Commit Graph

3135 Commits

Author SHA1 Message Date
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
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