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Remove the check of fw_patched from cs35l56_is_fw_reload_needed().
Also remove the redundant check for control of the reset GPIO.
The fw_patched flag is set when cs35l56_dsp_work() has completed its
steps to download firmware and power-up wm_adsp. There was a check in
cs35l56_is_fw_reload_needed() to make a quick exit of 'false' if
!fw_patched. The original idea was that the system might be suspended
before the driver has ever made any attempt to download firmware, and
in that case the driver doesn't need to return to a patched state
because it was never in a patched state.
This check of fw_patched is buggy because it prevented ever recovering
from a failed patch. If a previous attempt to patch and reboot the
silicon had failed it would leave fw_patched==false. This would mean
the driver never attempted another download even though the fault may
have been cleared (by a hard reset, for example).
It is also a redundant check because the calling code already makes
a quick exit if cs35l56_component_probe() has not been called, which
deals with the original intent of this check but in a safer way.
The check for reset GPIO is redundant: if the silicon was hard-reset
the FIRMWARE_MISSING flag will be 1. But this check created an
expectation that the suspend/resume code toggles reset. This can't
easily be protected against accidental code breakage. The only reason
for the check was to skip runtime-resuming the driver to read the
PROTECTION_STATUS register when it already knows it reset the silicon.
But in that case the driver will have to be runtime-resumed to do
the firmware download. So it created an assumption for no benefit.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240129162737.497-7-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Move the call to cs35l56_set_patch() earlier in cs35l56_init() so
that it only adds the register patch on first-time initialization.
The call was after the post_soft_reset label, so every time this
function was run to re-initialize the hardware after a reset it would
call regmap_register_patch() and add the same reg_sequence again.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 898673b905b9 ("ASoC: cs35l56: Move shared data into a common data structure")
Link: https://msgid.link/r/20240129162737.497-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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cs35l56_component_remove() must call wm_adsp_power_down() and
wm_adsp2_component_remove().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The cs35l56->component pointer is used by the suspend-resume handling to
know whether the driver is fully instantiated. This is to prevent it
queuing dsp_work which would result in calling wm_adsp when the driver
is not an instantiated ASoC component. So this pointer must be cleared
by cs35l56_component_remove().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There's no need to overwrite fwf_name with a kstrdup() of the cs_dsp part
name. It is trivial to select either fwf_name or cs_dsp.part as the string
to use when building the filename in wm_adsp_request_firmware_file().
This leaves fwf_name entirely owned by the codec driver.
It also avoids problems with freeing the pointer. With the original code
fwf_name was either a pointer owned by the codec driver, or a kstrdup()
created by wm_adsp. This meant wm_adsp must free it if it set it, but not
if the codec driver set it. The code was handling this by using
devm_kstrdup().
But there is no absolute requirement that wm_adsp_common_init() must be
called from probe(), so this was a pseudo-memory leak - each new call to
wm_adsp_common_init() would allocate another block of memory but these
would only be freed if the owning codec driver was removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240129162737.497-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Check for the cases of system-specific bin file without a
wmfw before falling back to looking for a generic wmfw.
All system-specific options should be tried before falling
back to loading a generic wmfw/bin. With the original code,
the presence of a fallback generic wmfw on the filesystem
would prevent using a system-specific tuning with a ROM
firmware.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 0e7d82cbea8b ("ASoC: wm_adsp: Add support for loading bin files without wmfw")
Link: https://msgid.link/r/20240129162737.497-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
Quite a lot of fixes that came in since the merge window, a large
portion for for Qualcomm and ES8326.
The 8 DAI support for Qualcomm is just raising a constant to allow for
devies that otherwise only need DTs, and there's a few other device ID
updates for sunxi (Allwinner) and AMD platforms.
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There currently exists two thinkpad headset jack fixups:
ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK
ALC285_FIXUP_THINKPAD_HEADSET_JACK
The latter is applied to alc285 and alc287 thinkpads which contain
bass speakers.
However, the former was only being applied to alc285 thinkpads,
leaving non-bass alc287 thinkpads with no headset button controls.
This patch fixes that by adding ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK
to the alc287 chains, allowing the detection of headset buttons.
Signed-off-by: José Relvas <josemonsantorelvas@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240131113407.34698-3-josemonsantorelvas@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add 4 missing formats to 'snd_pcm_format_names' array in order to be
able to get their names with 'snd_pcm_format_name' function.
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Link: https://lore.kernel.org/r/20240125223522.1122765-1-ivan.orlov0322@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adds sound support for ASUS Zenbook UM3402YAR with missing DSD
Signed-off-by: Chhayly Leang <clw.leang@gmail.com>
Link: https://lore.kernel.org/r/20240126080912.87422-1-clw.leang@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add new model entry into configuration table.
Signed-off-by: Kenzo Gomez <kenzo.sgomez@gmail.com>
Link: https://lore.kernel.org/r/20240127164621.26431-1-kenzo.sgomez@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Previous commit that added support for Huawei MateBook D16 2021
with Ryzen 4600H (HVY-WXX9 M1010) was incomplete.
To activate support for this laptop, the DMI table in
acp3x-es83xx machine driver must also be updated.
Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop")
Signed-off-by: Marian Postevca <posteuca@mutex.one>
Link: https://msgid.link/r/20240128172229.657142-1-posteuca@mutex.one
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge series from Chen-Yu Tsai <wens@kernel.org>:
This series adds SPDIF controllers for the H616 and H618.
There's also a fix for SPDIF on H6: the controller also has a
receiver that was not correctly modeled.
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The SPDIF hardware block found in the H616 SoC has the same layout as
the one found in the H6 SoC, except that it is missing the receiver
side.
Since the driver currently only supports the transmit function, support
for the H616 is identical to what is currently done for the H6.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://msgid.link/r/20240127163247.384439-4-wens@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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The laptop requires a quirk ID to enable its internal microphone. Add
it to the DMI quirk table.
Reported-by: Techno Mooney <techno.mooney@gmail.com>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218402
Cc: stable@vger.kernel.org
Signed-off-by: Techno Mooney <techno.mooney@gmail.com>
Signed-off-by: Bagas Sanjaya <bagasdotme@gmail.com>
Link: https://msgid.link/r/20240129081148.1044891-1-bagasdotme@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Many devices with a single alternate setting do not have a Valid
Alternate Setting Control and validation performed by
validate_sample_rate_table_v2v3() doesn't work on them and is not
really needed. So check the presense of control before sending
altsetting validation requests.
MOTU Microbook IIc is suffering the most without this check. It
takes up to 40 seconds to bootup due to how slow it switches
sampling rates:
[ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60
[ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0
[ 2659.164829] usb 3-2: Product: MicroBook IIc
[ 2659.164830] usb 3-2: Manufacturer: MOTU
[ 2659.166204] usb 3-2: Found last interface = 3
[ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug
[ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3
[ 2679.322321] usb 3-2: Creating new data endpoint #3
[ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1
[ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110
[ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug
[ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84
[ 2694.444721] usb 3-2: Creating new data endpoint #84
[ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240129121254.3454481-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This reverts commit 67794f882adca00d043899ac248bc002751da9f6.
We need to explicitly set up the clock selector to workaround a problem
with the Behringer mixers. This was originally done in d2e8f641257d
("ALSA: usb-audio: Explicitly set up the clock selector")
The problem with MOTU M Series mentioned in commit message was fixed in
a different way by checking control capabilities of clock selectors.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240128132338.819273-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This HP Laptop uses ALC236 codec with COEF 0x07 controlling the
mute LED. Enable existing quirk for this device.
Signed-off-by: Luka Guzenko <l.guzenko@web.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240128155704.2333812-1-l.guzenko@web.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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1 SF114-32
If you connect an external headset/microphone to the 3.5mm jack on the
Acer Swift 1 SF114-32 it does not recognize the microphone. This fixes
that and gives the user the ability to choose between internal and
headset mic.
Signed-off-by: David Senoner <seda18@rolmail.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240126155626.2304465-1-seda18@rolmail.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Clock selector control might be read-only. Add corresponding checks
to prevent sending control requests that would fail.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240125205457.28258-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The quirk table entries should be put in the USB ID order, but some
entries have been put in random places. Re-sort them.
Fixes: bf990c102319 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue")
Fixes: fd28941cff1c ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless")
Fixes: dfd5fe19db7d ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless")
Fixes: 4a63e68a2951 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.")
Fixes: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Fixes: 4fb7c24f69c4 ("ALSA: usb-audio: Add quirk for Fiero SC-01")
Fixes: 2307a0e1ca0b ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)")
Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The RODE NT-USB+ is marketed as a professional usb microphone, however the
usb audio interface is a mess:
[ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd
[ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4
[ 1.503912] usb 1-5: config 1 has no interface number 4
[ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09
[ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ 1.519697] usb 1-5: Product: RØDE NT-USB+
[ 1.519699] usb 1-5: Manufacturer: RØDE
[ 1.519700] usb 1-5: SerialNumber: 1D773A1A
[ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82
[ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82
[ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2
Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get.
I have asked Rode support to fix it, but they show no interest.
Signed-off-by: Sean Young <sean@mess.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601
Cc: <stable@vger.kernel.org>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240124130239.358298-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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virtqueue_enable_cb() will call virtqueue_poll() which will check if
queue is broken at beginning, so remove the virtqueue_is_broken() call
Signed-off-by: Li RongQing <lirongqing@baidu.com>
Reviewed-by: Stefan Hajnoczi <stefanha@redhat.com>
Link: https://lore.kernel.org/r/20240124120834.49410-1-lirongqing@baidu.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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fix typo in midi fallback log.
Signed-off-by: Jacob Siverskog <jacob@teenage.engineering>
Fixes: ff49d1df79ae ("ALSA: usb-audio: USB MIDI 2.0 UMP support")
Link: https://lore.kernel.org/r/20240124101827.35433-1-jacob@teenage.engineering
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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SSID 0x0c0d platform. It can't mute speaker when HP plugged.
This patch add quirk to fill speaker pin verbtable.
And disable speaker passthrough.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/38b82976a875451d833d514cee34ff6a@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since commit 086b957cc17f5 ("ALSA: usb-audio: Skip the clock selector
inquiry for single connections") we are already skipping clock selector
inquiry if only one clock source is connected, but we are still sending
a set request. Lets skip that too.
This should fix errors when setting a sample rate on devices that don't
have any controls present within the clock selector. An example of such
device is the new revision of MOTU M Series (07fd:000b):
AudioControl Interface Descriptor:
bLength 8
bDescriptorType 36
bDescriptorSubtype 11 (CLOCK_SELECTOR)
bClockID 1
bNrInPins 1
baCSourceID(0) 2
bmControls 0x00
iClockSelector 0
Perhaps we also should check if clock selectors are readable and writeable
like we already do for clock sources, but this is out of scope of this
patch.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240123134635.54026-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The device fails to initialize otherwise, giving the following error:
[ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1
Signed-off-by: Julian Sikorski <belegdol+github@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240123084935.2745-1-belegdol+github@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Customer has reported an issue with specific desktop platform
where two CS42L42 codecs are connected to CS8409 HDA bridge.
If "Master Volume Control" is created then on Ubuntu OS UCM
left/right balance slider in UI audio settings has no effect.
This patch will fix this issue for a target paltform.
Fixes: 20e507724113 ("ALSA: hda/cs8409: Add support for dolphin")
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240122184710.5802-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Apollo Lake seems to also suffer from IRQ timing issues. After being up for ~4
minutes, a Pentium N4200 system ends up falling back to workqueue-based IRQ
handling:
[ 208.019906] snd_hda_intel 0000:00:0e.0: IRQ timing workaround is activated
for card #0. Suggest a bigger bdl_pos_adj.
Unfortunately, the Baytrail and Braswell workaround value of 32 samples isn't
enough to fix the issue here. Default to 64 samples.
Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240122114512.55808-3-rsalvaterra@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We have self-explanatory constants for Intel HDA devices, let's use them instead
of magic numbers and code comments.
Signed-off-by: Rui Salvaterra <rsalvaterra@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240122114512.55808-2-rsalvaterra@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge series from Johan Hovold <johan+linaro@kernel.org>:
To reduce the risk of speaker damage the PA gain needs to be limited on
machines like the Lenovo Thinkpad X13s until we have active speaker
protection in place.
Limit the gain to the current default setting provided by the UCM
configuration which most user have so far been using (due to a bug in
the configuration files which prevented hardware volume control [1]).
The wsa883x PA volume control also turned out to be broken, which meant
that the default setting used by UCM configuration is actually the
lowest level (-3 dB). With the codec driver fixed, hardware volume
control also works as expected.
Note that the new wsa884x driver most likely suffers from a similar bug,
I'll send a fix for that once I've got that confirmed.
Included is also a related fix for the LPASS WSA macro driver, which
was changing the digital gain setting behind the back of user space and
which can result in excessive (or too low) digital gain.
There are further Qualcomm codec drivers that similarly appear to
manipulate various gain settings, but on closer inspection it turns out
that they only write back the current settings. Tests reveal that these
writes are indeed needed for any prior updates to take effect (at least
for the WSA and RX macros).
[1] https://github.com/alsa-project/alsa-ucm-conf/pull/382
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Merge series from Zhu Ning <zhuning0077@gmail.com>:
We get some issues regarding crosstalk, THD+N performance and pop
noise from customer's project.
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The UCM configuration for the Lenovo ThinkPad X13s has up until now
been setting the speaker PA volume to the minimum -3 dB when enabling
the speakers, but this does not prevent the user from increasing the
volume further.
Limit the digital gain and PA volumes to a combined -3 dB in the machine
driver to reduce the risk of speaker damage until we have active speaker
protection in place (or higher safe levels have been established).
Note that the PA volume limit cannot be set lower than 0 dB or
PulseAudio gets confused when the first 16 levels all map to -3 dB.
Also note that this will probably need to be generalised using
machine-specific limits, but a common limit should do for now.
Cc: <stable@vger.kernel.org> # 6.5
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240122181819.4038-3-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Remove the executable bit that was unintentionally turned on.
Fixes: ee09084fbf9f ("ASoC: codecs: ES8326: Add chip version flag")
Signed-off-by: Fei Shao <fshao@chromium.org>
Link: https://msgid.link/r/20240122062055.1673597-1-fshao@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Sound card on Qualcomm X1E80100 CRD board will use eight DAIs in one DAI
link, so increase the limit.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117160144.1305127-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Driver does not use any regulator code, so drop redundant include of
regulator/consumer.h header.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-3-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Probe calls wcd938x_populate_dt_data() which already prints all the
error cases with dev_err_probe(), so skip the additional dev_err().
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-2-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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WCD938x sound codec driver ignores return status of getting regulators
and returns EINVAL instead of EPROBE_DEFER. If regulator provider
probes after the codec, system is left without probed audio:
wcd938x_codec audio-codec: wcd938x_probe: Fail to obtain platform data
wcd938x_codec: probe of audio-codec failed with error -22
Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add condition check for cpu dai link initialization for amplifier
codec path, as same pcm id uses for both headset and speaker path
for RENOIR platforms.
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://msgid.link/r/20240118143023.1903984-3-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Update platform name for various boards based on rembrandt
and renoir platforms.
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://msgid.link/r/20240118143023.1903984-2-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Set and enable rt5682s codec bclk and lrclk rates when
acp is in slave mode.
Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://msgid.link/r/20240118143023.1903984-1-venkataprasad.potturu@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The lowest headphones volume setting does not mute so the leave the TLV
mute flag unset.
This is specifically needed to let the sound server use the lowest gain
setting.
Fixes: c03226ba15fe ("ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR")
Cc: <stable@vger.kernel.org> # 6.5
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240122091130.27463-1-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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The vendor driver appears to be modifying the gain settings behind the
back of user space but these hacks never made it upstream except for
some essentially dead code that adds a constant zero to the current gain
setting on DAPM events.
Note that the volume registers still need to be written after enabling
clocks in order for any prior updates to take effect.
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-5-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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The LPASS WSA macro codec driver is updating the digital gain settings
behind the back of user space on DAPM events if companding has been
enabled.
As compander control is exported to user space, this can result in the
digital gain setting being incremented (or decremented) every time the
sound server is started and the codec suspended depending on what the
UCM configuration looks like.
Soon enough playback will become distorted (or too quiet).
This is specifically a problem on the Lenovo ThinkPad X13s as this
bypasses the limit for the digital gain setting that has been set by the
machine driver.
Fix this by simply dropping the compander gain offset hack. If someone
cares about modelling the impact of the compander setting this can
possibly be done by exporting it as a volume control later.
Note that the volume registers still need to be written after enabling
clocks in order for any prior updates to take effect.
Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Cc: stable@vger.kernel.org # 5.11
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-4-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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The PA gain can be set in steps of 1.5 dB from -3 dB to 18 dB, that is,
in 15 levels.
Fix the dB values for the PA volume control as experiments using wsa8835
show that the first 16 levels all map to the same lowest gain while the
last three map to the highest gain.
These values specifically need to be correct for the sound server to
provide proper volume control.
Note that level 0 (-3 dB) does not mute the PA so the mute flag should
also not be set.
Fixes: cdb09e623143 ("ASoC: codecs: wsa883x: add control, dapm widgets and map")
Cc: stable@vger.kernel.org # 6.0
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-2-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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We get a noise issue during the startup of recording. We update the
register setting and dapm widgets to fix this issue.
we change callback type of es8326_mute function to mute_stream.
ES8326_ADC_MUTE is moved to es8326_mute function so it can
be turned on at last and turned off at first.
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240120101240.12496-6-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We modify the register settings to minimize headphone pop noise
during ES8326 power-up and music start/stop.
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240120101240.12496-5-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We update the values of some registers in the initialization
sequence in es8326_resume function to improve THD+N performance.
THD+N performance decreases if the output level on headphone is
close to full scale. So we change the register setting in
es8326_jack_detect_handler function to improve THD+N performance
if headphone pulgged. Also, the register setting should be restored
when the headset is unplugged
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240120101240.12496-3-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We change the crosstalk parameter in es8326_resume function
to improve crosstalk performance.
Adding crosstalk kcontrol to enhance the flexibility of crosstalk
debugging in machine.
Adding ES8326_DAC_CROSSTALK macro to declare the crosstalk register.
Signed-off-by: Zhu Ning <zhuning0077@gmail.com>
Link: https://msgid.link/r/20240120101240.12496-2-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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