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>
