Currently ASoC card initialisation is completed by a function called
snd_soc_register_card().  As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix the old-style trigger callback in s3c2443-ac97.c:
    sound/soc/s3c24xx/s3c2443-ac97.c:378: warning: initialization from incompatible pointer type
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the wrong shutdown callback type.  Also removed the unused variables
there:

  sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
  sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
  sound/soc/pxa/corgi.c: At top level:
  sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit 9171e5e6.

I can't reproduce the compile warnings any more.  The warnings
might be some weird cross-compiling set up.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The dependency on SND_SOC is already fulfilled in sound/soc/Kconfig,
thus no more need in Kconfig of each sub directory.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit e669dae6, since it
is incomplete, and clashes with fuller patches and the sparc 32/64
unification effort.
Requested-by: NDavid Miller <davem@davemloft.net>
Acked-by: NAl Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

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

Hide annoying uninitialized warnings:
  sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function
  sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

switch to __init for those; unlike powerpc sparc has no hotplug support
for that stuff and their ->probe() tends to call __init functions while
being declared __devinit.
Signed-off-by: NAl Viro <viro@zeniv.linux.org.uk>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

This patch enables more routing functions for tlv320aic3x codecs.
It is now possible to

 - control the volume of the PGA bypass path for the HPL, HPR, HPLCOM
   and HPRCOM outputs individually
 - route right line1 input to the left ADC channel
 - route left line1 input to the right ADC channel
 - route right mic3 input to left DAC channel
 - route left mic3 input to right DAC channel
 - route left line1 input to right line1 output
 - route right line1 input to left line1 output
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There is no argument named @clk_id in snd_soc_dai_set_fmt,
remove its' comment.
Signed-off-by: NQinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NDmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch add ASoC support for TI SDP3430. It's based on Gumstix
Overo SoC code by Steve Sakoman.
Signed-off-by: NMisael Lopez Cruz <mesak82@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fixes Kconfig dependency of TWL4030 audio codec driver
with TWL4030 core driver on both overo and omap2evm
boards
Signed-off-by: NArun KS <arunks@mistralsolutions.com>
Acked-by: NDavid Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Patch adds support for mono audio links so that McBSP DAI can operate with
real mono codecs. In I2S, the signalling remains the same but only first
frame (left channel) is transmitting audio data and second frame having null
data. In DSP_A, only first frame is transmitted.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Prepare for upcoming McBSP DAI update adding support for mono links by
restricting number of channels to 2 in N810. This is due tlv320aic3x which
claims channels_min = 1 and playing pure mono audio over I2S would cause
it to be played only from left channel if both cpu and codec DAI's claim to
support mono.
Signed-off-by: NJarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Check the model type instead of PCI SSID for detection of the mic types
on Dell laptops with IDT 92HD73xx codecs.  In this way, a new laptop
can be tested via model module option.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine.  For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The AFG pin power-mapping isn't properly set for the fixed I/O pins
on IDT 92HD* codecs.  This resulted in the low power mode after the
boot until any jack detection is executed, thus no output from the
speaker.

This patch fixes the power mapping for the fixed pins, and also fixes
the GPIO bits and digital I/O pin settings properly in stac92xx_ini().

Reference: Novell bnc#446025
	https://bugzilla.novell.com/show_bug.cgi?id=446025Signed-off-by: NTakashi Iwai <tiwai@suse.de>

SPDIF status bits controls are written via snd_hda_codec_write()
without caching.  This causes a regression at resume that the bits
are lost.

Simply replacing it with the cached version fixes the problem.

Reference:
	http://lkml.org/lkml/2008/11/24/324Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Now that the ASoC resume has been punted to a workqueue for a release
cycle without attracting bug reports it should be safe to make the
log messages associated with it debug level, reducing noise and kernel
size in production configurations.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Special handling is required for suspend and resume of AC97 codecs
due to the control path going over the data bus.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Some of the gain controls in TWL (mostly those which are associated with
the outputs) are implemented in an interesting way:
 0x0 : Power down (mute)
 0x1 : 6dB
 0x2 : 0 dB
 0x3 : -6 dB
Inverting not going to help with these.
Custom volsw and volsw_2r get/put functions to handle these gains.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add CGAIN (Coarse gain control) to TWL4030 codec.
The range of the CGAIN is:
0 dB to 12 dB in 6 dB steps.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

TWL4030 FGAIN volume control has a range:
-62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Keep Soft-volume disabled for now, since if it is enabled
the FGAIN volume controls are not working in the current
configuration:
CODEC_MODE:OPT_MODE = 1
OPTION:ARXR2_EN = 1
OPTION:ARXL2_EN = 1
OPTION:ARXR1_EN = 0
OPTION:ARXL1_VRX_EN = 0
RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1)
RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1)

After the patch, FGAIN volume control works.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Print something a bit more verbose to help make errors a little more
obvious.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

It's not exported.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.

The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.

This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKarl Beldan <karl.beldan@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Clean up our record of the active streams in shutdown(), fixing
subsequent failures of snd_pcm_hw_constraints_complete after closure of
a stream.

NOTE:
- The ssm2602 allows pairs of non-matching PB/REC rates.
- This is a fix for less evil:
  The logic is flawed (e.g. the slave might startup before the
  master's rate and sample_bits are set).
Signed-off-by: NKarl Beldan <karl.beldan@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.

Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Added the matching model=dell-m6 for Dell Studio 15 laptop.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added STAC_DELL_M4_3 quirk for Dell systems, also reorganized the
board config switch to assign number of digital muxes, microphones,
and SPDIF muxes via the PCI quirk defined.
Signed-off-by: NMatthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>