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>
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
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Merge tag 'sound-5.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
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
...
Redone to remove the misfiled SPI commit
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Merge tag 'asoc-v5.2-5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
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.
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.
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Merge tag 'asoc-v5.2-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
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>
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.
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Merge tag 'spi-v5.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/spi
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
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 incldue:
- 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.
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Merge tag 'asoc-v5.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
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>
Fix compiler warning about uninitialized variable reported by
Stephen Rothwell <sfr@canb.auug.org.au>.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case create_singlethread_workqueue fails, the fix returns
-ENOMEM to avoid potential NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_kcalloc() may fail and return NULL. The fix returns ENOMEM
in case it fails to avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/codecs/tlv320aic32x4.c: In function 'aic32x4_setup_clocks':
sound/soc/codecs/tlv320aic32x4.c:669:16: warning: variable 'mclk_rate' set but not used [-Wunused-but-set-variable]
It is not used since introduction in
commit 96c3bb0023 ("ASoC: tlv320aic32x4: Dynamically Determine Clocking")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 24-bit TDM mode also applies to DSP_A and DSP_B modes.
Most dais on the SoC side can not interpret I2S/Left_j with other than 2
channels of audio.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_err message. Fix it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add missing <of_device_id> table for SPI driver relying on SPI
device match since compatible is in a DT binding or in a DTS.
Before this patch:
modinfo sound/soc/codecs/snd-soc-adau1977-spi.ko | grep alias
alias: spi:adau1979
alias: spi:adau1978
alias: spi:adau1977
After this patch:
modinfo sound/soc/codecs/snd-soc-adau1977-spi.ko | grep alias
alias: of:N*T*Cadi,adau1979C*
alias: of:N*T*Cadi,adau1979
alias: of:N*T*Cadi,adau1978C*
alias: of:N*T*Cadi,adau1978
alias: of:N*T*Cadi,adau1977C*
alias: of:N*T*Cadi,adau1977
alias: spi:adau1979
alias: spi:adau1978
alias: spi:adau1977
Reported-by: Javier Martinez Canillas <javier@dowhile0.org>
Signed-off-by: Daniel Gomez <dagmcr@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In ASoC driver, snd_hdac_device_register() will be called by
snd_hdac_ext_bus_device_init() and snd_hdac_device_unregister()
will called by snd_hdac_ext_bus_device_remove(). However when
ASoC codec driver call snd_hda_codec_device_new() to create a
new hda codec, it will assign snd_hda_codec_dev_free() to the
dev_free ops and snd_hda_codec_dev_free() will call
snd_hdac_device_unregister(). As a result, snd_hdac_device_unregister()
will be called twice in ASoC driver. To prevent it, we use hdev
type to determine if the hda codec is registered by legacy HDA
driver or ASoC driver and unregister device in snd_hda_codec_dev_free()
only if it is a legacy HDA device.
This patch will overwrite the hdev type so that we can know it is
a ASoC device.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In rt5682 codec driver, a mutex called "calibrate_mutex" is used
in rt5682_calibrate() before initialization, which causes warning
in lock debug. Move the initialization before the usage of mutex.
Signed-off-by: Xun Zhang <xun2.zhang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In resume from S3, HDAC HDMI codec driver dapm event callback may be
operated before HDMI codec driver turns on the display audio power
domain because of the contest between display driver and hdmi codec driver.
This patch adds the device_link between soc card device (consumer) and
hdmi codec device (supplier) to make sure the sequence is always correct.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
wcd9335.c: undefined reference to 'devm_regmap_add_irq_chip'
Signed-off-by: Marc Gonzalez <marc.w.gonzalez@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix some cosmetic spacing issues reported by Julia Lawall
<julia.lawall@lip6.fr>.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Providing a range for usleep_range() allows the hrtimer subsystem to
coalesce timers - as this delay has no upper limit anyway (interrupts
or context switch is possible) it should not hurt to extend this
from 2 to 2-4 milliseconds.
Signed-off-by: Nicholas Mc Guire <hofrat@opentech.at>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use SND_SOC_DAPM_SUPPLY for mic bias DAPM
instead of deprecated SND_SOC_DAPM_MICBIAS.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The chips main power supplies VA and VP are enabled during probe but
then never disabled, this will cause warnings from the regulator
framework on driver removal. Fix this by adding a remove callback and
disabling the supplies, whilst doing so follow best practice and put the
chip back into reset as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add sleep PM callbacks to support system low power modes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support of master mode for cs42l51 cirrus audio codec.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add cs42l51 audio codec power supply management
through regulator framework.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is unsafe to call snd_compr_stop_error from outside of the
compressed ops. Firstly the compressed device lock needs to be held
and secondly it queues error work to issue a trigger stop which
should not happen after the stream has been freed. To avoid these
issues use the same trick used for the IRQ handling, simply send a
snd_compr_fragment_elapsed to cause user-space to wake on the poll,
then report the error when user-space issues the pointer request
after it wakes.
Fixes: a2bcbc1b9a ("ASoC: wm_adsp: Shutdown any compressed streams on DSP watchdog timeout")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@kernel.org
According the publicly available datasheet (and some test) the max98357a
also supports 32, 44.1 and 88.2 kHz sample rate.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PowerTune controls the power level of the chip. On playback this
indirectly controls things like the gain of the various output
amplifiers. This can allow for the decrease of output levels
from the codec. This adds controls for those power levels to
the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a switch for setting common mode voltage. This can allow
for higher drive levels on the amplifier outputs.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update the copyright dates and use the SPDX identifier instead
of reciting the license.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The author of these files has changed her name. Update
instances in the code of her dead name to current legal
name.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tidy up some instances of dereferencing to obtain things that are
already stored in local variables.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm_adsp_compr_detach is NULL aware so there is no need to check for NULL
before calling it, remove the redundant check.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trigger stop can be called in situations where trigger start failed
and as such it can't be assumed the buffer is already attached to
the compressed stream or a NULL pointer may be dereferenced.
Fixes: 639e5eb3c7 ("ASoC: wm_adsp: Correct handling of compressed streams that restart")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The call to of_parse_phandle returns a node pointer with refcount
incremented thus it must be explicitly decremented after the last
usage.
Detected by coccinelle with the following warnings:
./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function.
Signed-off-by: Wen Yang <wen.yang99@zte.com.cn>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Vinod Koul <vkoul@kernel.org>
Cc: Dan Carpenter <dan.carpenter@oracle.com> (commit_signer:1/11=9%,authored:1/11=9%)
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can optionally detect headset or microphone button presses.
Add support for this by passing this event to the jack layer.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This device can detect the insertion/removal of headphones and headsets.
Enable reporting this status by enabling this interrupt and forwarding
this to upper-layers if a jack has been defined.
This jack definition and the resulting operation from a jack detection
event must currently be defined by sound card platform code until CODEC
outputs to jack mappings can be defined generically.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On some devices (Teclast X98+ II tablet, maybe others), the jack
detection has been wired backwards, so when the ES8316 reports
headphones being present it means they are actually not plugged.
Use a quirk around this incorrect behaviour, which can be enabled
through the 'everest,jack-detect-inverted' boolean device property.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the ACPI ID for the product "chromebook pixel 2015" to match the
coreboot settings.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Support multiple endpoints on cs42L51 codec port
when used in of_graph context.
This patch allows to share the codec port between two CPU DAIs.
Example:
STM32MP157C-DK2 board uses CS42L51 audio codec.
This codec is connected to two serial audio interfaces,
which are configured either as rx or tx.
From AsoC point of view the topolgy is the following:
// 2 CPU DAIs (SAI2A/B), 1 Codec (CS42L51)
Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0
Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0
In the DT two endpoints have to be associated to the codec port:
cs42l51_port: port {
cs42l51_tx_endpoint: endpoint@0 {
remote-endpoint = <&sai2a_endpoint>;
};
cs42l51_rx_endpoint: endpoint@1 {
remote-endpoint = <&sai2b_endpoint>;
};
};
However, when the audio graph card parses the codec nodes, it expects
to find DAI interface indexes matching the endpoints indexes.
The current patch forces the use of DAI id 0 for both endpoints,
which allows to share the codec DAI between the two CPU DAIs
for playback and capture streams respectively.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The common pins were mistakenly not added to the DAPM graph.
Adding these pins will allow valid graphs to be created.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some architectures do not yet support the common clock API at all but
the tlv320aic32x4 driver now requires it.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@kernel.org>
The clocking and processing blocks are now properly set up to
support 192000 sample rates. Allow drivers to ask for that.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
mclk is not used by anything anymore. Remove support for it.
All that information now comes from the clock tree.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sysclk is now managed by the CCF. Change this function
to merely find the system clock and set it using
clk_set_rate.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code uses a static lookup table to determine the
settings of the various clock devices on board the chip. This is
limiting in a couple of ways. First, this doesn't allow for any
master clock rates other than the three that have been
precalculated. Additionally, new sample rates are difficult to
add to the table. Witness that the chip is capable of 192000 Hz
sampling, but it is not provided by this driver. Last, if the
driver is clocked by something that isn't a crystal, the
upstream clock may not be able to achieve exactly the rate
requested in the driver. This will mean that clocking will be
slightly off for the sampling clock or that it won't work at all.
This patch determines the settings for all of the clocks at
runtime considering the real conditions of the clocks in the
system. The rules for the clocks are in TI's SLAA557 application
guide on pages 37, 51 and 77.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move these to separate helper functions. This looks cleaner and fits
better with the new clock setting in CCF.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Control the clock gating to the various clock components to use
the CCF. This allows us to prepare_enalbe only 3 clocks and the
relationships assigned to them will cause upstream clockss to
enable automatically. Additionally we can do this in a single
call to the CCF.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage BDIV divider as components in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage DAC/ADC dividers as components in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage codec clock input as a component in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Model and manage the on-board PLL as a component in the Core
Clock Framework. This should allow us to do some more complex
clock management and power control. Also, some of the
on-board chip clocks can be exposed to the outside, and this
change will make those clocks easier to consume by other
parts of the kernel.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Unlike other drivers probe method, of_match_node return value
is not used or checked. This patch removes the redundant code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Reviewed-by: Steven Price <steven.price@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do division with div_u64 for the PLL calculation.
These errors are fixed and list as follows:
1."__udivdi3" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
2."__aeabi_uldivmod" [sound/soc/codecs/snd-soc-nau8810.ko] undefined!
3. nau8810.c:(.text.nau8810_calc_pll+0xd8): undefined reference to
`__udivdi3'
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Lochnagar is an evaluation and development board for Cirrus
Logic Smart CODEC and Amp devices. It allows the connection of
most Cirrus Logic devices on mini-cards, as well as allowing
connection of various application processor systems to provide a
full evaluation platform.
Lochnagar 2 provides a set of line inputs/outputs, and a USB audio
device. This driver adds support for these analog line connections and
the Lochnagar side of the USB audio link.
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Different processing blocks are required for different sampling
rates and power parameters. Set the processing blocks based
on this information.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
For the purposes of platforms which use the codec as DAI clock
master for the CPU and other codec devices, there is the need to
not only expose the clock gating of BCLK and WCLK but also the
ability to set those rates without going through the ASoC APIs.
To make this possible, the previous CCF implementation in the
driver has been extended to separate BCLK and WCLK out. WCLK is
the parent clock to BCLK, and is also the clock gate for both.
BCLK in HW is a factor/multiplier of WCLK so derives from whatever
SR is chosen for WCLK, hence the need to make it a child of WCLK
for the purposes of CCF. Enabling/disabling either BCLK or WCLK
will result in clocks being ungated/gated accordingly. To simplify
matters, these clocks can only be configured if the codec is set
as master, otherwise CCF control is disallowed.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS4270 does not by default increment the register address on
consecutive writes. During normal operation it doesn't matter as all
register accesses are done individually. At resume time after suspend,
however, the regcache code gathers the biggest possible block of
registers to sync and sends them one on one go.
To fix this, set the INCR bit in all cases.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm_adsp_ops structures should be static and correct two printf
specifiers.
Fixes: 170b1e123f ("ASoC: wm_adsp: Add support for new Halo core DSPs")
Fixes: 4e08d50d1f ("ASoC: wm_adsp: Factor out DSP specific operations")
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Break the clock setting logic out from the main hw_params. It's
rather large and unweildy and makes for a large function. This
also better enables some of the following changes to the clock
tree access in the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A Halo Core DSP has a memory protection unit that can trap and signal
memory access faults. This patch adds a function that dumps the fault
information.
The interrupt reaches the host via the parent codec interrupt controller
so this fault function is exported to be called by the codec driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Halo core is a new generation of audio DSP architecture from
Cirrus Logic. A new iteration of the WMFW file format (v3) is also
added, for this new architecture. Currently this format is not
supported on the old ADSP2 architecture however support may be
added for it in the future.
Signed-off-by: Wen Shi <wenshi@opensource.cirrus.com>
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for the addition of more types of DSP core refactor the
handling of DSP specific operations such as starting the memory or
enabling the core into a set of callbacks. This should make it easier to
add new core types and allow for more code reuse between them.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to duplicate this code for both ADSP1 and 2 as the
handling is exactly the same.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for further additions refactor the reading of the
firmware status.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The original wm_adsp2_early_event took an additional frequency
argument for clocking control so could not be used directly as a
DAPM callback. But this setup could equally be done by the codec
driver function wrapping wm_adsp2_early event. In preparation
for adding support for new core types wm_adsp2_set_dspclk has
been exported, and the freq argument removed so that it can
be used directly as a DAPM callback.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This function is not presently called from outside the adsp code and nor
should it be, as such stop exporting it.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a watchdog timeout is received from the DSP it is safe to assume the
DSP is not functioning anymore and as such any active compressed streams
should be put into an error state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Best to lock across handling the bus error to ensure the DSP doesn't
change power state as we are reading the status registers.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During recent logging improvements it seems two error messages lost
their updates during patch application/rebasing. Add these back in.
Fixes: 0d3fba3e7a ("ASoC: wm_adsp: Improve logging messages")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If ASRC turns on, HW will use clk_dac as the reference clock
whether recording or playback.
Both of clk_dac and clk_adc should set proper clock while using ASRC.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The jack type detection needs the main bias power of analog.
The modification makes sure the main bias power on/off while jack plug/unplug.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The IRQ function may not work when system suspend.
We remove snd_soc_dapm_force_enable_pin function call to
make sure the bias off when idle and run into suspend/resume function.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards use a jack-receptacle with a switch which reports the
jack-inserted status as active-high, rather then the standard active-low
reporting most jacks use.
This commit adds support for it. This is activated by a boolean
"realtek,jack-detect-not-inverted" device-property. The not-inverted
in the device-property name, rather then active-high, was chosen to keep
the device-property naming consistent with the rt5640 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some amplifier may not have a GPIO to control the power, but instead simply
rely on the regulator to power up and down the amplifier.
In order to support those setups, let's make the GPIO optional.
Signed-off-by: Mylène Josserand <mylene.josserand@bootlin.com>
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver will select correct BCLK automatically according to
BCLK and FS information in I2S master mode.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver changes the stream name of DAC and ADC to avoid the issue of
widget with prefixed name. When the machine adds prefixed name for codec,
the stream name of DAI may not find the widgets.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has two issues when machine add prefix name for codec.
(1)The stream name of DAI can't find the AIF widgets.
(2)The drivr can enable/disalbe the MICBIAS and SAR widgets.
The patch will fix these issues caused by prefixed name added.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current implementation of the hdac_hda codec results in zero-valued
samples on capture and noise with headset playback when SOF is used on
platforms with an on-board HDaudio codec. This is root-caused to SOF
using be_hw_params_fixup, and the prepare() call using invalid runtime
fields to determine the format.
This patch moves the format handling to the hw_params() callback, as
done already for hdac_hdmi, to make sure the fixed-up information is
taken into account but keeps the codec initialization in prepare() as
the stream_tag is only available at that time. Moving everything in the
prepare() callback is possible but the code is less elegant so this
two-step solution was chosen.
The solution was tested with the SST driver with no regressions, and all
the issues with SOF playback and capture are solved.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Limiting the value of the passed in params->msbits in the hw_params()
callback is redundant on three counts:
1. We already specify in the DAI driver that we can only handle up to
24 bits. This means msbits will be limited to 24 via the ALSA
constraints imposed by the ASoC core, unless we have multiple codecs
that can handle more bits.
2. Nothing in our hw_params() implementation uses this value.
3. The copy of the params that we are passed by the ASoC core never
reads back the msbits value.
Consequently, this code is unnecessary and does nothing useful. Remove
it.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch
cases where we are expecting to fall through.
This patch fixes the following warning:
In file included from sound/soc/codecs/ab8500-codec.c:24:
sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’:
./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=]
_dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’
dev_err(dai->component->dev,
^~~~~~~
sound/soc/codecs/ab8500-codec.c:2132:2: note: here
default:
^~~~~~~
Warning level 3 was used: -Wimplicit-fallthrough=3
This patch is part of the ongoing efforts to enable
-Wimplicit-fallthrough.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When using the S/PDIF DAI, there is no requirement to call
snd_soc_dai_set_fmt() as there is no DAI format definition that defines
S/PDIF. In any case, S/PDIF does not have separate clocks, this is
embedded into the data stream.
Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt
to configure TDA998x via the hw_params callback fails as the
hdmi_codec_daifmt is left initialised to zero.
Since the S/PDIF DAI will only be used by S/PDIF, prepare the
hdmi_codec_daifmt structure for this format.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes a bug that prevents freeing the reset gpio on unloading
the module.
aic3x_i2c_probe is called when loading the module and it calls list_add
with a probably uninitialized list entry aic3x->list (next = prev = NULL)).
So even if list_del is called it does nothing and in the end the gpio_reset
is not freed. Then a repeated module probing fails silently because
gpio_request fails.
When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move
list_del to aic3x_i2c_remove because aic3x_remove may be called
multiple times without aic3x_i2c_remove being called which leads to
a NULL pointer dereference.
Signed-off-by: Philipp Puschmann <philipp.puschmann@emlix.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the compressed stream implementation has acquired support for
multiple DAI links and compressed streams it has become harder to
interpret messages in the kernel log. Add additional macros to include
the compressed DAI name in the log messages, allowing different streams
to be easily disambiguated.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, only a single compressed stream is supported per firmware.
Add support for multiple compressed streams on a single firmware, this
allows additional features like completely independent trigger words or
separate debug capture streams to be implemented.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the code slightly clearer and prepare things for the addition of
multiple compressed streams on a single DSP core.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for more refactoring add a helper function to strip the
padding from ADSP data.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq_get_irq_data() function doesn't return error pointers, it
returns NULL.
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, compressed buffers can only be specified in the XM memory
region. There is no reason to have such a restriction with the newer
meta-data based way of specifying the buffers, so remove it.
Signed-off-by: Andrew Ford <aford@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a compressed stream is restarted after getting an error, the cached
error value will still be used on the next pointer request, preventing
the stream from starting. Resolve this by ensuring the error status is
updated on trigger start.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADCs are sleeping when the SLEEP bit is set and running when it's
cleared, so the bit should be inverted.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
If platform_data is NULL add reading of optional adi,micbias
property from DT. If adi,micbias is not set keep the default
value for micbias.
Signed-off-by: Bogdan Togorean <bogdan.togorean@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warning:
sound/soc/codecs/wm8741.c:371:5: warning:
symbol 'wm8741_mute' was not declared. Should it be static?
Fixes: 36b1599340 ("ASoC: wm8741: Add digital mute callback")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warnings:
sound/soc/codecs/cs35l36.c:135:20: warning:
symbol 'cs35l36_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:248:6: warning:
symbol 'cs35l36_readable_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:398:6: warning:
symbol 'cs35l36_precious_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:410:6: warning:
symbol 'cs35l36_volatile_reg' was not declared. Should it be static?
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The on-chip PLL can be disabled if on the MCLKI pin we have an external
clock at 512 x fs. This clock can be used as direct internal clock for
ADCs or DACs.
To support this, we add an extra clock id that can be configured
using the set_sysclk() callback.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver only supports DPS_A for DAC, which is configured at probe.
This patch adds support for DSP_A and I2S modes by using the set_fmt()
callback.
A trivial break is also removed from a case's default branch.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By default, the codec starts to interpret the left (first) channel on
the falling edge (low polarity) of LRCLK. However, for DSP_A, the left
channel needs to start on the rising edge of LRCLK. This patch fixes
this channel swap by toggling the bit which selects the LRCLK polarity.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DACs and ADCs on ad193x codecs require a 32 bit slot size. We should
assure that no other size is used.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some ad193x codecs don't have ADCs, so they have no capture capabilities.
This way, we can use this driver in multicodec cards.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous implementatation was restrictive with regards to
BCLK rates for slave mode where the driver would not allow rates
the codec couldn't provide itself as clock master. The codec
is able to automatically determine and handle whatever rate is
provided so this restriction isn't necessary for slave mode. The
code was also flawed with regards to setting of the frame offset
as using rx_mask to explicitly set the offset has the knock on
effect of impacting the min and max channels for the codec, in
soc_pcm_hw_params() through the call to
soc_pcm_codec_params_fixup().
With this update, the driver now only limits frame size if codec
is clock master, and dynamically determines the BCLK offset
relating to WCLK using the tx_mask for slot offset along with the
slot width provided.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously the driver would default the BCLK periods per WCLK to
64, to cover all possible non-TDM scenarios when the codec was
DAI clock master. However some devices require a lower BCLK rate
to operate correctly so with this in mind, this commit updates
the code to be more dynamic, with BCLK rate now based on SR and
word length provided to hw_params().
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warning:
sound/soc/codecs/cros_ec_codec.c:209:27: warning:
symbol 'cros_ec_dai' was not declared. Should it be static?
Fixes: b291f42a37 ("ASoC: cros_ec_codec: Add codec driver for Cros EC")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove including <linux/version.h> that don't need it.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As stated in 'TLV320AIC3254 Application Reference Guide' ([1]):
3.2 Device Startup Lockout Times
After the TLV320AIC3254 initializes through hardware reset at power-up
or software reset, the internal registers initialize to default values.
This initialization takes place within 1ms after pulling the RESET
signal high. During this initialization phase, no register-read or
register-write operation should be performed on ADC or DAC coefficient
buffers. Also, no block within the codec should be powered up during
the initialization phase.
[1] http://www.ti.com/lit/an/slaa408a/slaa408a.pdf
Signed-off-by: Peter Seiderer <ps.report@gmx.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
For correct operation of the digital filtering and other processing on the
WM8741, the user must ensure the correct value of OSR[1:0] is set at all
times.[1] Hence, depending the selected sampling rate, set the OSR (over-
sampling rate) mode in hw_params().
References:
[1] "WM8741 Data Sheet"
Signed-off-by: Sergej Sawazki <sergej@taudac.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sergej Sawazki <sergej@taudac.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Loading/unloading modules exposes issues with memory allocation, which
is a mix of devm_kzalloc and manual kzalloc. Move to devm_k routines
everywhere to simplify all this.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ADC and DAC can be clocked from separate or same sources CLK1 and CLK2.
By default, ADC is clocked from CLK1, and DAC - from CLK2.
This commits allows sound cards to selest a proper clock source during
`hw_params()` via `snd_soc_dai_set_sysclk()`. It makes possible to have a
single clock source for both ADC and DAC.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
Softly reset registers values on module probe
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing license and copyright information which never made it
into the analog driver when the original driver was split in two as part
of the review process.
Link: https://lkml.kernel.org/r/1465582725-30183-3-git-send-email-srinivas.kandagatla@linaro.org
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Regulator notifiers, that were registered during codec driver probing,
must be unregistered during driver release, or device managed versions
have to be used. This patch fixes codec drivers, that weren't explicitly
unregistering notifiers and simplifies those, that did that manually.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add driver support for Cirrus Logic CS35L36 boosted
speaker amplifier
Signed-off-by: James Schulman <james.schulman@cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the header comment to use C++ style, so that it looks more
consistent with the rest of ASoC.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Show the knob to enable or disable the jz4740-codec driver, add a
proper description, and add a dependency on MIPS || COMPILE_TEST, as
this driver is only useful on MIPS.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add license information as a standard SPDX license notifier instead of
custom text.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in the SOC_SINGLE control name. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add jz4725b-codec driver to support the internal CODEC found in the
JZ4725B SoC from Ingenic.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop old license header and switch to SPDX-License-Identifier.
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
These are arrays, not pointers, and they can't be NULL.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the ssm2603 data sheet (control register sequencing), the
digital core should be activated only after all necessary bits in the
power register are enabled, and a delay determined by the decoupling
capacitor on the VMID pin has passed. If the digital core is activated
too early, or even before the ADC is powered up, audible artifacts
appear at the beginning of the recorded signal.
The digital core is also needed for playback, so when recording starts
it may already be enabled. This means we cannot get the power sequence
correct when we want to be able to start recording after playback.
As a workaround put the MIC mute switch into the DAPM routes. This
way we can keep the recording disabled until the MIC Bias has settled
and thus get rid of audible artifacts.
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
m.felsch@pengutronix.de: adapt commit message
m.felsch@pengutronix.de: drop of configuration as mentioned by Mark:
https://patchwork.kernel.org/patch/10407449/
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver use the gpio consumer interface.
Add the header as it's needed.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set the channel number on each AIF widget to allow unused channels not
to be powered up across AIFs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a codec driver to control ChromeOS EC codec.
Use EC Host command to enable/disable I2S recording and control other
configurations.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reviewed-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds audio routing for both playback and capture.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds required dapm widgets for playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds basic controls found in wcd9335 codec.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series.
This patchset adds basic CLASS-H controller apis for WCD codecs after
wcd9335 to use.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC,
It supports both I2S/I2C and SLIMbus audio interfaces.
On slimbus interface it supports two data lanes; 16 Tx ports
and 8 Rx ports. It has Seven DACs and nine dedicated interpolators,
Seven (six audio ADCs, and one VBAT ADC), Multibutton headset
control (MBHC), Active noise cancellation and Sidetone paths
and processing.
This patchset adds very basic support for playback and capture
via the 9 interpolators and ADC respectively.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In function rt5651_i2c_probe(), local variable "ret" could
be uninitialized if function regmap_read() returns -EINVAL.
However, this value is used in if statement. This is
potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rationale behind the current calculation is somewhat obscure [1]
and can yield slightly wrong dividers in certain cases, which the
machine drivers for some boards (like the HiFiBerry DAC+ Pro)
seemingly try to circumvent, by updating the rate fraction so as to
suit this calculation.
The updated calculation should correctly yield the smallest bit clock
rate that would fit the frame.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144219.html
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external
oscillators, to generate 44.1 or 48kHz multiples and are forced to
resort to hacks [1] in order to support 24-bit data without ending up
with fractional dividers. This patch allows the machine driver to use
32-bit frames for 24-bit data to avoid such issues.
Although the datasheet (p. 15) seems to suggest that only a handful
of ratios are supported, it's not very explicit about it, so we allow
the full range of values supported by the underlying register in the
callback, to avoid needlessly rejecting potentially usable
configurations.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.html
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AD/DA ASRC function control two ASRC clock sources separately.
Whether AD/DA filter select which clock source, we enable AD/DA ASRC
function for all cases.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
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Merge tag 'asoc-fix-v5.0-rc2' into asoc-5.1
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
hdmi-codec oopses the kernel when it is unbound from a successfully
bound audio subsystem, and is then rebound:
Unable to handle kernel NULL pointer dereference at virtual address 0000001c
pgd = ee3f0000
[0000001c] *pgd=3cc59831
Internal error: Oops: 817 [#1] PREEMPT ARM
Modules linked in: ext2 snd_soc_spdif_tx vmeta dove_thermal snd_soc_kirkwood ofpart marvell_cesa m25p80 orion_wdt mtd spi_nor des_generic gpio_ir_recv snd_soc_kirkwood_spdif bmm_dmabuf auth_rpcgss nfsd autofs4 etnaviv thermal_sys hwmon gpu_sched tda9950
CPU: 0 PID: 1005 Comm: bash Not tainted 4.20.0+ #1762
Hardware name: Marvell Dove (Cubox)
PC is at hdmi_dai_probe+0x68/0x80
LR is at find_held_lock+0x20/0x94
pc : [<c04c7de0>] lr : [<c0063bf4>] psr: 600f0013
sp : ee15bd28 ip : eebd8b1c fp : c093b488
r10: ee048000 r9 : eebdab18 r8 : ee048600
r7 : 00000001 r6 : 00000000 r5 : 00000000 r4 : ee82c100
r3 : 00000006 r2 : 00000001 r1 : c067e38c r0 : ee82c100
Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none[ 297.318599] Control: 10c5387d Table: 2e3f0019 DAC: 00000051
Process bash (pid: 1005, stack limit = 0xee15a248)
...
[<c04c7de0>] (hdmi_dai_probe) from [<c04b7060>] (soc_probe_dai.part.9+0x34/0x70)
[<c04b7060>] (soc_probe_dai.part.9) from [<c04b81a8>] (snd_soc_instantiate_card+0x734/0xc9c)
[<c04b81a8>] (snd_soc_instantiate_card) from [<c04b8b6c>] (snd_soc_add_component+0x29c/0x378)
[<c04b8b6c>] (snd_soc_add_component) from [<c04b8c8c>] (snd_soc_register_component+0x44/0x54)
[<c04b8c8c>] (snd_soc_register_component) from [<c04c64b4>] (devm_snd_soc_register_component+0x48/0x84)
[<c04c64b4>] (devm_snd_soc_register_component) from [<c04c7be8>] (hdmi_codec_probe+0x150/0x260)
[<c04c7be8>] (hdmi_codec_probe) from [<c0373124>] (platform_drv_probe+0x48/0x98)
This happens because hdmi_dai_probe() attempts to access the HDMI
codec private data, but this has not been assigned by hdmi_dai_probe()
before it calls devm_snd_soc_register_component(). Move the call to
dev_set_drvdata() before devm_snd_soc_register_component() to avoid
this oops.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
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Merge tag 'asoc-fix-v5.0-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
Add a default pdata which can fit most HW design. So we don't need to
add a lot of DMI checking in this driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a potential NULL pointer dereference in case devm_kzalloc()
fails and returns NULL.
Fix this by adding a NULL check on rt5514_dsp.
This issue was detected with the help of Coccinelle.
Fixes: 6eebf35b0e ("ASoC: rt5514: add rt5514 SPI driver")
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix typo which causes headphone no sound while using BCLK
as PLL source.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix below build error:
ERROR: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-msm8916-digital.ko] undefined!
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes unused variables which also fixes below warnings:
msm8916-wcd-digital.c:245:30: warning: 'rx2_mix2_inp1_chain_enum'
defined but not used [-Wunused-const-variable=]
static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE(
^~~~~~~~~~~~~~~~~~~~~~~~
msm8916-wcd-digital.c:234:30: warning: 'rx_mix2_inp1_chain_enum'
defined but not used [-Wunused-const-variable=]
static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE(
^~~~~~~~~~~~~~~~~~~~~~~
msm8916-wcd-digital.c:224:26: warning: 'adc2_mux_text'
defined but not used [-Wunused-const-variable=]
static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
^~~~~~~~~~~~~
msm8916-wcd-digital.c:223:26: warning: 'rx_mix2_text'
defined but not used [-Wunused-const-variable=]
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver cs4341 can be built with SPI and/or I2C, but it has to be one
of them at least. When I2C is set as a module we see the warning below:
sound/soc/codecs/cs4341.c:213:12: warning: ‘cs4341_probe’
defined but not used [-Wunused-function]
static int cs4341_probe(struct device *dev)
^~~~~~~~~~~~
Rework so that we use IS_ENABLED instead of defined. Also change so
SND_SOC_CS4341 depends on SND_SOC_I2C_AND_SPI to we dont' get a link
error when SND_SOC_CS4341=y, I2C=m and REGMAP_I2C=m is set.
Suggested-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Anders Roxell <anders.roxell@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the refactoring of HD-audio display power management, the
display power status is managed per domain. Meanwhile the ASoC
hdac_hdmi driver still keeps and relies (incorrectly) on the
refcounting together with ASoC skl driver, and this leads to the
display state always on.
This patch is an attempt to address the regression by simplifying the
PM code of ASoC skl and hdac_hdmi drivers. Basically, since the
refactoring, we don't have to manage the display power at HD-audio
controller suspend / resume but only at HD-audio HDMI codec suspend /
resume. So the patch drops the superfluous snd_hdac_display_power()
calls in skl driver.
Meanwhile, in hdac_hdmi side, we rewrite the PM call just to re-use
the runtime PM callbacks like other drivers do. Now the logic is
simple: turn off at suspend and turn on at resume.
The patch also fixes the possibly missing display-power off at skl
driver removal as well as some error paths at probe.
Fixes: 029d92c289 ("ALSA: hda: Refactor display power management")
Reported-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SND_SOC_DAPM_MICBIAS is deprecated, replace it with SND_SOC_DAPM_SUPPLY.
MICBIAS voltage wasn't supplied to the microphone with the older
SND_SOC_DAPM_MICBIAS widget, hence the microphone wouldn't work.
This patch fixes the problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By making MCLK parent of DAI clocks, when querying the rate of the
clock the rate returned is now given from the parent clock so
gives the MCLK rate rather than 0 as previously returned. This is
a bit misleading, and actually there's no major reason why we can't
at least return the DAI WCLK rate, as set in HW, so that's what we
now do.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For platforms using the Common Clock Framework to control the
codec's DAI clocks, MCLK should be enabled prior to DAI clocks
being turned on. For some platforms the codec is already
provided with an MCLK reference and can therefore control MCLK
itself as it needs to.
To improve functionality MCLK is now added as a parent to the
DAI clocks, if MCLK was provided, so that if they are enabled MCLK
will automatically be enabled as a prerequisite by the CCF.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In function rt274_jack_detect(), local variable "buf" could
be uninitialized if function regmap_read() returns -EINVAL.
However, it will be used to calculate "hp" and "mic" and
make their value unpredictable while those value are used
in the caller. This is potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture through some of dmic we observe a glitch at the
start of record. This is because we start capturing even before
dmic is ready to send out data.
The optional delay will be applied after enabling the mic.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some board designs hook the jack-detect up to an external GPIO,
rather then to one of the codec pins, add support for this.
Figuring out which GPIO to use is pretty much board specific so I've
chosen to let the machine driver pass the gpio_desc as data argument to
snd_soc_component_set_jack() rather then add support for getting the
GPIO to the codec driver. This keeps the codec code nice and clean.
Note that using an external GPIO for this conflicts with button-press
support, so this commit disables button-press support when an
external GPIO is used.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some BYT platforms have a RT5651 codec while using an ACPI node with
a HID of 10EC5640 to describe the coded. Add the 10EC5640 HID to the
acpi_device_id list, so that the rt5651 will bind to the codec on these
devices.
Like the rt5645 and rt5670 drivers which also have the 10EC5640 ACPI HID
in their acpi_device_id list for similar reasons, the rt5651 driver checks
the codecs device-id register so that it will only bind if the codec
actually is a rt5651 and it will ignore actual rt5640 codecs.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch introduces "sclk-strength" property to allow SCLK pad drive
strength to be changed via device tree.
When running playback test on LS1028ARDB, Tx Frame sync error interrupt
will occur sometimes. Some noises also exist. After changing SCLK pad
drive strength to the maximum value, the issues are gone.
Signed-off-by: Alison Wang <alison.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During the bootup of the kernel, the DAPM bias level is in the OFF
state. As soon as the DAPM framework kicks in it pushes the codec
into STANDBY state.
The probe function doesn't prepare the clock, and STANDBY state
does a clk_disable_unprepare() without checking the previous state.
This leads to an OOPS.
Not transitioning from an OFF state to the STANDBY state fixes the
problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Reported by Coccinelle:
sound/soc/codecs/rt298.c:992:6-8: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:995:6-9: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:317:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:320:5-8: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:348:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:351:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The buf in rl6347a_hw_read is __be32.
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The eq parameters binary is stored in __be. However, it is unsigned short
in rt5645_eq_param_s{} which will cause incorrect type assignment. So add
struct rt5645_eq_param_s_be16{} to store the eq binary and convert it to
unsigned short in rt5645->eq_param.
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Sparse:
da7219.c:841:57: warning: dubious: x & !y
Cc: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Sparse.
da7219.c:440:44: warning: cast to restricted __le16
da7219.c:461:13: warning: incorrect type in assignment (different base types)
da7219.c:461:13: expected unsigned short [unsigned] [usertype] val
da7219.c:461:13: got restricted __le16 [usertype] <noident>
da7219.c:1451:16: warning: incorrect type in assignment (different base types)
da7219.c:1451:16: expected unsigned short [unsigned] [usertype] offset
da7219.c:1451:16: got restricted __le16 [usertype] <noident>
da7219-aad.c:150:37: warning: incorrect type in assignment (different base types)
da7219-aad.c:150:37: expected unsigned short [unsigned] [usertype] tonegen_freq_hptest
da7219-aad.c:150:37: got restricted __le16 [usertype] <noident>
da7219-aad.c:157:37: warning: incorrect type in assignment (different base types)
da7219-aad.c:157:37: expected unsigned short [unsigned] [usertype] tonegen_freq_hptest
da7219-aad.c:157:37: got restricted __le16 [usertype] <noident>
Cc: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
mt6351.c:1418:5-8: Unneeded variable: "ret". Return "0" on line 1437
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/tscs42xx.c:392:5-31: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
nau8824.c:810:6-12: ERROR: Assignment of bool to non-0/1 constant
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt5651.c:750:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5651.c:754:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5651.c:2192:1-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/max98927.c:508:2-20: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:889:3-28: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:891:3-28: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:893:2-27: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt5640.c:980:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5640.c:984:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5640.c:2825:1-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt286.c:927:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:930:5-8: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:299:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:302:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt274.c:958:6-8: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:961:6-9: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:384:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:387:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/cs4271.c:226:2-16: WARNING: Assignment of bool to 0/1
sound/soc/codecs/cs4271.c:229:2-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/max98373.c:411:2-20: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98373.c:922:2-27: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98373.c:924:2-27: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some mux/mixer are not used. Remove them from the driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5645_if3_adc_in_mux, rt5645_inr_mux, and rt5645_inl_mux are not used.
Remove them from the driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Missing or spurious parameter descriptions. Fix warnings with W=1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix warnings with W=1
If these variables are useful this driver should be modified to expose
them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix warnings with W=1
If these variables are useful then this driver should be modified to
expose them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No reason why this is global, fix warnings with W=1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AK4497 is a 32-bit 2ch DAC and has the same register
map as AK4458 with few exceptions:
* AK4497 has one more register at the end of register space
DFS_READ which is a read only register that allows users
to read FS Auto Detection mode. We currently do not use
this register so we use the same regmap structure as for ak4458.
* Because AK4458 is an 8ch DAC there are some fields that are
only used by AK4458 and marked as reserved for AK4497, so for
this reason we need to have a distinct set of controls, widgets
and routes.
Datasheet for AK4497 is at:
https://www.akm.com/akm/en/file/ev-board-manual/AK4497EQ.pdf
Datasheet for AK4458 is at:
https://www.akm.com/akm/en/file/datasheet/AK4458VN.pdf
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Export the DAC functionality to mix left + right together and then output
the same (mixed) signal on both outputs.
Various (x86) tablets with an ES8316 codec use a single speaker
connected between the headhpone LOUT and ROUT pins, expecting the output
to be in a mono differential mode. Presumably this is done to use the
power of both the left and right outputs to allow the speaker to be
louder.
The ES8316 codec does not have a differential output mode, but we can
emulate this by making both channels output the same through the mono mix
switch, combined with setting the Playback Polarity control to "R Invert",
which applias a 180 degrees phase inversion to the right channel.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adding jack-detect support may seem weird for a codec with only
a single output, but it is necessary. The ES8316 appnote showing
the intended usage uses a jack-receptacle which physically disconnects
the speakers from the output when a jack is plugged in.
But all 3 devices using the es8316 which I have (2 Cherry Trail
devices and one Bay Trail CR device), use an analog mux to disconnect
the speakers, driven by a GPIO. In order to enable/disable the speakers
at the right time, we need jack-detect.
The same goes for the microphone where we must correctly set the mux
for the single ADC to either the internal or the headset microphone.
All devices I have support the es8316's builtin jack-detect functionality.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Inside function rt274_i2c_probe(), if regmap_read() function
returns -EINVAL, then local variable "val" leaves uninitialized
but used in if statement. This is potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds reset and precharge in shutdown of PCM device.
ACODEC goes to silence if we change Fs to 44.1kHz from 48kHz. This
workaround seems to work but I don't know this workaround is correct
sequence or not for ACODEC.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for audio CODEC core of rk3328.
Rockchip does not publish detail specification of this core
but driver source code is opened on their GitHub repository.
https://github.com/rockchip-linux/kernel
So I ported this code to linux-next and added some trivial fixes.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
MCLK input is needed when accessing any register after enabling SYSCLK.
This also fixes imbalance of clk_enable / clk_disable when transitioning
between ON -> STANDBY -> ON bias levels.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Save 2x unsigned int of .rodata.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For platforms that use the audio-graph-card driver, the codec is
not selected by SoC-platform driver. Make it available.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Cirrus Logic CS4341.
This is a very simple, playback only, stereo DAC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC mixer setting needs to restore to default value
after calibration.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We accidentally call mutex_unlock(&pcm512x->mutex); twice in a row.
I re-wrote the error handling to use "goto unlock;" instead of returning
directly. Hopefully, it makes the code a little simpler.
Fixes: 3500f1c589 ("ASoC: pcm512x: Implement the digital_mute interface")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviwed-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
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Merge tag 'asoc-v4.21' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.21
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
Clicks and pops of various volumes can be produced while the device is
opened, closed, put into and taken out of standby, or reconfigured.
Fix this, by implementing the digital_mute interface, so that the
output is muted during such operations.
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Saravanan Sekar