eeprom's contents here. Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Avoid unexpected breakage with ALC889A hackTakashi Iwai1-1/+9
The last ALC889A hack may break on some devices with certain model presets since patch_alc*() have different model tables. So, now it's handled in the original patch_alc882() but fly to patch_alc883() in model=auto appropriately. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Fix ALC889A codec supportTakashi Iwai1-0/+2
ALC889A is recognized ALC885/ALC882 but it's actually closer to ALC888/ALC883. Cc: Kasper Sandberg <lkml@metanurb.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda: Add 5.1 support for second headphone jackMatthew Ranostay1-1/+59
Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks, the second headphone jack should be used for the 5.1 surround sound. Add support for 'Headphone as Line Out' switch, which allows it be used in 5.1 surround sound. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] oxygen: generalize DAC volume TLV handlingClemens Ladisch5-26/+15
Add a pointer for DAC volume TLV data to the model structure so that the model driver do not need to manually assign it in their control filter. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] oxygen: mute by defaultClemens Ladisch4-18/+20
Initialize the playback volume controls as being muted and having minimal volume. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] oxygen: generalize handling of DAC volume limitsClemens Ladisch6-28/+17
Add fields for the DAC volume limits to the module structure so that model drivers do not need to install their own control info handlers. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hifier: remove empty hifier_mixer_init()Clemens Ladisch1-6/+0
The empty hifier_mixer_init() function is useless; remove it. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Add support of AD1989A/AD1989BTakashi Iwai1-3/+25
Added the support of AD1989A and AD1989B codecs. These codecs can have multiple SPDIF devices, but currently we handle only one SPDIF. If any real devices with two SPDIF interfaces (likely one for SPDIF and one for HDMI), we'll fix this rightly. Otherwise, these codecs are pretty similar with AD1988. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: fix DX front panel I/OClemens Ladisch1-31/+20
Fix the GPIO 1 mixer control to enable I/O through the front panel connector of the Xonar DX. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda-intel: Add Quanta IL1 ALC267 modelHerton Ronaldo Krzesinski1-0/+76
This adds support for Quanta IL1 mini-notebook to alsa, defining a new model for it. It comes with an ALC267 codec chip. Some notes about this model: * In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common amp mute, to avoid conflict with mixer switch (mixer and automute use the same nid). * The only connected capture sources in the hardware are the internal mic and external mic jack. So instead of using an input source selector like on other ALC268 models, the mic automute automatically switch between captures. Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda: EAPD power managementMatthew Ranostay1-6/+19
Power management support for EAPD enabled laptops, when headphones are sensed it pulls the EAPD GPIO line low to power it down. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda: Correct SPDIF out default configMatthew Ranostay1-0/+7
Several laptops have have the SPDIF out defined as 'Digital other out' when it should be 'SPDIF out' in the default config. Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Fujitsu Lifebook PC speaker signalTony Vroon1-0/+2
The legacy PC speaker signal was not routed to outputs. The codec is not prevented from powering down in this patch, although I suppose one could argue that perhaps it should be. Let me know if anyone feels strongly one way or the other. Signed-off-by: Tony Vroon <tony@linx.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - PCI quirk for laptop LG which use CMI9880Jiang zhe1-0/+1
Please refer to [0003874] on the alsa mantis. This patch added the pci quirk. Signed-off-by: Jiang zhe <zhe.jiang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880Jiang zhe1-1/+1
To mute the output of Pin widget 15 in ALC880, we should use the HDA_OUTPUT. However, current code looks like : snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits); It may be a misspelling. Signed-off-by: Jiang zhe <zhe.jiang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda-codec - Fix unbalanced mutexFrederik Deweerdt1-1/+1
On Wed, Apr 02, 2008 at 08:19:29AM -0400, Miles Lane wrote: > [ 48.765906] [ BUG: bad unlock balance detected! ] > [ 48.765912] ------------------------------------- > [ 48.765918] pulseaudio/4277 is trying to release lock > (&codec->spdif_mutex) at: > [ 48.765930] [<c03031b7>] mutex_unlock+0x8/0xa > [ 48.765945] but there are no more locks to release! > [ 48.765950] > [ 48.765952] other info that might help us debug this: > [ 48.765959] 2 locks held by pulseaudio/4277: > [ 48.765965] #0: (&pcm->open_mutex){--..}, at: [<f89f134b>] > snd_pcm_open+0xc1/0x1ba [snd_pcm] > [ 48.766003] #1: (&chip->open_mutex){--..}, at: [<f8b4f13d>] > azx_pcm_open+0x36/0x184 [snd_hda_intel] > [ 48.766057] > [ 48.766059] stack backtrace: > [ 48.766066] Pid: 4277, comm: pulseaudio Not tainted 2.6.25-rc8-mm1 #12 > [ 48.766086] [<c013afc6>] print_unlock_inbalance_bug+0xce/0xd8 > [ 48.766107] [<c0109e1c>] ? save_stack_trace+0x1d/0x3b > [ 48.766130] [<c012f54e>] ? __kernel_text_address+0x1b/0x27 > [ 48.766146] [<c0104533>] ? dump_trace+0xcd/0xd9 > [ 48.766160] [<c0109d9e>] ? save_stack_address+0x0/0x2c > [ 48.766176] [<c013b80a>] ? find_usage_backwards+0xa4/0xc3 > [ 48.766193] [<c013cfb5>] lock_release_non_nested+0x84/0x120 > [ 48.766209] [<c03031b7>] ? mutex_unlock+0x8/0xa > [ 48.766222] [<c013d1bb>] lock_release+0x16a/0x199 > [ 48.766238] [<c0303137>] __mutex_unlock_slowpath+0xa9/0x121 > [ 48.766252] [<c03031b7>] mutex_unlock+0x8/0xa > [ 48.766263] [<f8b4ffd8>] snd_hda_multi_out_analog_open+0xd3/0xef > [snd_hda_intel] The following patch should fix it. Cc: "Miles Lane" <miles.lane@gmail.com> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] es1968 - fix coding style in the last patchAndrew Morton1-2/+1
WARNING: braces {} are not necessary for single statement blocks #40: FILE: sound/pci/es1968.c:1831: + if (diff > 1) { + __maestro_write(chip, IDR0_DATA_PORT, cp1); + } total: 0 errors, 1 warnings, 35 lines checked ./patches/es1968-fix-jitter-on-some-maestro-cards.patch has style problems, please review. If any of these errors are false positives report them to the maintainer, see CHECKPATCH in MAINTAINERS. Please run checkpatch prior to sending patches Cc: Andreas Mueller <andreas@stapelspeicher.org> Tested-by: Rene Herman <rene.herman@keyaccess.nl> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] es1968: fix jitter on some maestro cardsAndreas Mueller1-1/+21
This patch suppresses jitter on several Maestro cards in stereo mode (ALSA of course). The patch is also incorporated in the *BSD drivers where I "ported" it from. Without this patch most of the stereo audio gets out of sync and really distorted (oss-emulation with mplayer at 48000khz worked somehow). Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] sound/pci/rme9652/hdspm.c: stop inlining largish static functionsDenys Vlasenko1-7/+8
sound/pci/rme9652/hdspm.c has unusually large number of static inline functions - 22. I looked through them and some of them seem to be too big to warrant inlining. This patch removes "inline" from these static functions (regardless of number of callsites - gcc nowadays auto-inlines statics with one callsite). Size difference on 32bit x86: text data bss dec hex filename 20437 2160 516 23113 5a49 linux-2.6-ALLYES/sound/pci/rme9652/hdspm.o 18036 2160 516 20712 50e8 linux-2.6.inline-ALLYES/sound/pci/rme9652/hdspm.o [coding fix by Takashi Iwai <tiwai@suse.de>] Signed-off-by: Denys Vlasenko <vda.linux@googlemail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] hda-codec - PCI quirk for MSI laptopJiang zhe1-0/+4
Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Signed-off-by: Jiang zhe <zhe.jiang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: initialize two-wire control registerClemens Ladisch1-3/+4
On the Xonar DX, initialize all bits of the two-wire control register. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: add GPIO 1 mixer controlClemens Ladisch1-0/+53
Add a mixer control for switching whatever it is that is connected to GPIO pin 1 on the Xonar DX. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] oxygen: use SPDIF input only if presentClemens Ladisch4-28/+40
If the card model does not have a digital input or an AC97 codec, disable the respective interrupt and mixer controls. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: correctly switch input jack on Xonar DXClemens Ladisch3-4/+24
When selecting the capture source on the Xonar DX, the input jack must be routed to either the line input or the microphone input by setting a GPIO pin. This requires an additional callback so that the model driver can hook into the toggling of AC97 switches. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: add Xonar DX supportClemens Ladisch4-4/+365
Add support for the Asus Xonar DX. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: fix typoClemens Ladisch1-1/+1
Fix a (fortunately harmless) typo. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: change card short nameClemens Ladisch1-2/+2
Change the card short name to show to show the card name instead of the chip name. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: set PCM1796 oversampling rateClemens Ladisch1-2/+0
When playing data at 96 kHz or higher, reduce the DAC oversampling rate to 32. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: move some code to xonar_common_init()Clemens Ladisch1-26/+51
Move the code that is common to all Xonar models to a separate function, and make it more generic in preparation for another model. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: allow both CS5381 and CS5361Clemens Ladisch1-14/+15
Rename all CS5381 symbols to CS53x1 because they can also be used for Xonar models with a CS5361. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] virtuoso: separate D2/D2X init functionsClemens Ladisch1-44/+74
Use separate model structures for the D2 and D2X so that the init function does not have to check for the model again. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] oxygen: add I2C supportClemens Ladisch2-2/+22
Add a function to write I2C registers. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] aw2: remove duplicate MODULE_LICENSEClemens Ladisch1-1/+0
"GPL 2" does not mean that there have to be two MODULE_LICENSE("GPL") entries. ;-) Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24[ALSA] oxygen: fix line-in recording selection (now for real)Clemens Ladisch5-92/+31
On C-Media cards, the GPIO pin 0 of the CM9780 must be handled exactly like on Xonar cards, so move the Xonar code to the common mixer code. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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