This is in preparation for the removal of struct snd_soc_device.

The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

- Add aic3x_set_headset_detection() function to define the headset
  detection mode for tlv32aic3x chips
- added aic3x_button_pressed()
- Read from the real-time registers in aic3x_headset_detected() to query
  headset presence without an occured interrupt
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>

The TWL4030 codec device has two ADCs. Both of them can have
several inputs routed to them, but TRM says that only one source
can be selected for every ADC, even though every source has a
dedicated bit in the registers.

This patch adds input source controls. It modifies default register
values to have no inputs selected and ADCs disabled. When some
input is selected, control handlers enable apropriate input
amplifier and ADC. If a microphone is selected, bias power is
automatically enabled. When some input is deselected, unused
chip parts are disabled.

Microphone and line input recording tested on OMAP3 pandora board.
Signed-off-by: NGrazvydas Ignotas <notasas@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card.  Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

All outputs have dedicated gain controls except the
HandsFree output.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add Playback volume controls for all four DACs.
All four paths has three levels of volume controls:
Digital Fine gain, Digital Coarse gain, Analog gain.

The controls are named to reflect their connection to the DACs.
Per DAC volume can be performed, if needed:
amixer sset 'DAC1 Analog'  5,10
DACL1 analog gain to 5
DACR1 analog gain to 10
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The digital Capture gain control has a range:
0 to 31 dB in 1 dB steps.
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>

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

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>