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>

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>

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>

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>