Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver

To avoid breakage, change all depending drivers, files
to use the new types.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NSamuel Ortiz <sameo@linux.intel.com>

Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: NSamuel Ortiz <sameo@linux.intel.com>

With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.

Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: NSamuel Ortiz <sameo@linux.intel.com>

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

The cell's platform_data is now accessed with a helper function;
change clients to use that, and remove the now-unused data_size.
Signed-off-by: NAndres Salomon <dilinger@queued.net>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NSamuel Ortiz <sameo@linux.intel.com>

twl4030 series of codecs supports S32_LE with msbits=24.
Replace the S24_LE with S32_LE format, and add constraint
for 24msbit in case of 32 S32_LE format.
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>

As we allocate memory for twl4030 in twl4030_codec_probe(),
twl4030_codec_remove() is a better place to free the memory.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

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>

Change the busy loop delays with usleep_range or msleep calls.
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>

Add back the register restore call, when the codec driver is
removed.
This does not affect normal operation, but it is usefull when
debugging audio through the twl4030 class codecs.
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 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>

Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.
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 handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the 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>

When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.

Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.

Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.

Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias 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>

Restructure the DAPM connections in order to enable
only the needed DAC (out of four in twl4030 series).
I need to keep the 'AIF Enable' supply connected to
the L2/R2 digital path, since the digital loopback
needs AIF and APLL running.
If no valid route available, than none of the DAC will
be powered, but the AIF and APLL is going to be enabled.
Furthermore, if only one audio path have valid route,
than only the corresponding DAC will be powered.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

When the gain is configured using dB value it was
not possible to use -24dB since the loopback got
muted instead of -24dB.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Machine driver can instruct the codec driver to reset the
chip registers to their default values at probe time.

If machine driver does not provide setup data, then the
registers are going to be reseted to their defaults, to
be safe.

If the developer on the platform confirms that the register
reset is not needed, than it can be skipped, saving ~20ms
time in probe.

As safety measure do the register reset at remove time also.
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>

Restructure the codec power code in order to be able to hit
off when the codec is not in use.

Since the audio registers are accessible while the codec is powered
down, there is no need for additional safety mechanism.
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>

It seams at least on twl5031 that the ARXR2_APGA_CTL register
does not have the same default value as it is written in
the TRM.
Since the codec part of the PM chip has not been actually
changed according to TI, assuming, that all version has
the same problem, so writing there the TRM value.
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 twl4030 codec driver supports different version
of the PM chip, a helper function can come handy, which
can check the driver's default versus the chip values.
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 reg cache now contains the chip default values
for all registers (REG_OPTION is reset to it's default
within this patch), there is no longer need to rewrite
_all_ registers.
Initialize only few selected registers.

According to the latest information, the offset cancellation
need to be done only once, when the codec is powered on, so
there is no need for the power up wrapper.

Move all chip initialization code under chip_init, and do
it when the codec is initialized.
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 means for machine drivers to select the path for offset
cancellation.
Reset the reg cache value to the chip reset value at the
same time.

Machine drivers can specify which path need to be used for
offset cancellation via the twl4030_setup.offset_cncl_path.
For paths use the defines from
include/linux/mfd/twl4030-codec.h:
TWL4030_OFFSET_CNCL_SEL_*
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 for the power down wrapper.
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>

Reset most of the codec registers to their chip reset
value.
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 codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.
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>

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>

Conversions to snd_soc_codec_{get|set}_drvdata() were missing in some files
in the previous commit.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.

To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h

percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files.  percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.

percpu.h -> slab.h dependency is about to be removed.  Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability.  As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.

  http://userweb.kernel.org/~tj/misc/slabh-sweep.py

The script does the followings.

* Scan files for gfp and slab usages and update includes such that
  only the necessary includes are there.  ie. if only gfp is used,
  gfp.h, if slab is used, slab.h.

* When the script inserts a new include, it looks at the include
  blocks and try to put the new include such that its order conforms
  to its surrounding.  It's put in the include block which contains
  core kernel includes, in the same order that the rest are ordered -
  alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
  doesn't seem to be any matching order.

* If the script can't find a place to put a new include (mostly
  because the file doesn't have fitting include block), it prints out
  an error message indicating which .h file needs to be added to the
  file.

The conversion was done in the following steps.

1. The initial automatic conversion of all .c files updated slightly
   over 4000 files, deleting around 700 includes and adding ~480 gfp.h
   and ~3000 slab.h inclusions.  The script emitted errors for ~400
   files.

2. Each error was manually checked.  Some didn't need the inclusion,
   some needed manual addition while adding it to implementation .h or
   embedding .c file was more appropriate for others.  This step added
   inclusions to around 150 files.

3. The script was run again and the output was compared to the edits
   from #2 to make sure no file was left behind.

4. Several build tests were done and a couple of problems were fixed.
   e.g. lib/decompress_*.c used malloc/free() wrappers around slab
   APIs requiring slab.h to be added manually.

5. The script was run on all .h files but without automatically
   editing them as sprinkling gfp.h and slab.h inclusions around .h
   files could easily lead to inclusion dependency hell.  Most gfp.h
   inclusion directives were ignored as stuff from gfp.h was usually
   wildly available and often used in preprocessor macros.  Each
   slab.h inclusion directive was examined and added manually as
   necessary.

6. percpu.h was updated not to include slab.h.

7. Build test were done on the following configurations and failures
   were fixed.  CONFIG_GCOV_KERNEL was turned off for all tests (as my
   distributed build env didn't work with gcov compiles) and a few
   more options had to be turned off depending on archs to make things
   build (like ipr on powerpc/64 which failed due to missing writeq).

   * x86 and x86_64 UP and SMP allmodconfig and a custom test config.
   * powerpc and powerpc64 SMP allmodconfig
   * sparc and sparc64 SMP allmodconfig
   * ia64 SMP allmodconfig
   * s390 SMP allmodconfig
   * alpha SMP allmodconfig
   * um on x86_64 SMP allmodconfig

8. percpu.h modifications were reverted so that it could be applied as
   a separate patch and serve as bisection point.

Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.
Signed-off-by: NTejun Heo <tj@kernel.org>
Guess-its-ok-by: NChristoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>

Gain controls on outputs affect the power consumption
when the gain is set to non 0 value.

Outputs with amps have one register to configure the
routing and the gain:
PREDL_CTL (0x25):
bit 0: Voice enable
bit 1: Audio L1 enable
bit 2: Audio L2 enable
bit 3: Audio R2 enable
bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB)

bit 0 - 3: is handled in DAPM domain (DAPM_MIXER)
bit 4 - 5: has simple volume control

If there is no audio activity (BIAS_STANDBY), and
user changes the volume, than the output amplifier will
be enabled.
If the user changes the routing (but the codec remains in
BIAS_STANDBY), than the cached gain value also be written
to the register, which enables the amplifier.

The existing workaround for this is to have virtual
PGAs associated with the outputs, and whit DAPM PMD
the gain on the output will be forced to 0 (off) by
bypassing the regcache.
This failed to disable the amplifiers in several
scenario (as mentioned above).

Also if the codec is in BIAS_ON state, and user modifies
a volume control, which path is actually not enabled, than
that amplifier will be enabled as well, but it will
be not turned off, since there is no DAPM path, which
would make mute it.

To prevent amps being enabled, when they are not
needed, introduce the following workaround:
Track the state of each of this type of output.
In twl4030_write only allow actual write, when the
given output is enabled, otherwise only update
the reg_cache.
The PGA event handlers on power up will write the cached
value to the chip (restoring gain, routing selection).
On power down 0 is written to the register (disabling
the amp, and also just in case clearing the routing).
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>

Disable the amplifiers for the headset outputs, and do not select
routings by default to the headset outputs.
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>

The serial interface (TDM/I2S) for the audio block have been
constantly enabled.
Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so
the interface is only enabled, when there is a need for it.
For example when only the analog loopback is enabled, there
is no need to keep the serial interface active.

I have added the persons who contributed to the Voice path
of twl4030 codec driver, so they might have the ability
to test this patch, and send an update for the Voice path,
if it is necessary
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>

The module unloading path had several problems:
- it freed up the private structure twice
- it freed up the codec structure, which was allocated as part
  of the private structure
- it did not freed up the reg_cache
- it did not unregistered the dais and the codec
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>

Change the legacy default register configuration, which left some
internal components on.
Now we have either DAPM, or other ways to control these bits,
so there is no need to enable them by default.

The affected parts:
Disable ADCL and ADCR
Disable ARXL2 and ARXR2 analog PGA (playback)
Disable APLL by default
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>

The codec structure initialization statements should be
separated by semicolons.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch renames function names like twl4030_i2c_write_u8,
twl4030_i2c_read_u8 to twl_i2c_write_u8, twl_i2c_read_u8
and also common variable in twl-core.c
Signed-off-by: NRajendra Nayak <rnayak@ti.com>
Signed-off-by: NBalaji T K <balajitk@ti.com>
Signed-off-by: NSantosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: NKevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: NSamuel Ortiz <sameo@linux.intel.com>