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: NTakashi Iwai <tiwai@suse.de>

snd_minor_info_oss_* is an function returning int _or_ comment,
depending on config parameters. That is truly evil, fix it.
Signed-off-by: NPavel Machek <pavel@suse.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the GPIO 1 mixer control to enable I/O through the front panel
connector of the Xonar DX.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch corrects the input channel order of hardware supported by
snd_usb_caiaq.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

sound: kernel log levels are 0-7

Kernel log levels are 0-7, not 0-9.
Signed-off-by: NNick Andrew <nick@nick-andrew.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NHerton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NKay Sievers <kay.sievers@vrfy.org>
Signed-off-by: NDavid Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMatthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMatthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTony Vroon <tony@linx.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Please refer to [0003874] on the alsa mantis.
This patch added the pci quirk.
Signed-off-by: NJiang zhe <zhe.jiang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NJiang zhe <zhe.jiang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NPavel Machek <pavel@suse.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NPavel Machek <pavel@suse.cz>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Dallas USB speakers are buggy in more than one way. One of configs
they offer does not work at all.
Signed-off-by: NPavel Machek <pavel@suse.cz>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NRene Herman <rene.herman@keyaccess.nl>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDenys Vlasenko <vda.linux@googlemail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When logging register changes in DAPM debug output include the register
number.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Please refer to [0003848] on the alsa mantis.
This patch adds the pci quirk and Mic-Int controller.
Signed-off-by: NJiang zhe <zhe.jiang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On the Xonar DX, initialize all bits of the two-wire control register.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a mixer control for switching whatever it is that is connected to
GPIO pin 1 on the Xonar DX.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If the card model does not have a digital input or an AC97 codec,
disable the respective interrupt and mixer controls.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add support for the Asus Xonar DX.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix a (fortunately harmless) typo.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Change the card short name to show to show the card name instead of the
chip name.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When playing data at 96 kHz or higher, reduce the DAC oversampling rate
to 32.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Rename all CS5381 symbols to CS53x1 because they can also be used for
Xonar models with a CS5361.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a function to write I2C registers.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

"GPL 2" does not mean that there have to be two MODULE_LICENSE("GPL")
entries.  ;-)
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Two sentences seem to be spliced into one in comment, fix that and fix
english. Also fix codingstyle.
Signed-off-by: NPavel Machek <pavel@suse.cz>
Signed-off-by: NJaroslav Kysela <perex@perex.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>