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Commit Graph

92164 Commits

Author SHA1 Message Date
Takashi Iwai
cb308f97ae [ALSA] hda - Fix ALC889A codec support
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 12:00:37 +02:00
Matthew Ranostay
7c2ba97b8a [ALSA] hda: Add 5.1 support for second headphone jack
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 12:00:37 +02:00
Mark Brown
0a08478c0f [ALSA] soc - wm9712: Remove unneeded AC97_EXTENDED_MID updates
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Clemens Ladisch
4972a177fe [ALSA] oxygen: generalize DAC volume TLV handling
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 12:00:37 +02:00
Clemens Ladisch
e983532e44 [ALSA] oxygen: mute by default
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 12:00:37 +02:00
Clemens Ladisch
193e813814 [ALSA] oxygen: generalize handling of DAC volume limits
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 12:00:36 +02:00
Clemens Ladisch
0c0e6daf14 [ALSA] hifier: remove empty hifier_mixer_init()
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 12:00:36 +02:00
Takashi Iwai
3adb8abc70 [ALSA] hda - Add support of AD1989A/AD1989B
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 12:00:36 +02:00
Pavel Machek
ad5fada51d [ALSA] sound/core.h: evil #ifdefs
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>
2008-04-24 12:00:36 +02:00
Clemens Ladisch
a8bb1bad9b [ALSA] virtuoso: fix DX front panel I/O
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 12:00:36 +02:00
Daniel Mack
6e9fc6bd5d [ALSA] snd_usb_caiaq: make high sample rates work with A8DJ
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>
2008-04-24 12:00:36 +02:00
Daniel Mack
6849d49c48 [ALSA] snd_usb_caiaq: correct input channel order
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>
2008-04-24 12:00:36 +02:00
Daniel Mack
8d048841e8 [ALSA] snd_usb_caiaq: fix potential lockups locking
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>
2008-04-24 12:00:35 +02:00
Jarkko Nikula
f57ab97e76 [ALSA] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X
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>
2008-04-24 12:00:35 +02:00
Mark Brown
87b57fe2d3 [ALSA] wm9713: Don't control touch screen power on suspend
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>
2008-04-24 12:00:35 +02:00
Nick Andrew
a295e09e89 [ALSA] sound: this amplifier only goes up to 7
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>
2008-04-24 12:00:35 +02:00
Herton Ronaldo Krzesinski
eb5a662166 [ALSA] hda-intel: Add Quanta IL1 ALC267 model
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 12:00:35 +02:00
Kay Sievers
8b45a20993 [ALSA] sound: fix platform driver hotplug/coldplug
Since 43cc71eed1, 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>
2008-04-24 12:00:35 +02:00
Matthew Ranostay
0fc9dec46f [ALSA] hda: EAPD power management
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 12:00:35 +02:00
Matthew Ranostay
780c8be4ab [ALSA] hda: Correct SPDIF out default config
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 12:00:34 +02:00
Tony Vroon
06a9c30cdd [ALSA] hda - Fujitsu Lifebook PC speaker signal
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 12:00:34 +02:00
Jiang zhe
5b030389e4 [ALSA] hda - PCI quirk for laptop LG which use CMI9880
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 12:00:34 +02:00
Jiang zhe
64654c2f9e [ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880
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 12:00:34 +02:00
Pavel Machek
07f51a7274 [ALSA] sound/usb/usbaudio.c: coding style
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>
2008-04-24 12:00:34 +02:00
Pavel Machek
2a56f51bcc [ALSA] usb audio: make quirk handling more readable, and fix commented-out code
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>
2008-04-24 12:00:34 +02:00
Pavel Machek
b9d43bcd06 [ALSA] usb audio: Fix another Dallas quirk
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>
2008-04-24 12:00:33 +02:00
Frederik Deweerdt
eaa9985b4e [ALSA] hda-codec - Fix unbalanced mutex
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 12:00:33 +02:00
Andrew Morton
66c9aa6043 [ALSA] es1968 - fix coding style in the last patch
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 12:00:33 +02:00
Andreas Mueller
f24bfa53da [ALSA] es1968: fix jitter on some maestro cards
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 12:00:33 +02:00
Denys Vlasenko
62cef8212f [ALSA] sound/pci/rme9652/hdspm.c: stop inlining largish static functions
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 12:00:33 +02:00
Mark Brown
32f4876e62 [ALSA] soc - Include register in DAPM debug output
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>
2008-04-24 12:00:33 +02:00
Jiang zhe
4383fae0ec [ALSA] hda-codec - PCI quirk for MSI laptop
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 12:00:33 +02:00
Clemens Ladisch
80060ecc45 [ALSA] virtuoso: initialize two-wire control register
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 12:00:32 +02:00
Clemens Ladisch
387fb6a206 [ALSA] virtuoso: add GPIO 1 mixer control
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 12:00:32 +02:00
Clemens Ladisch
1d98c7d4be [ALSA] oxygen: use SPDIF input only if present
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 12:00:32 +02:00
Clemens Ladisch
11864b4b84 [ALSA] virtuoso: correctly switch input jack on Xonar DX
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 12:00:32 +02:00
Clemens Ladisch
a9d3cc485e [ALSA] virtuoso: add Xonar DX support
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 12:00:32 +02:00
Clemens Ladisch
80647ee26e [ALSA] virtuoso: fix typo
Fix a (fortunately harmless) typo.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:32 +02:00
Clemens Ladisch
aef1a535c4 [ALSA] virtuoso: change card short name
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 12:00:32 +02:00
Clemens Ladisch
d08267a9df [ALSA] virtuoso: set PCM1796 oversampling rate
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 12:00:31 +02:00
Clemens Ladisch
af9af1741f [ALSA] virtuoso: move some code to xonar_common_init()
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 12:00:31 +02:00
Clemens Ladisch
a694a6a0e4 [ALSA] virtuoso: allow both CS5381 and CS5361
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 12:00:31 +02:00
Clemens Ladisch
271ebfca58 [ALSA] virtuoso: separate D2/D2X init functions
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 12:00:31 +02:00
Clemens Ladisch
10e6d5f9b6 [ALSA] oxygen: add I2C support
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 12:00:31 +02:00
Clemens Ladisch
7a43567472 [ALSA] aw2: remove duplicate MODULE_LICENSE
"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 12:00:31 +02:00
Pavel Machek
edab938e63 [ALSA] fix comments in sound/core.h
Two sentences seem to be spliced into one in comment, fix that and fix
english. Also fix codingstyle.

Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:31 +02:00
Clemens Ladisch
e97f79994a [ALSA] oxygen: fix line-in recording selection (now for real)
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>
2008-04-24 12:00:30 +02:00
Herton Ronaldo Krzesinski
0c4cc4430f [ALSA] hda-codec - Support mic automute for Clevo M720R/SR
Add support for mic automute in clevo-m720r ALC883 model, and rename it
to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt
accordingly.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:30 +02:00
Herton Ronaldo Krzesinski
213f0bfe90 [ALSA] hda-codec - Map clevo-m720r ALC883 model for Clevo M720SR
Map clevo-m720r ALC883 model for Clevo M720SR.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:30 +02:00
Stas Sergeev
c81d80cbf6 [ALSA] pcsp: remove downsampling
pcsp: remove S16->U8 downsampling as dmix now supports U8 natively.

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:30 +02:00