Put parentheses around macro argument uses. This avoids pitfalls
for the programmer, where the argument expansion does not give the
expected result, for example:

SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1);
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@ti.com>

To get correct calibration, we can decrease the time
needed for the OSC to calibrate itself.
With this change we can save ~15ms in the OSC
calibration phase.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Fixes generated by 'codespell' and manually reviewed.
Signed-off-by: NLucas De Marchi <lucas.demarchi@profusion.mobi>

Move the codec power on (in reg 0x01, bit 4) from
set_bias_level:SND_SOC_BIAS_ON to a DAPM supply.
In this way we can be sure, that all the things within
the codec is powered before the external amp is
going to be enabled.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Register 0x40, 0x41 need to be restored after power up, since
it contains gain related fields, which affects playback volume.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The lock is used within the interrupt handler.
Correct the spinlock usage, and use irqsave/irqrestore
flavour of spin_lock/unlock.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The device table is required to load modules based on modaliases.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The L/R LOM line can be invertined side of the
corresponding DAC, or inverted from the corresponding
LOP.
Add control for user space to select the source of the
LOM inversion.
When only the analog bypass is enabled, and the LOM
is inverted from DAC output, we need to power the
corresponding DAC.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add support for 24 bit audio (with S32_LE msbits 24).
The reason to limit the msbits to 24, is that the FIFO
can be configured for 16 or 24 bit layout.
It is unknown how the codec would downsample from 32 to
24 bit, if the interface is configured to receive 32
bit data.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Change the structure of FIFO handling in order to
pave the way for adding 32/24 bit audio support.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The manual FIFO configuration was the first version to enable
the use of the FIFO in the codec.
It had served it's purpose as debugging aid, but the automatic
FIFO configuration is much safer to use.
The removal of the manual controls, and configuration makes
it easier to add new features for the codec later, since
the manual mode neded different ways to calculate, and
protect against misconfiguration.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

If the following scenario has been followed:
1. Enable analog bypass
amixer sset 'Analog Left Bypass' on
amixer sset 'Analog Right Bypass' on

2. Start playback
aplay -fdat -d3 /dev/zero

After the playback stopped (3 sec), and the soc timeout (5 sec),
the digital parts of the codec will remain powered up.
This means that the DAI clocks are continue to run, the
oscillator remain operational, etc.

Use the SND_SOC_DAPM_POST_PMD widget to get notification
about the stopped stream, and power down the digital
part of the codec.
If the analog bypass is enabled, than the codec will remain in
BIAS_ON level, and things will work correctly.
In case, if the bypass is disabled, than the codec will
fall to BIAS_STANDBY than to BIAS_OFF level, as it used
to.

The digital part of DAC33 is initialized at every stream start
(DAPM_PRE:PRE_PMU event), so subsequent streams (within 5 sec)
will have working DAI.
When the codec is coming out from BIAS_OFF, the full power-up
sequence followed by the same DAPM_PRE widget event will power up
the digital part.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Fix the compilation error introduced by patch:
ASoC: tlv320dac33: Avoid multiple soft power up
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The power for the DACs need to be enabled, even when only
the analog bypass is in use with the codec, otherwise
the audio is going to be distorted.
Make sure that the DACs are powered all the time, when
there is audio activity.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Use better name for the widget, and remove the 'Power'
from it's name.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

During playback start the codec has been already powered at
BIAS_ON event time, so there's no need to enable the codec again.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

No need to enable the codec at this time.
The codec will be enabled  later by other events
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

There is no need to include soc-dapm.h since soc.h includes it.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.

This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.

This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.

Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Do not allow invalid (too big) nSample value, when FIFO Mode1
and automatic fifo configuration has been selected.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Limit the time window to maximum 1s in the macro.
The driver deals with much shorter times (<200ms).
This will fix a rare division by zero bug in Mode1.
This could happen, when the work is not executed in
time (within mode1_latency) after the interrupt.
In this case the DAC33 will not receive the needed
nSample command in time, and enters to an unknown
state, and won't recover.
In such event the time window will increase, and
eventually going to be bigger than 1s, resulting
devision by zero.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Correct/Implement handling of broken chip.
Fail the soc_prope if the communication with the chip
fails (can not read chip ID).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Switch to use the more precise usleep_range instead of
msleep().
Replace the udelay with usleep_range to remove the busy loop
waiting.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Borwn <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

New control to select the line output gain.
This gain control affects the linein-to-lineout and
dac-to-loneout gain differently.
Use enum type to select the desired gain combination.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This assignment is done by the snd_soc_register_codec so there is no need
to redo it in probe function of a codec driver.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.

 struct snd_soc_codec    --->  struct snd_soc_codec (device data)
                          +->  struct snd_soc_codec_driver (driver data)

 struct snd_soc_platform --->  struct snd_soc_platform (device data)
                          +->  struct snd_soc_platform_driver (driver data)

 struct snd_soc_dai      --->  struct snd_soc_dai (device data)
                          +->  struct snd_soc_dai_driver (driver data)

 struct snd_soc_device   --->  deleted

This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.

The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.

This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.

Other notable multi-component changes:-

 * Stream operations now de-reference less structures.
 * close_delayed work() now runs on a DAI basis rather than looping all DAIs
   in a card.
 * PM suspend()/resume() operations can now handle N CODECs and Platforms
   per sound card.
 * Added soc_bind_dai_link() to bind the component devices to the sound card.
 * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
   DAI link components.
 * sysfs entries can now be registered per component per card.
 * snd_soc_new_pcms() functionailty rolled into dai_link_probe().
 * snd_soc_register_codec() now does all the codec list and mutex init.

This patch changes the probe() and remove() of the CODEC drivers as follows:-

 o Make CODEC driver a platform driver
 o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
 o Removed all static codec pointers (drivers now support > 1 codec dev)
 o snd_soc_register_pcms() now done by core.
 o snd_soc_register_dai() folded into snd_soc_register_codec().

CS4270 portions:
Acked-by: NTimur Tabi <timur@freescale.com>

Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: NRyan Mallon <ryan@bluewatersys.com>

TI CODEC and OMAP fixes
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NJanusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>

Samsung platform and misc fixes :-
Signed-off-by: NChanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: NJoonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: NKyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: NJassi Brar <jassi.brar@samsung.com>
Signed-off-by: NSeungwhan Youn <sw.youn@samsung.com>

MPC8610 and PPC fixes.
Signed-off-by: NTimur Tabi <timur@freescale.com>

i.MX fixes and some core fixes.
Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de>

J4740 platform fixes:-
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>

CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.
Signed-off-by: NFelipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The core will ensure that the device is in either STANDBY or OFF bias
before suspending, restoring the bias in the driver is unneeded. Some
drivers doing slightly more roundabout things have been left alone
for now.
Tested-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Do not change the codec defaults for the following registers:
0x40, 0x41: Line output gains, do not use amplification
0x42: LOM/LOP Voltage hold, and selection
0x44: LOM inversion control

It has been found, that the values configured to these registers
can cause amplification, which can make the output of DAC33
distorted.

The codec reset values are considered safe in all environmnts.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Let the codec to hit OFF instead of STANDBY, when there is no activity.
When the codec is off, than the associated regulator can be also turned
off (if the number of users on the regulator is 0).

After initialization, the codec remains in power off, it is only turned
on for reading the ID registers (also testing the regulators).

The codec power is enabled, when the codec is moving from BIAS_OFF
to BIAS_STANDBY.
The codec is turned off, when it hits BIAS_OFF.

There are few scenarios, which has to be taken care::
1. Analog bypass caused BIAS_OFF -> BIAS_ON
   We need to power on the codec, and do the chip init, but we does not
   need to execute the playback related configuration
2. Playback caused  BIAS_OFF -> BIAS_ON
   We need to power on the codec, and do the chip init, and also we need
   to execute the playback related configuration.
3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON)
   We need to execute the playback related configuration. The codec is
   already on.
4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON)
   Nothing need to be done.
5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON)
   We need to execute the playback related configuration. The codec is
   still on.

Since the power up, and the codec init is optimized, the added overhead
in stream start is minimal.

Withing this patch, the hard_power function is now only doing what it
supposed to: only handle the powers, and GPIO reset line.
The codec initialization and state restore has been moved out.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

As a preparation for supporting codec to be turned off,
when we are in BIAS_STANDBY.

The substream must be easily available in other places than
pcm_* callbacks.

Manage a pointer in _startup, and _shutdown for this.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Optimize the way how tlv320dac33 is powered uppon module and
soc initialization.
Also read the DAC33 ID registers, and update the reg_cache
to reflect it.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

On power up we only need to initialize the codec, and
restore only registers, which are not in either in DAPM
nor in the playback start sequence.
These are mostly gain related registers.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Delay reporting for the three implemented DAC33 FIFO modes.
DAC33 has FIFO depth status register(s), but it can not be used, since
inside of pcm_pointer we can not send I2C commands.
Timestamp based estimation need to be used. The method of calculating
the delay depends on the active FIFO mode.

Bypass mode: FIFO is bypassed, report 0 as delay

Mode1: nSample fill mode. In this mode I need to use two timestamp
ts1: taken when the interrupt has been received
ts2: taken before writing to nSample register.

Interrupts are coming when DAC33 FIFO depth goes under alarm threshold.

Phase1: when we received the alarm threshold, but our workqueue has
        not been executed (safeguard phase). Just count the played out
        samples since ts1 and subtract it from the alarm threshold
        value.
Phase2: During nSample burst (after writing to nSample register), count
        the played out samples since ts1, count the samples received
        since ts2 (in a burst). Estimate the FIFO depth using these and
        alarm threshold value.
Phase3: Draining phase (after the burst read), count the played out
        samples since ts1. Estimate the FIFO depth using the nSample
        configuration and the alarm threshold value.

Mode7: Threshold based fill mode. In this mode one timestamp is enough.
ts1: taken when the interrupt has been received

Interrupts are coming when DAC33 FIFO depth reaches upper threshold.

Phase1: Draining phase (after the burst), counting the played out
        samples since ts1, and subtract it from the upper threshold
        value.
Phase2: During burst operation. Using the pre calculated time needed to
        play out samples from the buffer during the drain period (from
        upper to lower threshold), move the time window to cover the
        estimated time from the burst start to the current time.
        Calculate the samples played out since lower threshold and also
        the samples received during the same time.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>