[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>

[ALSA] hda-codec - Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX

Map 3stack-6ch-dig ALC662 model for Asus P5GC-MX.

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

[ALSA] hda-codec - Fix ALC662 DAC mixer mutes

Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03,
0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to
mute?). The result is that currently mute for "Front Playback Switch",
"Surround Playback Switch", "Center Playback Switch" and "LFE Playback
Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard
with 3stack-6ch model).

The solution I found for this is to mute the proper inputs in 0x0c,
0x0d, 0x0e audio mixers.

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

[ALSA] hda-codec - Fix orphan Headphone controls in STAC codecs

Currently, the headphone controls are created as Master wrongly in
some cases, and this prevents the virtual master controls.
The patch fixes the problem by simply using "Headphone" always for
headphone controls.

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

[ALSA] hda: 92HD73xxx distortion fix

Fixed issue on some laptops that if the Master mixer and DAC mixers are
turned all the way up that will cause distortion. This is fixed by limiting
the max volume with the volume knob nid.

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

[ALSA] pcsp: locking fix

pcsp: locking fix.

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Improve ALC262 ultra model

Improved ALC262 ultra model for Samsung Q1 Ultra series.

- clean up mixers
- support of input from HP jack as a mic
- add quirk for Q1 EL

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

[ALSA] at73c213: remove redundant private_free routine

snd_pcm_lib_preallocate_free_for_all() is called from snd_pcm_free() just
after calling the private_free routine.  So there should be no need to call
it in driver's private_free routine.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] aw2 - Rename aw2-tsl.h to aw2-tsl.c

aw2-tsl.h should be rather a C file to be included since it's referred
only in aw2-saa6146.c and includes a table data.

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

[ALSA] hda-intel - Fix microphone capture with ALC880 F1734 model

The default capture source should be the mic which is 0x01 on this model.
In addition to that the change to VREF50 allows for higher capture volume.

Signed-off-by: Michael Gruber <lists.mg@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda: Reorganized DAC outputs

Changed so that internal speakers point to the Front mixer instead of Surround.

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

[ALSA] release 1.0.16

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

[ALSA] at73c213: monaural support

Add support for monaural playback to at73c213 driver.  The sound will be apear
on L-channel.  Tested on AT91SAM9260-EK.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] at73c213: fix error checking for clk API

The clk_round_rate() and clk_set_rate() will return int, so not store thier
return value to unsigned long variable.  This bug hides real error on these
API.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] HDA Codecs: add support for Toshiba Equium L30

This patch adds support for the Toshiba Equium L30 laptop and renames the mixer
controls to match Laptop usages.

Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound: Use BUG_ON

if (...) BUG(); should be replaced with BUG_ON(...) when the test has no
side-effects to allow a definition of BUG_ON that drops the code completely.

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

// <smpl>
@ disable unlikely @ expression E,f; @@

(
  if (<... f(...) ...>) { BUG(); }
|
- if (unlikely(E)) { BUG(); }
+ BUG_ON(E);
)

@@ expression E,f; @@

(
  if (<... f(...) ...>) { BUG(); }
|
- if (E) { BUG(); }
+ BUG_ON(E);
)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>

[ALSA] hda-codec - Add internal mic item for ALC268 acer model

Added the internal mic as a capture source item for ALC268 acer model.

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

[ALSA] hda-codec - Fix dmics on ALC268 in auto configuration

Fixed the handling of dmics on ALC268 in the auto-configuration mode.

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

[ALSA] hda_intel: Add the DIDs of nvidia MCP79 HD audio controller to hda_intel.c

Add the Device IDs of nvidia MCP79 HD audio controller to hda_intel.c

Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - model for cx20549 to support laptop HP530

Currently the model laptop-hpsense use the 0x12 as ExtMic,
and use 0x14 as Internal IntMic.
But the hp530 only have one ExtMic, the Pin widget is 0x14.

In this patch, I changed the mixer item for them.
I still reserved the IntMic item, it will be helpful if
other machine may use this model.

ALSA bug#3821.

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

[ALSA] hda-codec - model for alc883 to support FUJITSU Pi2515

There is no suitable model for Pi2515.
This model is to support it.  ALSA bug#3800.

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

[ALSA] pcsp: add description

update ALSA-Configuration.txt

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] pcsp: improve "enable" option handling

Simplify init code.

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] ALC288 - Add NEC S970 to the quirk table

NEC S970 has no sound in the internal speakers when autodetection is
used.
With targa-dig model, there is sound in the speakers and it gets
correctly muted when pluging headphones.

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

[ALSA] pcsp - clean ups

- make pcsp_start_timer_tasklet static
- remove redundant includes. <asm/i8253.h> is not available on all platforms.

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - model for alc883 to support M720R

There is no suitable model for M720R (ALSA bug#3781).
This patch is to support HP jack-sensing and mixer.

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

[ALSA] aw2 - Remove endian dependency

Removed unnecessary dependency on the little-endianess.

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

[ALSA] sound: replace remaining __FUNCTION__ occurences

__FUNCTION__ is gcc-specific, use __func__

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda_intel needs dma-mapping.h

sparc32:

sound/pci/hda/hda_intel.c: In function 'azx_create':
sound/pci/hda/hda_intel.c:1838: error: 'DMA_64BIT_MASK' undeclared (first use in this function)
sound/pci/hda/hda_intel.c:1838: error: (Each undeclared identifier is reported only once
sound/pci/hda/hda_intel.c:1838: 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] soc - Add Invert Switch for ROUT2

GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2
needs to be inverted. This patch adds a mixer control for this.

Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] Add PC-speaker sound driver

Added PC-speaker sound driver (snd-pcsp).

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Fix the array over-range access with stac92hd71bxx codec

Add the check of the array range for dac_nids to prevent the over-range
access.

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

[ALSA] IEC958 definitions for consumer status channel, byte 4

Added definition for byte 4 of SPDIF channel status, according to
second edition of IEC 60958-3 (consumer) spec.

Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda: add verbs for 92hd73xxx laptops

Added core_init[] for several 92hd73xxx laptops.

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

[ALSA] hda: disable power management on fixed ports

Power management can't be enabled on fixed ports, since the presence
will always return false and prevent output.

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

[ALSA] sound: hda: missing includes of hda_patch.h

Move the array declaration to hda_codec.c where it is used and add includes
where the individual presets are declared.

Fixes the following sparse warnings:
sound/pci/hda/patch_realtek.c:13744:25: warning: symbol 'snd_hda_preset_realtek' was not declared. Should it be static?
sound/pci/hda/patch_cmedia.c:729:25: warning: symbol 'snd_hda_preset_cmedia' was not declared. Should it be static?
sound/pci/hda/patch_analog.c:3656:25: warning: symbol 'snd_hda_preset_analog' was not declared. Should it be static?
sound/pci/hda/patch_sigmatel.c:3995:25: warning: symbol 'snd_hda_preset_sigmatel' was not declared. Should it be static?
sound/pci/hda/patch_si3054.c:286:25: warning: symbol 'snd_hda_preset_si3054' was not declared. Should it be static?
sound/pci/hda/patch_atihdmi.c:156:25: warning: symbol 'snd_hda_preset_atihdmi' was not declared. Should it be static?
sound/pci/hda/patch_conexant.c:1721:25: warning: symbol 'snd_hda_preset_conexant' was not declared. Should it be static?
sound/pci/hda/patch_via.c:1962:25: warning: symbol 'snd_hda_preset_via' was not declared. Should it be static?

Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

[ALSA] hda-codec - Use int instead of long in patch_sigmatel.c

The HD-audio parameters are at most 32bit int.

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