The definitions of S3C2412_IISMOD_SDF_MSB and S3C2412_IISMOD_SDF_LSB
are incorrect, being the same S3C2412_IISMOD_SDF_IIS which is the
only correct one in this series.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.30' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Remove a non-existing Kconfig CONFIG_SND_SOC_WM8750_SPI.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

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

Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add support for true pause and unpause.  Without this, mplayer will drop some
audio (less than one second, but still noticeable) when pausing playback.

Remove support for PM suspend and resume from the trigger function, since the
driver doesn't support PM anyway.

Optimize the delay after starting capture.  Instead of delaying 1ms, the driver
now polls the hardware.  The new delay is shorter by over 90% yet still
effective.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Upgrade the severity of some failure messages from debug level so
they're displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The recent set of S3C64xx patches re-added a lot of uses of DBG() that
had previously been removed - revert this so the standard pr_debug()
macro is used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

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

Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add headset jack detection for SDP3430 boards using SoC jack
reporting interface. Headset detection on SDP3430 board is
achieved through TWL4030 GPIO_2 pin.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add a new device tree property for the SSI node: "fsl,ssi-asynchronous".  If
defined, the SSI is programmed into asynchronous mode, otherwise it is
programmed into synchronous mode.  In asynchronous mode, pin SRCK must be
connected to the same clock source as STFS, and pin SRFS must be connected to
the same signal as STFS.  Asynchronous mode allows playback and capture to
use different sample sizes.  It also technically allows different sample rates,
but the driver does not support that.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

We now support the 64xx series as well as the 24xx series - make sure
people using Kconfig know this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch
is a special case of a mixer with only one input) but this wasn't
correctly handled in the code.

Also fix the coding style for the switch below while we're here.
Reported-by: Joonyoung Shim <dofmind@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add two more bitfields for the PSP register. As they seem to exist
for PXA3xx only, define them conditionally.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Essentially simple code motion to facilitate refactoring of the power
decisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

A bit in PXA's SSCR0 register was erroneously named ADC but its name is
in fact ACS (audio clock select).
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Enum type for selecting the desired ramp delay for the headset output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Remove uneeded startup callback and use snd_soc_dapm_nc_pin()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Select the relevant DMA implementation when the
sound driver is selected.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add the initial code to support the S3C64XX I2S hardware using the
s3c-i2s-v2 core code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.

As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add support for the Jive's WM8750 codec attached via the S3C2412 IIS.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add DAPM machine domain widgets to SDP3430 machine driver.
Interconnection:
* Ext Mic: MAINMIC, SUBMIC
* Ext Spk: HFL, HFR
* Headset Jack: HSMIC, HSOL, HSOR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The <mach/audio.h> file needs to be common to both ARCH_S3C2410 and
ARCH_S3C64XX as they share common driver code, so move it to <plat/audio.h>.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Move the IIS headers to their correct place.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

When setting WM8510_MCLKDIV the pll was turned off.

When setting pll frequency you got twice the expected freq, because
the  code calculated  with postscaler of 8,  but  the hardware divide by 4.
Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.

Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.

All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).

* Queue work in the alsa PCM_START .trigger to flush registers
  as soon as the link is running. This replaces the .prepare
  and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
  its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

This removes a misspelled comment and got rid of superfluous switch
case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>