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Merge series from Mario Limonciello <mario.limonciello@amd.com>:
It's been reported that a number of laptops have a low volume
level from the digital microphone compared to Windows.
AMD offers a register that can adjust the gain for PDM which is not
configured at maximum gain by default.
To fix this change the default for all 3 drivers to raise the gain
but also offer a module parameter. The module parameter can be used
for debugging if the gain is too high on a given laptop.
This is intentionally split into multiple patches for default and
parameter so that if the default really does behave better universally
we can bring it back to stable too later.
In case of regressions for any users that the new pdm_gain value is
too high and for additional debugging, introduce a module parameter
that would let them configure it.
This parameter should be removed in the future:
* If it's determined that the parameter is not needed, just hardcode
the correct value as before
* If users do end up using it to debug and report different values
we should introduce a config knob that can have policy set by ucm.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-7-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
No issues have been reported yet for DMIC audio level on ps platforms,
but as problems were found both on YC (Rembrandt) and Renoir based
designs it's very likely they happen on ps too.
Increase the PDM gain to solve this problem.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-6-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of regressions for any users that the new pdm_gain value is
too high and for additional debugging, introduce a module parameter
that would let them configure it.
This parameter should be removed in the future:
* If it's determined that the parameter is not needed, just hardcode
the correct value as before
* If users do end up using it to debug and report different values
we should introduce a config knob that can have policy set by ucm.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-5-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A similar issue that was reported on Rembrandt based laptops with
low DMIC volume is also being reported for Barcelo based laptops
that use renoir acp3x.
Increase the PDM gain to overcome this problem.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-4-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of regressions for any users that the new pdm_gain value is
too high and for additional debugging, introduce a module parameter
that would let them configure it.
This parameter should be removed in the future:
* If it's determined that the parameter is not needed, just hardcode
the correct value as before
* If users do end up using it to debug and report different values
we should introduce a config knob that can have policy set by ucm.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-3-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A number of users for Lenovo Rembrandt based laptops are
reporting that the microphone is too quiet relative to
Windows with a dual boot.
Increase the PDM gain to overcome this problem.
Signed-off-by: Mario Limonciello <mario.limonciello@amd.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230131184653.10216-2-mario.limonciello@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Using the control_unload for graph type of elem will lead surprises on
module unload.
The correct callback to use is the dapm_route_unload.
Fixes: 31e9273912bf ("ASoC: topology: Use unload() op directly")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230201112846.27707-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The conversion to use generic helpers missed the else for the dai
direction check which leads to failure when loading playback widgets
Fixes: 323f09a61d43 ("ASoC: sof: use helper function")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20230201112846.27707-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A bit higher volume of changes than wished, but each change is
relatively small and the fix targets are mostly device-specific,
so those should be safe as a late stage merge.
The most significant LoC is about the memalloc helper fix, which
is applied only to Xen PV. The other major parts are ASoC Intel
SOF and AVS fixes that are scattered as various small code
changes. The rest are device-specific fixes and quirks for HD-
and USB-audio, FireWire and ASoC AMD / HDMI.
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Merge tag 'sound-6.2-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A bit higher volume of changes than wished, but each change is
relatively small and the fix targets are mostly device-specific, so
those should be safe as a late stage merge.
The most significant LoC is about the memalloc helper fix, which is
applied only to Xen PV. The other major parts are ASoC Intel SOF and
AVS fixes that are scattered as various small code changes. The rest
are device-specific fixes and quirks for HD- and USB-audio, FireWire
and ASoC AMD / HDMI"
* tag 'sound-6.2-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
ALSA: firewire-motu: fix unreleased lock warning in hwdep device
ALSA: memalloc: Workaround for Xen PV
ASoC: cs42l56: fix DT probe
ASoC: codecs: wsa883x: correct playback min/max rates
ALSA: hda/realtek: Add Acer Predator PH315-54
ASoC: amd: yc: Add Xiaomi Redmi Book Pro 15 2022 into DMI table
ALSA: hda: Do not unset preset when cleaning up codec
ASoC: SOF: sof-audio: prepare_widgets: Check swidget for NULL on sink failure
ASoC: hdmi-codec: zero clear HDMI pdata
ASoC: SOF: ipc4-mtrace: prevent underflow in sof_ipc4_priority_mask_dfs_write()
ASoC: Intel: sof_ssp_amp: always set dpcm_capture for amplifiers
ASoC: Intel: sof_nau8825: always set dpcm_capture for amplifiers
ASoC: Intel: sof_cs42l42: always set dpcm_capture for amplifiers
ASoC: Intel: sof_rt5682: always set dpcm_capture for amplifiers
ALSA: hda/via: Avoid potential array out-of-bound in add_secret_dac_path()
ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless
ALSA: hda/realtek: fix mute/micmute LEDs, speaker don't work for a HP platform
ASoC: SOF: keep prepare/unprepare widgets in sink path
ASoC: SOF: sof-audio: skip prepare/unprepare if swidget is NULL
ASoC: SOF: sof-audio: unprepare when swidget->use_count > 0
...
The ucb1400 MFD driver and its gpio and touchscreen child
drivers were only used on a few PXA machines that were unused
for a while and are now removed.
Removing these leaves the AC97 support as ALSA specific,
no other drivers are now connected through this interface.
Cc: Linus Walleij <linus.walleij@linaro.org>
Cc: Bartosz Golaszewski <brgl@bgdev.pl>
Cc: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Cc: Lee Jones <lee@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Marek Vasut <marex@denx.de>
Cc: linux-kernel@vger.kernel.org
Cc: linux-gpio@vger.kernel.org
Cc: linux-input@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Most PXA/MMP boards were removed, so the board specific ASoC
support is no longer needed, leaving only support for DT
based ones, as well as the "gumstix" and "spitz" machines
that may get converted to DT later.
Cc: Ian Molton <spyro@f2s.com>
Cc: Ken McGuire <kenm@desertweyr.com>
Cc: Marek Vasut <marek.vasut@gmail.com>
Cc: Mike Rapoport <rppt@kernel.org>
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: alsa-devel@alsa-project.org
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Both the DAC and ADC have digital gain controls available
for their mixers, which go from -31 to 0db by step of 1db.
Signed-off-by: Christophe Branchereau <cbranchereau@gmail.com>
Link: https://lore.kernel.org/r/20230122210703.2552384-1-cbranchereau@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Samsung Galaxy Book2 Pro 360 (13" 2022 NP930QED-KA1FR) with codec SSID
144d:ca03 requires the same workaround for enabling the speaker amp
like other Samsung models with ALC298 codec.
Cc: <stable@vger.kernel.org>
Signed-off-by: Guillaume Pinot <texitoi@texitoi.eu>
Link: https://lore.kernel.org/r/20230129171338.17249-1-texitoi@texitoi.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This loop accidentally reuses the "i" iterator for both the inside and
the outside loop. The value of MAX_STREAM_BUFFER is 5. I believe that
chip->rmh.stat_len is in the 2-12 range. If the value of .stat_len is
4 or more then it will loop exactly one time, but if it's less then it
is a forever loop.
It looks like it was supposed to combined into one loop where
conditions are checked.
Fixes: 8e6320064c33 ("ALSA: lx_core: Remove useless #if 0 .. #endif")
Signed-off-by: Dan Carpenter <error27@gmail.com>
Link: https://lore.kernel.org/r/Y9jnJTis/mRFJAQp@kili
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is tested on V4H White Hawk + ARD-AUDIO-DA7212
Signed-off-by: Linh Phung <linh.phung.jy@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o7qe5ej5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch tidyups rsnd_dma_probe(), but there is no effect.
This is prepare for Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r0va5elq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ADG need to know output rate of 44.1kHz/48kHz.
It is using single variable for each, but this patch changes
it to array. Nothing is changed by this patch.
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tu065em3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg.c is assuming number of clkin/clkout are fixed, but it is
not correct on Gen4. This patch uses clkin/out_size to handling it.
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v8km5em7.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves clkout_name to top of the file to handling both
clkin/clkout in the same place.
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87wn525emc.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg.c is usig "clk" as clock IN, but is using "clkout" for
clock OUT. This patch arranges "clk" to "clkin".
This is prepare for R-Car Gen4 support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y1pi5emh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The flag LRCLK_ASYNC / AUDIO_OUT_48 had been added to handling
special case of Salvator-X board, but it is not used on upstream.
It makes code complex today, let's remove these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zg9y5emm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some SoC can't handle all requested hw rule. In such case, it will indicate
like below, but it is unclear why it didn't work to user.
This patch indicates warning in such case once, because player will try to
similar rule many times.
# aplay sound.wav
Playing WAVE 'sound.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
aplay: aplay.c: 1359: set_params: Assertion `err >= 0' failed.
Aborted by signal Aborted...
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87357q6t7b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_ssi_master_clk_start() indicates error message if it couldn't
handle requested clock/rate, but it is not caring all cases.
This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874js66t7g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit b43b8ae87c8e0a8 ("ASoC: rsnd: protect mod->status") removed
RSND_DEBUG_NO_DAI_CALL and rsnd_dbg_dai_call(), but these are still
exist on rsnd.h. This patch removes it.
Fixes: b43b8ae87c8e ("ASoC: rsnd: protect mod->status")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875ycm6t7l.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 1f9c82b5ab83ff2 ("ASoC: rsnd: add debugfs support") added
CONFIG_DEBUG_FS related definitions on rsnd.h, but it should be
added inside of RSND_H. This patch fixup it.
Fixes: 1f9c82b5ab83 ("ASoC: rsnd: add debugfs support")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877cx26t7r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd sets "channels_min" which is used from
snd_soc_dai_stream_valid() without checking DT playback/capture property.
Thus, "aplay -l" or "arecord -l" will indicate un-exising device.
This patch checks DT proerty and do nothing playback/capture settings if
not exist.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878rhi6t7x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Set driver name to allow matching different UCM2 configurations
for the multiple devices sharing the same APQ8096 ASoC.
Signed-off-by: Yassine Oudjana <y.oudjana@protonmail.com>
Link: https://lore.kernel.org/r/20220622061106.35071-1-y.oudjana@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
The CS42L42 has a SoundWire interface for control and audio. This
chain of patches adds support for this.
Patches #1 .. #5 split out various changes to the existing code that
are needed for adding Soundwire. These are mostly around clocking and
supporting the separate probe and enumeration stages in SoundWire.
Patches #6 .. #8 actually adds the SoundWire handling.
Merge series from Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>:
struct snd_soc_dai need to have info for playback/capture,
but it is using "playback/capture_xxx" or "tx/tx_xxx" or array.
This kind of random definition is very difficult to read.
This patch-set add helper functions and each driver use it.
And cleanup the definition.
Merge series from Claudiu Beznea <claudiu.beznea@microchip.com>:
This series adds runtime PM support for Microchip SPDIFRX driver.
Along with it I added few fixes identified while going though the code
and playing with Microchip SPDIFRX controller.
Merge series from wangweidong.a@awinic.com:
The Awinic AW88395 is an I2S/TDM input, high efficiency
digital Smart K audio amplifier with an integrated 10.25V
smart boost converter.
Add a DT schema for describing Awinic AW88395 audio amplifiers. They are
controlled using I2C
Merge series from Herve Codina <herve.codina@bootlin.com>:
The Renesas IDT821034 codec is four channel PCM codec with on-chip
filters and programmable gain setting. It also provides SLIC
(Subscriber Line Interface Circuit) signals as GPIOs.
Since clock stop causes bus reset on Intel controllers, we need
to wait for the debounce interval on resume, to ensure all the
interrupt status registers are set correctly.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-9-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
idle_bias_on was set because cs42l42 has a "VMID" type pseudo-midrail
supply (named FILT+), and these typically take a long time to charge.
But the driver never enabled pm_runtime so it would never have powered-
down the cs42l42 anyway.
In fact, FILT+ can charge to operating voltage within 12.5 milliseconds
of enabling HP or ADC. This time is already covered by the startup
delay of the HP/ADC.
The datasheet warning about FILT+ taking up to 1 second to charge only
applies in the special cases that either the PLL is started or
DETECT_MODE set to non-zero while both HP and ADC are off. The driver
never does either of these.
Removing idle_bias_on allows the Soundwire host controller to suspend
if there isn't a snd_soc_jack handler registered.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-8-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for using CS42L42 as a SoundWire device.
SoundWire-specifics are kept separate from the I2S implementation as
much as possible, aiming to limit the risk of breaking the I2C+I2S
support.
There are some important differences in the silicon behaviour between
I2S and SoundWire mode that are reflected in the implementation:
- ASP (I2S) most not be used in SoundWire mode because the two interfaces
share pins.
- The SoundWire capture (record) port only supports 1 channel. It does
not have left-to-right duplication like the ASP.
- DP2 can only be prepared if the HP has powered-up. DP1 can only be
prepared if the ADC has powered-up. (This ordering restriction does
not exist for ASPs.) The SoundWire core port-prepare step is
triggered by the DAI-link prepare(). This happens before the
codec DAI prepare() or the DAPM sequence so these cannot be used
to enable HP/ADC. Instead the HP/ADC enable/disable are done during
the port_prep callback.
- The SRCs are an integral part of the audio chain but in silicon their
power control is linked to the ASP. There is no equivalent power link
to SoundWire DPs so the driver must take "manual" control of SRC power.
- The SoundWire control registers occupy the lower part of the SoundWire
address space so cs42l42 registers are offset by 0x8000 (non-paged) in
SoundWire mode.
- Register addresses are 8-bit paged in I2C mode but 16-bit unpaged in
SoundWire.
- Special procedures are needed on register read/writes to (a) ensure
that the previous internal bus transaction has completed, and
(b) handle delayed read results, when the read value could not be
returned within the SoundWire read command.
There are also some differences in driver implementation between I2S
and SoundWire operation:
- CS42L42 I2S does not runtime_suspend, but runtime_suspend/resume support
has been added into the driver in SoundWire mode as the most convenient
way to power-up the bus manager and to handle the unattach_request
condition, though the CS42L42 chip does not itself suspend or resume.
- Intel SoundWire host controllers have a low-power clock-stop mode that
requires resetting all peripherals when resuming. This means that the
interrupt registers will be reset in between the interrupt being
generated and the interrupt being handled, and since the interrupt
status is debounced, these values may not be accurate immediately,
and may cause spurious unplug events before settling.
- As in I2S mode, the PLL is only used while audio is active because
of clocking quirks in the silicon. For SoundWire the cs42l42_pll_config()
is deferred until the DAI prepare(), to allow the cs42l42_bus_config()
callback to set the SCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-7-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Export functions that will be needed by a SoundWire module.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-6-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Setup of the ASP (audio serial port) was being done as a side-effect of
cs42l42_pll_config() and forces a restriction on the ratio of sample_rate
to bit_clock that is invalid for Soundwire.
Move the ASP setup into a dedicated function.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-5-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The chosen clocking configuration must give an internal MCLK (MCLKint)
that is an integer multiple of the sample rate.
On I2S each of the supported bit clock frequencies can only be generated
from one sample rate group (either the 44100 or the 48000) so the code
could use only the bitclock to look up a PLL config.
The relationship between sample rate and bitclock frequency is more
complex on Soundwire and so it is possible to set a frame shape to
generate a bitclock from the "wrong" group. For example 2*147 with a
48000 sample rate would give a bitclock of 14112000 which on I2S
could only be derived from a 44100 sample rate.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-4-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SOFT_RESET_REBOOT register is needed to recover CS42L42 state after
a Soundwire bus reset.
This is required to be set whenever there is severe/hard bus reset.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_dai has data for Playback/Capture, but it is very
random. Someone is array (A), someone is playback/capture (B),
and someone is tx/rx (C);
struct snd_soc_dai {
...
(A) unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1];
(B) struct snd_soc_dapm_widget *playback_widget;
(B) struct snd_soc_dapm_widget *capture_widget;
(B) void *playback_dma_data;
(B) void *capture_dma_data;
...
(C) unsigned int tx_mask;
(C) unsigned int rx_mask;
};
Because of it, the code was very complicated.
This patch creates new data structure to merge these into one,
and tidyup the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87cz6vea1v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has many helper function.
This patch use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87fsbrea25.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has many helper function.
This patch use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87ilgnea2p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
