Variable pulse_len is being initialized to 1 however this value is
never read and pulse_len is being re-assigned later in a switch
statement. Clean up the code by removing the redundant initialization.
Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Based on 1 normalized pattern(s):
this program is free software you can redistribute it and or modify
it under the terms of the gnu general public license as published by
the free software foundation either version 2 of the license or at
your option any later version this program is distributed in the
hope that it will be useful but without any warranty without even
the implied warranty of merchantability or fitness for a particular
purpose see the gnu general public license for more details you
should have received a copy of the gnu general public license along
with this program if not see the file copying or write to the free
software foundation inc 51 franklin st fifth floor boston ma 02110
1301 usa
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-or-later
has been chosen to replace the boilerplate/reference in 4 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Richard Fontana <rfontana@redhat.com>
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190520071859.003492977@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Initial commit of the Conexant CX2072X CODEC driver. Some features are
not present.
The CX2072X is a ultra low power stereo audio codec supports I2S/TDM
host interface with EQ, DRC features in playback mode.
Featues of CX2072X codec:
* Two 24 bits DACs and DACs.
* Stereo Headphone AMP.
* 2.8W per channel class-D output.
* Integrated seven bands per channel EQ and DRC.
* Fully integrated headset support with detect/switch.
* Stereo digital microphone for array applications.
* S/PDIF output.
* Bi-directional GPIOs.
* Support analog and digital PC Beeep.
* One headset button support.
* Supports a wide variety of host interfaces.
-wide variety of I2S and similar bit-stream formats
with word lengths of up to 24 bits.
-TDM stream supports up to 4 channels.
* AEC loopback support.
Further fixes by tiwai:
* Rebase to 5.2+
* Missing DAPM entry definitions
* Missing power hooks
* Fix uninitialized variable warning
* Rewrite jack detection stuff to use set_jack callback
* Plumbing jack detection code for Intel ASoC
* Move clk management into runtime PM
* Drop incorrect regcache usages
* Drop untested stuff: OF table, EQ/DRC handling
* Lots of code cleanups and minor refactoring
The OF code was dropped due to the lack of testability.
It should be easy to re-add once if someone can test it.
v1->v2: No change
v2->v3: Move register tables to appropriate place
Remove some confusing codes
Set snd_ctl_boolean_* helpers directly
Fix EQ put callback
Rename to "DAC1 Switch" from "DAC1 Mute Switch"
Drop superfluous regmap calls at shutdown
Avoid regmap_register_patch()
Add missing register definitions
Fix register access on big-endian machine
Remove regcache messes
v3->v4: Fix the wrong endianess conversion in reg write
Minor code cleanups
v4->v5: Move clk management to runtime PM
Sparse warning fixes
Some more code simplification
Drop tricky regcache fiddling
Apply mutex locks around possible racy sequences
Move exported jack detection stuff into set_jack callback
v5->v6: Drop buggy&untested EQ and DRC codes
Lots of code reduction/cleanup
Add more comments about platform-specific stuff
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=115531
Signed-off-by: Simon Ho <simon.ho@conexant.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Based on 2 normalized pattern(s):
this program is free software you can redistribute it and or modify
it under the terms of the gnu general public license as published by
the free software foundation either version 2 of the license or at
your option any later version this program is distributed in the
hope that it will be useful but without any warranty without even
the implied warranty of merchantability or fitness for a particular
purpose see the gnu general public license for more details you
should have received a copy of the gnu general public license along
with this program if not write to the free software foundation inc
51 franklin street fifth floor boston ma 02110 1301 usa
this program is free software you can redistribute it and or modify
it under the terms of the gnu general public license as published by
the free software foundation either version 2 of the license or at
your option [no]_[pad]_[ctrl] any later version this program is
distributed in the hope that it will be useful but without any
warranty without even the implied warranty of merchantability or
fitness for a particular purpose see the gnu general public license
for more details you should have received a copy of the gnu general
public license along with this program if not write to the free
software foundation inc 51 franklin street fifth floor boston ma
02110 1301 usa
extracted by the scancode license scanner the SPDX license identifier
GPL-2.0-or-later
has been chosen to replace the boilerplate/reference in 176 file(s).
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Reviewed-by: Jilayne Lovejoy <opensource@jilayne.com>
Reviewed-by: Steve Winslow <swinslow@gmail.com>
Reviewed-by: Allison Randal <allison@lohutok.net>
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Cc: linux-spdx@vger.kernel.org
Link: https://lkml.kernel.org/r/20190519154040.652910950@linutronix.de
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Add SPDX license identifiers to all Make/Kconfig files which:
- Have no license information of any form
These files fall under the project license, GPL v2 only. The resulting SPDX
license identifier is:
GPL-2.0-only
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Add SPDX license identifiers to all files which:
- Have no license information of any form
- Have MODULE_LICENCE("GPL*") inside which was used in the initial
scan/conversion to ignore the file
These files fall under the project license, GPL v2 only. The resulting SPDX
license identifier is:
GPL-2.0-only
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
There is a case when a we want to read a large number of bytes that
require a burst but is not a multiple of the word size (8). When this
happens rt5677_spi_reverse will run off the end of the buffer. The
solution is to tell spi_reverse the actual size of the destination and
stop if we reach it even if we have data left that we read.
Cc: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support to output level control for the analog high power output
drivers HPOUT and HPCOM.
Signed-off-by: Saravanan Sekar <sravanhome@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Handle the reset GPIO and reset the device every time we
start it.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add regcache_mark_dirty before regcache_sync for power
of codec may be lost at suspend, then all the register
need to be reconfigured.
Fixes: 0c516b4ff8 ("ASoC: cs42xx8: Add codec driver
support for CS42448/CS42888")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_component_update_bits() may return 1 if operation
was successful and the value of the register changed.
Return a non-zero in ak4458_rstn_control for an error only.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace the bit atomic operations by a mutex to ensure only one dai
at a time is active on the hdmi codec.
This is a follow up on change:
3fcf94ef4d ("ASoC: hdmi-codec: remove reference to the current substream")
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_gpiod_get_optional() returns EBUSY after component rebound.
Request GPIO in max98357a_platform_probe() to support component
rebinding.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AK4458 is probed successfully even if AK4458 is not present - this
is caused by probe function returning no error on i2c access failure.
Return an error on probe if i2c access has failed.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The cs4265_readable_register function stopped short of the maximum
register.
An example bug is taken from :
https://github.com/Audio-Injector/Ultra/issues/25
Where alsactl store fails with :
Cannot read control '2,0,0,C Data Buffer,0': Input/output error
This patch fixes the bug by setting the cs4265 to have readable
registers up to the maximum hardware register CS4265_MAX_REGISTER.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull sound updates from Takashi Iwai:
"The most significant changes at this cycle are the Sound Open Firmware
support from Intel for the common DSP framework along with its support
for Intel platforms. It's a door opened to a real "free" firmware (in
the sense of FOSS), and other parties show interests in it.
In addition to SOF, we've got a bunch of updates and fixes as usual.
Some highlights are below.
ALSA core:
- Cleanups and fixes in ALSA timer code to cover some races spotted
by syzkaller
- Cleanups and fixes in ALSA sequencer code to cover some races,
again unsurprisingly, spotted by syzkaller
- Optimize the common page allocation helper with alloc_pages_exact()
ASoC:
- Add SOF core support, as well as Intel SOF platform support
- Generic card driver improvements: support for MCLK/sample rate
ratio and pin switches
- A big set of improvements to TLV320AIC32x4 drivers
- New drivers for Freescale audio mixers, several Intel machines,
several Mediatek machines, Meson G12A, Spreadtrum compressed audio
and DMA devices
HD-audio:
- A few Realtek codec fixes for reducing pop noises
- Quirks for Chromebooks
- Workaround for faulty connection report on AMD/Nvidia HDMI
Others:
- A quirk for Focusrite Scarlett Solo USB-audio
- Add support for MOTU 8pre FireWire
- 24bit sample format support in aloop
- GUS patch format support (finally, over a decade) in native emux
synth code"
* tag 'sound-5.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (375 commits)
ASoC: SOF: Fix unused variable warnings
ALSA: line6: toneport: Fix broken usage of timer for delayed execution
ALSA: aica: Fix a long-time build breakage
ALSA: hda/realtek - Support low power consumption for ALC256
ASoC: stm32: i2s: update pcm hardware constraints
ASoC: codec: hdac_hdmi: no checking monitor in hw_params
ASoC: mediatek: mt6358: save PGA for mixer control
ASoC: mediatek: mt6358: save output volume for mixer controls
ASoC: mediatek: mt6358: initialize setting when ramping volume
ASoC: SOF: core: fix undefined nocodec reference
ASoC: SOF: xtensa: fix undefined references
ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly
ALSA: hdea/realtek - Headset fixup for System76 Gazelle (gaze14)
ALSA: hda/intel: add CometLake PCI IDs
ALSA: hda/realtek - Support low power consumption for ALC295
ASoC: rockchip: Fix an uninitialized variable compile warning
ASoC: SOF: Fix a compile warning with CONFIG_PCI=n
ASoC: da7219: Fix a compile warning at CONFIG_COMMON_CLK=n
ASoC: sound/soc/sof/: fix kconfig dependency warning
ASoC: stm32: spdifrx: change trace level on iec control
...
ASoC: Fixes for v5.2
A bunch of driver specific fixes that came in since the initial pull
request for v5.2, mainly warning fixes for the newly added Sound Open
Firmware code which people appeared to only start looking at after I'd
sent the pull request.
ASoC: Fixes for v5.2
A bunch of driver specific fixes that came in since the initial pull
request for v5.2, mainly warning fixes for the newly added Sound Open
Firmware code which people appeared to only start looking at after I'd
sent the pull request.
The dependency on the dai_id can be removed by setting different ops
for the i2s and spdif dai and storing the dai format information in
each dai structure. It simplies the code a bit.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Keeping the a pointer to the dai drivers is not necessary. It is not used
by the hdmi_codec after the probe.
Even if it was used, the 'struct snd_soc_dai_driver' can accessed through
the 'struct snd_soc_dai' so keeping the pointer in the private data
structure is not useful.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the hdmi-codec is on a codec-to-codec link, the substream pointer
it receives is completely made up by snd_soc_dai_link_event().
The pointer will be different between startup() and shutdown().
The hdmi-codec complains when this happens even if it is not really a
problem. The current_substream pointer is not used for anything useful
apart from getting the exclusive ownership of the device.
Remove current_substream pointer and replace the exclusive locking
mechanism with a simple variable and some atomic operations.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the debug traces only showing the function name on entry.
The same can be obtained using ftrace.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some userspace apps, like pulseaudio, may call open, hw_params,
prepare to judge whether the pcm is ready or not. Current hdac_hdmi
will return -ENODEV if monitor is not connected, which will cause
the apps believe the pcm is not ready. Actually PCM for hdmi is ready,
even the monitor is not connected.
This patch removes the check of monitor presence in hw_params, just like
what the legacy HD-Audio driver does.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
mt6358_amic_disable() resets PGA to 0.
Save the gain settings from mixer control and restore them when using
the microphone.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Output volume settings from mixer controls would be lost.
Imagine that "Headphone Volume" has set to -10dB via amixer:
- in mtk_hp_enable()
- hp_store_gain() saves the volume setting -10dB from regmap_read()
to ana_gain[AUDIO_ANALOG_VOLUME_HPOUTL]
- headset_volume_ramp() ramps up from -10dB to -10dB
- in mtk_hp_disable()
- headset_volume_ramp() ramps down from -10dB to -40dB
Next time in mtk_hp_enable(), hp_store_gain() would save -40dB but
not -10dB. As a result, headset_volume_ramp() would ramp from -10dB to
-40dB (which is mute).
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mt6358 ramps up from the smallest volume (i.e. -10dB) to target dB when
opening and ramps down from target dB to mute (i.e. -40dB) when closing.
If target is equal to -10dB when opening, headset_volume_ramp() simply
leaves current setting (which may not be -10dB) unchanged.
Execute the loop at least once to initialize the setting to the
starting point (i.e. from).
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add name to component driver so it is possible to lookup the component
when needed.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull spi updates from Mark Brown:
"One small feature was added this release but the bulk of the diffstat
and the changelog comes from the fact that several older drivers got
some fairly hefty reworks and a couple of new drivers were added:
- Support for detailed control of timing around chip selects from
Sowjanya Komatineni.
- A big set of fixes and imrovements for the Tegra114 driver from
Sowjanya Komatineni.
- A big simplification of the GPIO driver from Andrey Smirnov.
- DMA support and fixes for the Freescale LPSPI driver from Clark
Wang.
- Fixes and optimizations for the bcm2835aux from Martin Sparl.
- New drivers for Mediatek MT7621 (graduated from staging) and Zynq
QSPI"
[ This is a so-called "evil merge" that additionally removes a warning
due to an unused variable 'i' introduced by commit 1dfbf334f1 ("spi:
ep93xx: Convert to use CS GPIO descriptors") - Linus ]
* tag 'spi-v5.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi: (127 commits)
spi: rspi: Fix handling of QSPI code when transmit and receive
spi: atmel-quadspi: fix crash while suspending
spi: stm32: return the get_irq error
spi: tegra114: fix PIO transfer
spi: pxa2xx: fix SCR (divisor) calculation
spi: Clear SPI_CS_HIGH flag from bad_bits for GPIO chip-select
spi: ep93xx: Convert to use CS GPIO descriptors
spi: AD ASoC: declare missing of table
spi: spi-mem: zynq-qspi: Fix build error on architectures missing readsl/writesl
spi: stm32-qspi: manage the get_irq error case
spi/spi-bcm2835: Split transfers that exceed DLEN
spi: expand mode support
dt-bindings: spi: spi-mt65xx: add support for MT8516
spi: pxa2xx: Add support for Intel Comet Lake
spi/trace: Cap buffer contents at 64 bytes
spi: Release spi_res after finalizing message
spi: Remove warning in spi_split_transfers_maxsize()
spi: Remove one needless transfer speed fall back case
spi: sh-msiof: Document r8a77470 bindings
spi: pxa2xx: use a module softdep for dw_dmac
...
The DACs volume can go over 0, both according to the data sheet and
real world testing. The control can go up to +30dB.
This was tested by playing audio at full volume on a samus chromebook.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Reviewed-by: Hsinyu Chao <hychao@chromium.org>
Signed-off-by: Ross Zwisler <zwisler@google.com>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
A trivial fix for the randconfig build error:
sound/soc/codecs/da7219.c:2366:6: warning: unused variable ‘i’ [-Wunused-variable]
Fixes: d90ba6c8b5 ("ASoC: da7219: Expose BCLK and WCLK control through CCF")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the SPI driver and the main codec share the same name. This
will become confusing when looking up components when using both
drivers.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The max98090 driver defines 3 DAPM muxes; one for the right line output
(LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux)
and one for the right headphone mixer source (MIXHPRSEL Mux). The same
bit is used for the mux as well as the DAPM enable, and although the mux
can be correctly configured, after playback has completed, the mux will
be reset during the disable phase. This is preventing the state of these
muxes from being saved and restored correctly on system reboot. Fix this
by marking these muxes as SND_SOC_NOPM.
Note this has been verified this on the Tegra124 Nyan Big which features
the MAX98090 codec.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The current algorithm allows 3 types of transfers, 16bit, 32bit and
burst. According to Realtek, 16bit transfers have a special restriction
in that it is restricted to the memory region of
0x18020000 ~ 0x18021000. This region is the memory location of the I2C
registers. The current algorithm does not uphold this restriction and
therefore fails to complete writes.
Since this has been broken for some time it likely no one is using it.
Better to simply disable the 16 bit writes. This will allow users to
properly load firmware over SPI without data corruption.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
ASoC: Updates for v5.2
This is a pretty huge set of changes, it's been a pretty active release
all round but the big thing with this release is the Sound Open Firmware
changes from Intel, providing another DSP framework for use with the
DSPs in their SoCs. This one works with the firmware of the same name
which is free software (unlike the previous DSP firmwares and framework)
and there has been some interest in adoption by other systems already so
hopefully we will see adoption by other vendors in the future.
Other highlights include:
- Support for MCLK/sample rate ratio setting in the generic cards.
- Support for pin switches in the generic cards.
- A big set of improvements to the TLV320AIC32x4 drivers from Annaliese
McDermond.
- New drivers for Freescale audio mixers, several Intel machines,
several Mediatek machines, Meson G12A, Sound Open Firmware and
Spreadtrum compressed audio and DMA devices.
startup() should have run before hw_params() is called, so the
current_substream pointer should already be properly set. There
is no reason to call hdmi_codec_new_stream() again in the
hw_params() callback
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the hdmi codec startup fails, it should clear the current_substream
pointer to free the device. This is properly done for the audio_startup()
callback but for snd_pcm_hw_constraint_eld().
Make sure the pointer cleared if an error is reported.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If we want to set rate to 64000 on da7219, it fails and returns
"snd_pcm_hw_params: Invalid argument".
We should remove 64000 from support rate list because it is not
available.
Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some platforms where DA7219 is the DAI clock master, BCLK/WCLK
will be set and enabled prior to the codec's hw_params() function
being called. It is possible the platform requires a different
BCLK configuration than would be chosen by hw_params(), for
example S16_LE format needed with a 64-bit frame to satisfy certain
devices using the clocks.
To handle those kinds of scenarios, the use of clk_round_rate() is
now employed as part of hw_params(). If BCLK is already enabled
then this function will just return the currently set rate, if it
is valid for the desired frame size, so the subsequent call to
clk_set_rate() will succeed and nothing changes with regards to
clocking. In addition the specific BCLK & WCLK recalc_rate()
implementations needed updating to always give back a real value,
as those functions are called as part of the clk init code and a
real value is needed for the clk_round_rate() call to work as
expected.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The LattePanda board has a sound card chtrt5645, when there is nothing
plugged in the headphone jack, the system thinks the headphone is
plugged in, while we plug a headphone in the jack, the system thinks
the headphone is unplugged.
If adding quirk=0x21 in the module parameter, the headphone jack can
work well. So let us fix it via platform_data.
https://bugs.launchpad.net/bugs/182459
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Due to a typo the wrong base is being supplied for the primary algorithm
on Halo firmwares, which will cause the controls to not function.
Fixes: 170b1e123f ("ASoC: wm_adsp: Add support for new Halo core DSPs")
Reported-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Whilst this isn't strictly necessary as the code is already DSP specific
better to use the pointers to avoid potential issues in the future if
one core ends up having multiple methods of stopping the watchdog.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the da7213 codec is configured as Master with the DAPM power down
delay time set, 'snd_soc_component_write' function overwrites the
DAI_CLK_EN bit of DAI_CLK_MODE register which leads to audio play
only once until it re-initialize after codec power up.
Signed-off-by: Logesh <logesh.kolandavel@timesys.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
