summaryrefslogtreecommitdiffstats
Commit message (Collapse)AuthorAgeFilesLines
* [ALSA] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SLRisto Suominen2008-04-242-36/+151
| | | | | | | | | Add mixer controls and correct headphone detection bits for PowerMacs 7300/7500 (AWACS) and G3 Beige (Screamer), and iMac G3 Slot-loading (Screamer). Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] snd-powermac: style pmac.cRisto Suominen2008-04-241-2/+2
| | | | | | | Coding style corrections for pmac.c. Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] snd-powermac: enable headphone detectionRisto Suominen2008-04-241-3/+3
| | | | | | | | Enable port change interrupt while initialising AWACS, Screamer, and Burgundy chipsets. Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] sound/drivers/dummy.c: fix negative snd_pcm_format_width() checkRoel Kluin2008-04-241-4/+5
| | | | | | | bps is unsigned, a negative snd_pcm_format_width() return value is not noticed Signed-off-by: Roel Kluin <12o3l@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] hda - Avoid unexpected breakage with ALC889A hackTakashi Iwai2008-04-241-1/+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>
* [ALSA] hda - Fix ALC889A codec supportTakashi Iwai2008-04-241-0/+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>
* [ALSA] hda: Add 5.1 support for second headphone jackMatthew Ranostay2008-04-241-1/+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>
* [ALSA] soc - wm9712: Remove unneeded AC97_EXTENDED_MID updatesMark Brown2008-04-241-8/+0
| | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] oxygen: generalize DAC volume TLV handlingClemens Ladisch2008-04-245-26/+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>
* [ALSA] oxygen: mute by defaultClemens Ladisch2008-04-244-18/+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>
* [ALSA] oxygen: generalize handling of DAC volume limitsClemens Ladisch2008-04-246-28/+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>
* [ALSA] hifier: remove empty hifier_mixer_init()Clemens Ladisch2008-04-241-6/+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>
* [ALSA] hda - Add support of AD1989A/AD1989BTakashi Iwai2008-04-242-4/+26
| | | | | | | | | | | 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>
* [ALSA] sound/core.h: evil #ifdefsPavel Machek2008-04-241-2/+2
| | | | | | | | snd_minor_info_oss_* is an function returning int _or_ comment, depending on config parameters. That is truly evil, fix it. Signed-off-by: Pavel Machek <pavel@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] virtuoso: fix DX front panel I/OClemens Ladisch2008-04-241-31/+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>
* [ALSA] snd_usb_caiaq: make high sample rates work with A8DJDaniel Mack2008-04-242-3/+7
| | | | | | | | This patch for snd_usb_caiaq makes sample rates higher dann 48KHz work with devices which have more than 2 stereo input/output pairs. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] snd_usb_caiaq: correct input channel orderDaniel Mack2008-04-242-2/+2
| | | | | | | | This patch corrects the input channel order of hardware supported by snd_usb_caiaq. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] snd_usb_caiaq: fix potential lockups lockingDaniel Mack2008-04-242-44/+31
| | | | | | | | | This patch fixes potential lockups in snd_usb_caiaq by refining the locking mechanims and by using usb_kill_urb() in favor to usb_unlink_urb(). Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3XJarkko Nikula2008-04-241-0/+11
| | | | | | Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] wm9713: Don't control touch screen power on suspendMark Brown2008-04-241-1/+10
| | | | | | | | | Leave the power bit for the touch screen alone when suspending the WM9713 so that the touch screen driver can handle it. This allows the touch screen to be used as a wakeup source when the system is suspended. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] sound: this amplifier only goes up to 7Nick Andrew2008-04-241-2/+2
| | | | | | | | | sound: kernel log levels are 0-7 Kernel log levels are 0-7, not 0-9. Signed-off-by: Nick Andrew <nick@nick-andrew.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] hda-intel: Add Quanta IL1 ALC267 modelHerton Ronaldo Krzesinski2008-04-242-1/+78
| | | | | | | | | | | | | | | 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>
* [ALSA] sound: fix platform driver hotplug/coldplugKay Sievers2008-04-243-0/+8
| | | | | | | | | | | | | Since 43cc71eed1250755986da4c0f9898f9a635cb3bf, the platform modalias is prefixed with "platform:". Add MODULE_ALIAS() to the hotpluggable sound platform drivers, to re-enable auto loading. [dbrownell@users.sourceforge.net: more drivers, registration fixes] Signed-off-by: Kay Sievers <kay.sievers@vrfy.org> Signed-off-by: David Brownell <dbrownell@users.sourceforge.net> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] hda: EAPD power managementMatthew Ranostay2008-04-241-6/+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>
* [ALSA] hda: Correct SPDIF out default configMatthew Ranostay2008-04-241-0/+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>
* [ALSA] hda - Fujitsu Lifebook PC speaker signalTony Vroon2008-04-241-0/+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>
* [ALSA] hda - PCI quirk for laptop LG which use CMI9880Jiang zhe2008-04-241-0/+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>
* [ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880Jiang zhe2008-04-241-1/+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>
* [ALSA] sound/usb/usbaudio.c: coding stylePavel Machek2008-04-241-31/+32
| | | | | | | | Putting space between ! and variable is a strange coding style, fix that, also make it fit into 80 columns where that is easy. Signed-off-by: Pavel Machek <pavel@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] usb audio: make quirk handling more readable, and fix commented-out codePavel Machek2008-04-241-5/+6
| | | | | | | | | | | | | usb audio contains useful debugging code, protected by #if 0. Unfortunately, it will not compile because variable names changed; fix it. Dallas workaround is formatted in a way where it is not quite obvious what is normal code and what is quirk. Reformat it to make it obvious. Signed-off-by: Pavel Machek <pavel@suse.cz> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] usb audio: Fix another Dallas quirkPavel Machek2008-04-241-1/+12
| | | | | | | | | Dallas USB speakers are buggy in more than one way. One of configs they offer does not work at all. Signed-off-by: Pavel Machek <pavel@suse.cz> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] hda-codec - Fix unbalanced mutexFrederik Deweerdt2008-04-241-1/+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>
* [ALSA] es1968 - fix coding style in the last patchAndrew Morton2008-04-241-2/+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>
* [ALSA] es1968: fix jitter on some maestro cardsAndreas Mueller2008-04-241-1/+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>
* [ALSA] sound/pci/rme9652/hdspm.c: stop inlining largish static functionsDenys Vlasenko2008-04-241-7/+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>
* [ALSA] soc - Include register in DAPM debug outputMark Brown2008-04-241-1/+1
| | | | | | | | When logging register changes in DAPM debug output include the register number. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] hda-codec - PCI quirk for MSI laptopJiang zhe2008-04-241-0/+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>
* [ALSA] virtuoso: initialize two-wire control registerClemens Ladisch2008-04-241-3/+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>
* [ALSA] virtuoso: add GPIO 1 mixer controlClemens Ladisch2008-04-241-0/+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>
* [ALSA] oxygen: use SPDIF input only if presentClemens Ladisch2008-04-244-28/+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>
* [ALSA] virtuoso: correctly switch input jack on Xonar DXClemens Ladisch2008-04-243-4/+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>
* [ALSA] virtuoso: add Xonar DX supportClemens Ladisch2008-04-244-4/+365
| | | | | | | Add support for the Asus Xonar DX. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] virtuoso: fix typoClemens Ladisch2008-04-241-1/+1
| | | | | | | Fix a (fortunately harmless) typo. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] virtuoso: change card short nameClemens Ladisch2008-04-241-2/+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>
* [ALSA] virtuoso: set PCM1796 oversampling rateClemens Ladisch2008-04-241-2/+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>
* [ALSA] virtuoso: move some code to xonar_common_init()Clemens Ladisch2008-04-241-26/+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>
* [ALSA] virtuoso: allow both CS5381 and CS5361Clemens Ladisch2008-04-241-14/+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>
* [ALSA] virtuoso: separate D2/D2X init functionsClemens Ladisch2008-04-241-44/+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>
* [ALSA] oxygen: add I2C supportClemens Ladisch2008-04-242-2/+22
| | | | | | | Add a function to write I2C registers. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* [ALSA] aw2: remove duplicate MODULE_LICENSEClemens Ladisch2008-04-241-1/+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>
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-05 10:29:37 +0000