OSS: dmabuf: fix negative DMAbuf_get_buffer_pointer() check

Since unsigned active_offs < 0 is even true when DMAbuf_get_buffer_pointer()
returns negative

Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] SOC: fix tests in cs4270_hw_params()

cs4270_hw_params does several times:

ret = snd_soc_write()
if (ret < 0)
...

This only works when ret is signed.

Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] usb-audio - Fix race in reconnection

Fix the race at reconnection of the device.
The disconnected usb_chip[] must be cleared before the next probe
call properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] Clean up snd_card_free*()

A little clean up of snd_card_free*().
Removed snd_card_free_prepare() since it's actually almost identical
with snd_card_disconnect().

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] Fix the race of card instance unregistration

Move the call of device_unregister() for the card instance in
snd_card_disconnect() to avoid the race of sysfs card entry, which
can be typically found on usb-audio reconnection.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] snd-powermac: style burgundy.c

Coding style corrections for burgundy.c.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] snd-powermac: Burgundy mixers for B&W and iMac

Add mixer controls and correct headphone detection bits for PowerMac
G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] snd-powermac: style awacs.s and awacs.h

Coding style corrections for awacs.c and awacs.h.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SL

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.c

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 detection

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() check

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 hack

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 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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[ALSA] sound: fix platform driver hotplug/coldplug

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 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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[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>

[ALSA] ymfpci - Fix race at removal

free_irq() must be called first to avoid races at removal.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Add missing models in ALSA-Configuration.txt

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Use common 3stack-6ch mixer for 3stack-hp model

Forgot one more: 3stack-hp model also have now the same mixer as
3stack-6ch (after DAC assignment fix in ALC883), so use it avoiding
duplicating the same mixer definition.

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

[ALSA] hda-codec - Use base ALC883 mixer for 6stack-dell model

After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the
same as alc883_base_mixer. Avoid duplicated code and use
alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer
definition.

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

[ALSA] hda-codec - Remove now uneeded 6stack-hp model from ALC883

After DAC assignment fix in ALC883, the 6stack-hp model is now the same
as 6stack-dig. So just remove 6stack-hp model and replace its use with
6stack-dig.

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

[ALSA] hda-codec - model for alc262 to support Lenovo 3000

This model is to support the Lenovo 3000 y410.
ALSA bug#3856:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3856

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

[ALSA] hda: 92hd71bxxx DMIC nid

Added missing DMIC verb to dell_4_1_pin_configs[].

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

[ALSA] ice1724 - Improved the Juli rate setting

* moving most of clock-specific code to card-specific routines
* support for ESI Juli
* to-be-researched - monitoring of analog/digital inputs

Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] sound/pci/pcxhr/pcxhr.c: fix warnings

sparc64:

sound/pci/pcxhr/pcxhr.c: In function `pcxhr_update_r_buffer':
sound/pci/pcxhr/pcxhr.c:459: warning: cast to pointer from integer of different size
sound/pci/pcxhr/pcxhr.c: In function `pcxhr_trigger_tasklet':
sound/pci/pcxhr/pcxhr.c:628: warning: long int format, different type arg (arg 4)

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

[ALSA] sound/pci/pcxhr/pcxhr_core.c: fix printk warning

sound/pci/pcxhr/pcxhr_core.c: In function `pcxhr_set_pipe_state':
sound/pci/pcxhr/pcxhr_core.c:899: warning: long int format, different type arg (arg 4)

suseconds_t is int on sparc64.

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

[ALSA] sound/pci/aw2/aw2-alsa.c needs dma-mapping.h

sparc32:

sound/pci/aw2/aw2-alsa.c: In function 'snd_aw2_create':
sound/pci/aw2/aw2-alsa.c:282: error: 'DMA_32BIT_MASK' undeclared (first use in this function)
sound/pci/aw2/aw2-alsa.c:282: error: (Each undeclared identifier is reported only once
sound/pci/aw2/aw2-alsa.c:282: error: for each function it appears in.)

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

[ALSA] oxygen: disable clock of unused I2S inputs

Disable the master clock outputs of any unused I2S inputs.

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

[ALSA] oxygen: move MIDI flag to model struct

Put the flag that enables the MIDI port into the model structure instead
of passing it as a separate parameter to oxygen_pci_probe().

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

[ALSA] oxygen: make SPI/2-wire configuration model-specific

Allow the model drivers to specify if the codec communication goes over
SPI or a 2-wire bus.

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

[ALSA] oxygen: change model-specific PCM device configuration

When specifying which PCM devices to use, model drivers now use flags
that also specify the routing between PCM devices and DMA channels
instead of just DMA channel bits.  This simplifies some code that checks
for these flags.

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

[ALSA] oxygen: add monitor controls

Add controls to enable monitoring of the analog and digital inputs.

To allow monitoring after loading the driver when nothing has been
played back or recorded yet, the I2S input and outputs are initialized
to a valid configuration.

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

[ALSA] virtuoso: move PCM1796 symbols to a header file

Move the PCM1796 register symbol definitions to their own header file.

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

[ALSA] oxygen: move WM8785 symbols to a header file

Move the WM8786 register symbol definitions to their own header file.

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

[ALSA] Removed deprecated sound/driver.h from Freescale MPC8610 drivers

With commit 9004acc70e8c49c50c4c7b652f906f1e0ed5709d, include/sound/driver.h
is deprecated.  This patch removes the #include from fsl_ssi.c and fsl_dma.c.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-intel - Add sync support

Addded the support of sync streams to hda-intel driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Support of Lenovo Thinkpad X300

Added the model thinkpad for Lenovo Thinkpad X300 with AD1984A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] soc - Add missing audio path between Mono Mixer and Mic PGAs

Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - keep the format verb at closing PCM streams

Keep the format verb at closing PCM streams.
Introduced snd_hda_codec_cleanup_stream() for the parcicular purpose.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Fix spekaer output of Panasonic CF-74

Add a new model "panasonic" for Panasonic CF-74 with STAC9200 codec
to fix the speaker output.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-intel - Add barrier

Add proper barriers in the RIRB communication code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267)

Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267).

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

[ALSA] hda-codec - Fix DAC assignment order in ALC883

Actually clfe and surround DACs are inverted in alc883_dac_nids array
(see ALC883 datasheet). I discovered this while testing multi-channel
setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that
has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped
in 6 channel mode (also in 4 channel mode you didn't get rear left/right
output). Other models also were affected by this bug, as can be seen by
the mixer layouts that "workaround" this (the real bug was not noticed,
and some other models simply played with mixer and initial verbs). Thus
along with fixing the order of dac nids, also change the models that
relied on previous dac ordering properly.

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

[ALSA] sound/drivers/pcsp/pcsp.c build fix

sound/drivers/pcsp/pcsp.c: In function 'snd_pcsp_create':
sound/drivers/pcsp/pcsp.c:54: error: 'loops_per_jiffy' undeclared (first use in\ this function)
sound/drivers/pcsp/pcsp.c:54: error: (Each undeclared identifier is reported on\ ly once
sound/drivers/pcsp/pcsp.c:54: error: for each function it appears in.)

Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] at73c213: Add constraints for periods value

The interrupt handler always provide runtime->period_size data, so it
works correctly only if buffer_size was a multiple of period_size.

This patch fixes periodic click noise.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] sound/pci: remove unused variable

The variable is_capture is initialized but never used otherwise.

The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@
type T;
identifier i;
constant C;
@@

(
extern T i;
|
- T i;
  <+... when != i
- i = C;
  ...+>
)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] ice1724 - Fix return codes in some pointis callbacks

Fixed the return codes (1 for changed values) in put callbacks of
pontis.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] usb-audio - Add a proper error check

The error in check_hw_params_convention() has to be checked and
handled properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] some fixes and cleanup for ICE1724 cards

* removing the hack with NON_AKM ak4xxx type
* support for card-specific flags in ak4114_stats
* definition of the flags for corresponding cards

Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] [ML403-AC97CR] Remove duplicate snd_card_set_dev()

We want to have snd_card_set_dev() in _probe(), but not a second one in
snd_ml403_ac97cr_create().

Signed-off-by: Joachim Foerster <JOFT@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] ice1724 - Fix the SPDIF input sample-rate on Juli@

AK4114 on Juli@ has the SPDIF input sample rate detection and
causes errors when an incompatible sample rate is chosen.
The patch adds the open hook to check the current rate and limit
the hw constraints.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Fujitsu Lifebook port replicator/dock headphone jack sense

The docking station headphone output had no audio and jack sense
was not considered.

Jack information from the laptop itself and the dock are combined, as
the dock does not obscure the connector.

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

[ALSA] hda-intel - Fix power-off hang on ASUS P5AD2

The hda-intel driver has a problem at power-off on ASUS P5AD2.
It's caused when the position-buffer is enabled -- most likely a
hardware-specific problem.

This patch adds a quirk to avoid the unnecessary enablement of
position-buffer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>