We no longer have users for the set_hs_extmute callback which has been
replaced by hs_extmute_gpio so the codec driver can handle the external
mute if it is needed by the board.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The external mute (if it is in use) is handled by a GPIO line. Prepare to
remove the set_hs_extmute callback and replace it with:
hs_extmute_gpio: the GPIO number to use for external mute
When the users of set_hs_extmute has been converted the callback can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use DAPM mapping for stream events and give unique names for the streams.
This change also fixes the following warning:
twl4030-codec twl4030-codec: Failed to create Capture debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to support CODEC<->CODEC links remove the assumption that there
is only a single CODEC on a DAI link by removing the use of the CODEC
pointer in the rtd from the CODEC drivers. They are already being passed
their DAI whenever they are passed an rtd and can get the CODEC from
there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the printk(KERN_ERR* instances with dev_err in the driver.
While we are here clean up some of the debug messages as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Correct spelling "memroy" to "memory" in
sound/soc/codecs/twl4030.c
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC
drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver
for devices can unconditionally support idle_bias_off.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Core can set the msbits constraint in behalf of the dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 1ee46ebd("ASoC: Make the DAI ops constant in the DAI structure")
introduced the possibility to have constant DAI ops structures, yet this is
barley used in both existing drivers and also new drivers being submitted,
although none of them modifies its DAI ops structure. The later is not
surprising since existing drivers are often used as templates for new drivers.
So this patch just constifies all existing snd_soc_dai_ops structs to eliminate
the issue altogether.
The patch was generated with the following coccinelle semantic patch:
// <smpl>
@@
identifier ops;
@@
-struct snd_soc_dai_ops ops =
+const struct snd_soc_dai_ops ops =
{ ... };
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the macros for controls require custom get/put function.
This is to make sure that the soc_mixer_control is used
consistently among the drivers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Arun KS <arunks@mistralsolutions.com>
Cc: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel 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: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: Samuel 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: Samuel Ortiz <sameo@linux.intel.com>
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: Andres Salomon <dilinger@queued.net>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Samuel 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need to include soc-dapm.h since soc.h includes it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark 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: Axel Lin <axel.lin@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko 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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the busy loop delays with usleep_range or msleep calls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need for the power down wrapper.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reset most of the codec registers to their chip reset
value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
OUTL/R are leftovers from the original driver, and they
are no longer needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark 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: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec structure initialization statements should be
separated by semicolons.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark 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: Rajendra Nayak <rnayak@ti.com>
Signed-off-by: Balaji T K <balajitk@ti.com>
Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
The upcoming TWL6030 is companion chip for OMAP4 like the current TWL4030
for OMAP3. The common modules like RTC, Regulator creates opportunity
to re-use the most of the code from twl4030.
This patch renames few common drivers twl4030* files to twl* to enable
the code re-use.
Signed-off-by: Rajendra Nayak <rnayak@ti.com>
Signed-off-by: Balaji T K <balajitk@ti.com>
Signed-off-by: Santosh Shilimkar <santosh.shilimkar@ti.com>
Acked-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
The DAPM widgets are now insntantiated by the core when creating the card
so there is no need for the individual CODEC drivers to do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
APLL_CTL register is configured by the twl4030-codec MFD
driver.
Remove code, which makes changes in the APLL_CTL register,
and replace those with checks against the configured
audio_mclk configuration done in the MFD driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set the codec->bias_level to SND_SOC_BIAS_OFF before changing
the initial bias level to STANDBY.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Capture path also need the APLL enabled, adding DAPM_SUPPLY
for the Virtual ADCs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.
If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.
Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.
Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
codec_muted is missleading, change it to apll_enabled,
which is what it is doing: enabing and disabling the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.
Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.
Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the way how the twl4030 soc codec driver is
loaded/probed.
Use the device probing via tlw4030_codec MFD device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dynamically control and control only the needed output amplifier
muting/un-muting.
The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.
Move these as separate PGA so only the needed amplifier will be touched.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add "set_tristate" callbacks for HiFi and Voice DAIs.
Machine drivers can enable and disable tristate for each
DAI with "snd_soc_dai_set_tristate" function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to TRM, an external FET controlled by a 1.8V output signal
can be used to reduce the pop-noise heard when the audio amplifier is
switched on. It is suggested that GPIO6 of TWL4030 be used, but any
other gpio can be used instead. This is indicated in machine driver
with the following twl4030_setup_data members:
-hs_extmute. Set to 1 if board has support for EXTMUTE.
-set_hs_extmute. Set to a callback funcion to control an external gpio
line. Set to NULL if MUTE[GPIO6] pin is used.
Codec driver takes care of enabling and disabling this output during
the headset pop attenuation sequence.
Also add a delay to let VMID settle in ramp up sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls,
removing word "Switch" from HandsfreeL/HandsfreeR widget name
for avoiding to duplicate it.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Event for voice sidetone was being interpreted as an
analog HiFi bypass event because VSTPGA register offset
is less than ARXR2_APGA_CTL offset. Reordering the
register checks allows to handle voice digital bypass
event properly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AVDAC clk priority allows to determine the path ADC must
be connected when the codec is in option2 and both HiFi
and Voice paths are enabled.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Voice interface of twl4030 codec supports: CBM_CFM and
CBS_CFS. It doesn't support CBS_CFM.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No need to define second copies of mode and format, they're declared
with exactly the same type at the head of the function and there's no
conflict in their use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the (likely cut-n-paste) error by commit
16a30fbb0d, which causes the error below:
sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache':
sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In addition to the operating mode check, also check the
codec's interface format in case of four channel mode.
If the codec is not in TDM (DSP_A) mode, return with error.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the codec->reg_cache instead of the array directly
in twl4030_init_chip for setting the default values.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DAPM switch for HeadsetL/R mute. Since all bits are are needed
for the HFL/R pop removal to work the switch is using the SW_SHADOW
no HW register for the HandsfreeL/R mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shadow, non HW register for dealing with the HandsfreeL/R
muting.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_E
to a more appropriate DAPM_PGA_E widget.
Also fix the power-up sequence to match with the TRM.
The power-down sequence is not described in the TRM, so do it
in a way, which seams like the correct sequence.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Give unique stream names for the two playback streams so
DAPM can figure out which codec_dai is in use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
twl4030_setup_data structure can be passed from platform drivers to
the codec via the snd_soc_device->codec_data pointer.
Currently the setup data has support for the Headset pop-removal
related configuration, which differs from board to board.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles
the headset ramp up and down sequences needed for the pop noise
removal.
With this patch the order of the internal components in the twl4030
codec is turned on and off in a correct order.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Restructuring the twl4030 codec's DAPM routing to be able to handle the power
sequences correctly.
The twl4030 codec internal implementation have this order:
DAC -> Analog PGA -> Mixer/Mux
While the ASoC framework expects the following order:
DAC -> Mixer -> Analog PGA
This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and
adds two levels of mixer to handle the digital and analog loopback
functionality.
Now the analog loopback does not powers on any of the DACs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a control for selecting the codec operation mode. TWL4030 codec
has two modes:
- Option 1. Audio only (4 audio DACs)
- Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC)
Control is restricted when a stream is ongoing, since codec's
operation mode cannot be changed on-the-fly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink
and downlink paths, respectively.
This patch also corrects voice 8/16kHz mode selection bit
(SEL_16K) of CODEC_MODE register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The inputs of the twl4030 codec can be mixed, so we will use the mixer
DAPM for the analog microphone registers(0x05, 0x06), but if we enable
more than one input at the same time, the input impedance of the input
amplifier will be reduced.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN
It has not caused problems, since
TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the VIBRA output on TWL4030 codec.
The VIBRA output can be driven with audio data or with
local vibrator driver.
Add the needed DAPM elements and routes for the VIBRA output and
controls for the VIBRA driver configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add voice digital loopback (sidetone) to the twl4030
driver. It mixes voice uplink attenuated (by sidetone gain) with
voice downlink when the codec is working in option2 (voice/audio
mode).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds voice downlink analog bypass switch. It follows
the same approach as in other analog bypass switches.
DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer',
that will also allow voice DAC to be powered in digital voice
loopback (sidetone).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The gain control for earpiece amplifier uses 0dB ~ 12dB according to the
TRM, but the present code is implemented to -6dB ~ 6dB.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
