- 24 Apr, 2008 40 commits
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Takashi Iwai authored
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>
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Matthew Ranostay authored
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:
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Mark Brown authored
Signed-off-by:
Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by:
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Clemens Ladisch authored
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:
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Clemens Ladisch authored
Initialize the playback volume controls as being muted and having minimal volume. Signed-off-by:
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Clemens Ladisch authored
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:
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Clemens Ladisch authored
The empty hifier_mixer_init() function is useless; remove it. Signed-off-by:
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Takashi Iwai authored
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: -
Pavel Machek authored
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:
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Clemens Ladisch authored
Fix the GPIO 1 mixer control to enable I/O through the front panel connector of the Xonar DX. Signed-off-by:
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Daniel Mack authored
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:
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Daniel Mack authored
This patch corrects the input channel order of hardware supported by snd_usb_caiaq. Signed-off-by:
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Daniel Mack authored
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:
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Jarkko Nikula authored
Signed-off-by:
Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by:
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Mark Brown authored
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:
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Nick Andrew authored
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:
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Herton Ronaldo Krzesinski authored
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:
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Kay Sievers authored
Since 43cc71ee, 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:
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Matthew Ranostay authored
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:
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Matthew Ranostay authored
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:
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Tony Vroon authored
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:
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Jiang zhe authored
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:
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Jiang zhe authored
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:
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Pavel Machek authored
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:
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Pavel Machek authored
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:
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Pavel Machek authored
Dallas USB speakers are buggy in more than one way. One of configs they offer does not work at all. Signed-off-by:
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Frederik Deweerdt authored
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 authored
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:
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Andreas Mueller authored
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:
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Denys Vlasenko authored
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:
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Mark Brown authored
When logging register changes in DAPM debug output include the register number. Signed-off-by:
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Jiang zhe authored
Please refer to [0003848] on the alsa mantis. This patch adds the pci quirk and Mic-Int controller. Signed-off-by:
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Clemens Ladisch authored
On the Xonar DX, initialize all bits of the two-wire control register. Signed-off-by:
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Clemens Ladisch authored
Add a mixer control for switching whatever it is that is connected to GPIO pin 1 on the Xonar DX. Signed-off-by:
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Clemens Ladisch authored
If the card model does not have a digital input or an AC97 codec, disable the respective interrupt and mixer controls. Signed-off-by:
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Clemens Ladisch authored
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:
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Clemens Ladisch authored
Add support for the Asus Xonar DX. Signed-off-by:
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Clemens Ladisch authored
Fix a (fortunately harmless) typo. Signed-off-by:
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Clemens Ladisch authored
Change the card short name to show to show the card name instead of the chip name. Signed-off-by:
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Clemens Ladisch authored
When playing data at 96 kHz or higher, reduce the DAC oversampling rate to 32. Signed-off-by:
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