This reverts commit 6f399115.

Since core has now support for limiting the volume on controls this
patch is not needed.  Furthermore, this patch actually prevents the core
to set new volume on the TPA.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)

If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:

snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);

This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35

The S3C DMA API doesn't make use of hw_addr.to/from and also
the FIFO addresses are provided from the I2S drivers. So these
fields are redundant.
This patch removes the hw_addr.to/from fields for I2S and the
inclusion of header, paving way for the header to be moved closer
to the I2S controller drivers.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc

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>

Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
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>

Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
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>

This patch adds support for integrated stereo speakers and digital
microphone found on Nokia RX-51 hardware. This is a cut down version based
on Maemo kernel sources and earlier patchset by Eduardo Valentin et al.

http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.htmlSigned-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Eduardo Valentin <eduardo.valentin@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NEduardo Valentin <eduardo.valentin@nokia.com>
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>

Now that we can specify feature of a particular controller, we can
avoid multiple copies of same code by defining the CDCLKCON bit
feature in controller specific code and detecting that flag in the
code common to all controllers.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In order to make s3c-i2s-v2.c manage controllers with minor
quirks and variation in features, we define a per-block flag
that indicates the availability/lack of a particular feature
to the s3c-i2s-v2.c

While adding support for new SoCs' I2S, check for the blocks
of older SoCs that have similar feature and set the flag for
that feature.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Now that the fields are defined for s3c2412, use them and avoid having
multiple copies of same defines.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock
doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock
and discard other two copies. Also, switch the users to make calls to the
newly defined and generic s3c_i2sv2_get_clock
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

No need to keep redundant field iis_clk in s3c_i2sv2_info.
iis_cclk and iis_pclk is all we need.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Until now, s3c2412_get_iisclk would return NULL since iis_clk was never
initialized.
Return appropriate pointer as per the selection made for source clock.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The IMS field of s3c2412/13 is essentially the same as that of s3c64xx.
That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit
selects source clock for signal generation.
For that reason, remove improper defines for IISMOD[11:10] field mask
and define two 1bit fields that can be set independent of each other.
As a consequence, corresponding fields for PLAT_S3C64XX too get to use
these new defines.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Define more bit definitions in the order of mainline
support for the SoC.
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The header for I2Sv2
   linux/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h
contains only controller specific definitions and nothing
SoC specific. So, it could be moved to sound/soc/s3c24xx/
Signed-off-by: NJassi Brar <jassi.brar@samsung.com>
Acked-by: NBen Dooks <ben-linux@fluff.org>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Both tpa6130a2, and tpa6140a2 is supported by the
same driver, but the gain dB scaling is different on
the amplifiers.

Provide different mixer control for the chips with correct
TLV mapping.

User space will see:
"TPA6130A2 Headphone Playback Volume" in case of 6130
"TPA6140A2 Headphone Playback Volume" in case of 6140

The way machine drivers are using this amplifier remained
the same.
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>

OUTL/R are leftovers from the original driver, and they
are no longer needed.
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 patch orders the APLL and AIF power sequence in
case of HiFi (audio in TWL4030 terms) playback/capture.

We also need to make sure that the AIF is running during
playback/capture, when there is no valid DAPM route
available. For this purpose I introduce these virtual
widgets:
/* To have complete playback route all the time */
DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */

/* To have complete capture route all the time */
DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */

/* To have complete playback route for the voice module */
DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */

The DAPM_SUPPLY widgets for APLL and AIF are placed in a way,
that during any audio activity the needed configuration of AIF
and APLL will be enabled (playback, capture, analog loopback,
digital loopback, and voice activity).

The apll reference counting code has been lifted,
and modified from Liam Girdwood's earlier patch.
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 WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.

Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control.  The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-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>

Remove bogus twl4030 pins
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>

Remove bogus TWL4030 pins.
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>

The SYSCLK source is automatically managed when configuring the PLL.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc

This patch adds the TLV320AIC3x supplies and enables all of them for the
entire lifetime of the device.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Move PLL enable from BIAS_ON state to BIAS_PREPARE to be pair with
BIAS_STANDBY where PLL is disabled. Remove also old comments about power
control.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

These ADC, DAC and output pin power off commands are needless in
aic3x_set_bias_level since they are not enabled in aic3x_init and they are
defined in aic3x_dapm_widgets so the ASoC DAPM will take care of them
anyway.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.
Signed-off-by: NVladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Reported-by: NAnti Sullin <anti.sullin@artecdesign.ee>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-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>