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commit 77ce96543b03f437c6b45f286d8110db2b6622a3 upstream.
The local helper function to compare the given pair of cycle count
evaluates them. If the left value is less than the right value, the
function returns negative value.
If the safe cycle is less than the current cycle, it is the case of
cycle lost. However, it is not currently handled properly.
This commit fixes the bug.
Cc: <stable@vger.kernel.org>
Fixes: 705794c53b00 ("ALSA: firewire-lib: check cycle continuity")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218033026.72577-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit eba2eb2495f47690400331c722868902784e59de ]
snd_soc_card_get_kcontrol() must be holding a read lock on
card->controls_rwsem while walking the controls list.
Compare with snd_ctl_find_numid().
The existing function is renamed snd_soc_card_get_kcontrol_locked()
so that it can be called from contexts that are already holding
card->controls_rwsem (for example, control get/put functions).
There are few direct or indirect callers of
snd_soc_card_get_kcontrol(), and most are safe. Three require
changes, which have been included in this patch:
codecs/cs35l45.c:
cs35l45_activate_ctl() is called from a control put() function so
is changed to call snd_soc_card_get_kcontrol_locked().
codecs/cs35l56.c:
cs35l56_sync_asp1_mixer_widgets_with_firmware() is called from
control get()/put() functions so is changed to call
snd_soc_card_get_kcontrol_locked().
fsl/fsl_xcvr.c:
fsl_xcvr_activate_ctl() is called from three places, one of which
already holds card->controls_rwsem:
1. fsl_xcvr_mode_put(), a control put function, which will
already be holding card->controls_rwsem.
2. fsl_xcvr_startup(), a DAI startup function.
3. fsl_xcvr_shutdown(), a DAI shutdown function.
To fix this, fsl_xcvr_activate_ctl() has been changed to call
snd_soc_card_get_kcontrol_locked() so that it is safe to call
directly from fsl_xcvr_mode_put().
The fsl_xcvr_startup() and fsl_xcvr_shutdown() functions have been
changed to take a read lock on card->controls_rsem() around calls
to fsl_xcvr_activate_ctl(). While this is not very elegant, it
keeps the change small, to avoid this patch creating a large
collateral churn in fsl/fsl_xcvr.c.
Analysis of other callers of snd_soc_card_get_kcontrol() is that
they do not need any changes, they are not holding card->controls_rwsem
when they call snd_soc_card_get_kcontrol().
Direct callers of snd_soc_card_get_kcontrol():
fsl/fsl_spdif.c: fsl_spdif_dai_probe() - DAI probe function
fsl/fsl_micfil.c: voice_detected_fn() - IRQ handler
Indirect callers via soc_component_notify_control():
codecs/cs42l43: cs42l43_mic_shutter() - IRQ handler
codecs/cs42l43: cs42l43_spk_shutter() - IRQ handler
codecs/ak4118.c: ak4118_irq_handler() - IRQ handler
codecs/wm_adsp.c: wm_adsp_write_ctl() - not currently used
Indirect callers via snd_soc_limit_volume():
qcom/sc8280xp.c: sc8280xp_snd_init() - DAIlink init function
ti/rx51.c: rx51_aic34_init() - DAI init function
I don't have hardware to test the fsl/*, qcom/sc828xp.c, ti/rx51.c
and ak4118.c changes.
Backport note:
The fsl/, qcom/, cs35l45, cs35l56 and cs42l43 callers were added
since the Fixes commit so won't all be present on older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 209c6cdfd283 ("ASoC: soc-card: move snd_soc_card_get_kcontrol() to soc-card")
Link: https://lore.kernel.org/r/20240221123710.690224-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c14f09f010cc569ae7e2f6ef02374f6bfef9917e ]
Rewrite the handling of ASP1 TX mixer mux initialization to prevent a
deadlock during component_remove().
The firmware can overwrite the ASP1 TX mixer registers with
system-specific settings. This is mainly for hardware that uses the
ASP as a chip-to-chip link controlled by the firmware. Because of this
the driver cannot know the starting state of the ASP1 mixer muxes until
the firmware has been downloaded and rebooted.
The original workaround for this was to queue a work function from the
dsp_work() job. This work then read the register values (populating the
regmap cache the first time around) and then called
snd_soc_dapm_mux_update_power(). The problem with this is that it was
ultimately triggered by cs35l56_component_probe() queueing dsp_work,
which meant that it would be running in parallel with the rest of the
ASoC component and card initialization. To prevent accessing DAPM before
it was fully initialized the work function took the card mutex. But this
would deadlock if cs35l56_component_remove() was called before the work job
had completed, because ASoC calls component_remove() with the card mutex
held.
This new version removes the work function. Instead the regmap cache and
DAPM mux widgets are initialized the first time any of the associated ALSA
controls is read or written.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Link: https://lore.kernel.org/r/20240208123742.1278104-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f6c967941c5d6fa526fdd64733a8d86bf2bfab31 ]
Put the silicon revision and secured flag in the wm_adsp fwf_name
string instead of including them in the part string.
This changes the format of the firmware name string from
cs35l56[s]-rev-misc[-system_name]
to
cs35l56-rev[-s]-misc[-system_name]
No firmware files have been published, so this doesn't cause a
compatibility break.
Silicon revision and secured flag are included in the firmware
filename to pick a firmware compatible with the part. These strings
were being added to the part string, but that is a misuse of the
string. The correct place for these is the fwf_name string, which
is specifically intended to select between multiple firmware files
for the same part.
Backport note:
This won't apply to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 608f1b0dbdde ("ASoC: cs35l56: Move DSP part string generation so that it is done only once")
Link: https://msgid.link/r/20240129162737.497-12-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 07f7d6e7a124d3e4de36771e2a4926d0e31c2258 ]
Defer initializing the state of the ASP1 mixer registers until
the firmware has been downloaded and rebooted.
On a SoundWire system the ASP is free for use as a chip-to-chip
interconnect. This can be either for the firmware on multiple
CS35L56 to share reference audio; or as a bridge to another
device. If it is a firmware interconnect it is owned by the
firmware and the Linux driver should avoid writing the registers.
However, if it is a bridge then Linux may take over and handle
it as a normal codec-to-codec link. Even if the ASP is used
as a firmware-firmware interconnect it is useful to have
ALSA controls for the ASP mixer. They are at least useful for
debugging.
CS35L56 is designed for SDCA and a generic SDCA driver would
know nothing about these chip-specific registers. So if the
ASP is being used on a SoundWire system the firmware sets up the
ASP mixer registers. This means that we can't assume the default
state of these registers. But we don't know the initial state
that the firmware set them to until after the firmware has been
downloaded and booted, which can take several seconds when
downloading multiple amps.
DAPM normally reads the initial state of mux registers during
probe() but this would mean blocking probe() for several seconds
until the firmware has initialized them. To avoid this, the
mixer muxes are set SND_SOC_NOPM to prevent DAPM trying to read
the register state. Custom get/set callbacks are implemented for
ALSA control access, and these can safely block waiting for the
firmware download.
After the firmware download has completed, the state of the
mux registers is known so a work job is queued to call
snd_soc_dapm_mux_update_power() on each of the mux widgets.
Backport note:
This won't apply cleanly to kernels older than v6.6.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: e49611252900 ("ASoC: cs35l56: Add driver for Cirrus Logic CS35L56")
Link: https://msgid.link/r/20240129162737.497-11-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 07687cd0539f8185b6ba0c0afba8473517116d6a ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ae861c466ee57e15a29d97629e1c564e3f714a4f ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1382d8b55129875b2e07c4d2a7ebc790183769ee ]
In the case where __lpass_get_dmactl_handle is called and the driver
id dai_id is invalid the pointer dmactl is not being assigned a value,
and dmactl contains a garbage value since it has not been initialized
and so the null check may not work. Fix this to initialize dmactl to
NULL. One could argue that modern compilers will set this to zero, but
it is useful to keep this initialized as per the same way in functions
__lpass_platform_codec_intf_init and lpass_cdc_dma_daiops_hw_params.
Cleans up clang scan build warning:
sound/soc/qcom/lpass-cdc-dma.c:275:7: warning: Branch condition
evaluates to a garbage value [core.uninitialized.Branch]
Fixes: b81af585ea54 ("ASoC: qcom: Add lpass CPU driver for codec dma control")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Link: https://msgid.link/r/20240221134804.3475989-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1d5a2b5dd0a8d2b2b535b5266699429dbd48e62f ]
ASoC is using 2 type of prefix (asoc_xxx() vs snd_soc_xxx()), but there
is no particular reason about that [1].
To reduce confusing, standarding these to snd_soc_xxx() is sensible.
This patch adds asoc_xxx() macro to keep compatible for a while.
It will be removed if all drivers were switched to new style.
Link: https://lore.kernel.org/r/87h6td3hus.wl-kuninori.morimoto.gx@renesas.com [1]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fs3ks26i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 1382d8b55129 ("ASoC: qcom: Fix uninitialized pointer dmactl")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4df49712eb54141be00a9312547436d55677f092 ]
We forgot to remove the line for snd-rtctimer from Makefile while
dropping the functionality. Get rid of the stale line.
Fixes: 34ce71a96dcb ("ALSA: timer: remove legacy rtctimer")
Link: https://lore.kernel.org/r/20240221092156.28695-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e33625c84b75e4f078d7f9bf58f01fe71ab99642 ]
The driver must write 0 to HALO_STATE before sending the SYSTEM_RESET
command to the firmware.
HALO_STATE is in DSP memory, which is preserved across a soft reset.
The SYSTEM_RESET command does not change the value of HALO_STATE.
There is period of time while the CS35L56 is resetting, before the
firmware has started to boot, where a read of HALO_STATE will return
the value it had before the SYSTEM_RESET. If the driver does not
clear HALO_STATE, this would return BOOT_DONE status even though the
firmware has not booted.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240216140535.1434933-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a6122b0b4211d132934ef99e7b737910e6d54d2f ]
This driver includes the legacy GPIO APIs <linux/gpio.h> and
<linux/of_gpio.h> but does not use any symbols from any of
them.
Drop the includes.
Further the driver is requesting "reset-gpios" rather than
just "reset" from the GPIO framework. This is wrong because
the gpiolib core will add "-gpios" before processing the
request from e.g. device tree. Drop the suffix.
The last problem means that the optional RESET GPIO has
never been properly retrieved and used even if it existed,
but nobody noticed.
Fixes: c1124c09e103 ("ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver.")
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-3-ee9f9d4655eb@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit eaa1b01fe709d6a236a9cec74813e0400601fd23 ]
For devices with multiple clock sources connected to a selector, we need
to check what a clock selector control request has returned. This is
needed to ensure that a requested clock source is indeed selected and for
autoclock feature to work.
For devices with single clock source connected, if we get an error there
is nothing else we can do about it. We can't skip clock selector setup as
it is required by some devices. So lets just ignore error in this case.
This should fix various buggy Mackie devices:
[ 649.109785] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.111946] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.113822] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
There is also interesting info from the Windows documentation [1] (this
is probably why manufacturers dont't even test this feature):
"The USB Audio 2.0 driver doesn't support clock selection. The driver
uses the Clock Source Entity, which is selected by default and never
issues a Clock Selector Control SET CUR request."
Link: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/usb-2-0-audio-drivers [1]
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217314
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218175
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218342
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240201115308.17838-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit daf3f0f99cde93a066240462b7a87cdfeedc04c0 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0adf963b8463faa44653e22e56ce55f747e68868 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 346f59d1e8ed0eed41c80e1acb657e484c308e6a ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 56beedc88405fd8022edfd1c2e63d1bc6c95efcb ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6cc2aa9a75f2397d42b78d4c159bc06722183c78 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 34a1066981a967eab619938e7b35a9be6b4c34e1 upstream.
The tascodec_init() of the snd-soc-tas2781-comlib module is called from
snd-soc-tas2781-i2c and snd-hda-scodec-tas2781-i2c modules. It calls
request_firmware_nowait() with parameter THIS_MODULE and a cont/callback
from the latter modules.
The latter modules can be removed while their callbacks are running,
resulting in a general protection failure.
Add module parameter to tascodec_init() so request_firmware_nowait() can
be called with the module of the callback.
Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Link: https://lore.kernel.org/r/118dad922cef50525e5aab09badef2fa0eb796e5.1707076603.git.soyer@irl.hu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fcbe4873089c84da641df75cda9cac2e9addbb4b upstream.
commit 74ad8ed65121 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops")
introduced a new allocation before the upper bounds check in
do_rx_work. As a result A DSP can cause bad allocations if spewing
garbage.
Fixes: 74ad8ed65121 ("ASoC: SOF: ipc3: Implement rx_msg IPC ops")
Reported-by: Tim Van Patten <timvp@google.com>
Cc: stable@vger.kernel.org
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240213123834.4827-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4639c5021029d49fd2f97fa8d74731f167f98919 upstream.
The SWS JS201D need a different pinconfig from windows driver.
Add a quirk to use a specific pinconfig to SWS JS201D.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240205013802.51907-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 32f03f4002c5df837fb920eb23fcd2f4af9b0b23 upstream.
The HP mt645 G7 Thin Client uses an ALC236 codec and needs the
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make the mute and
micmute LEDs work.
There are two variants of the USB-C PD chip on this device. Each uses
a different BIOS and board ID, hence the two entries.
Signed-off-by: Eniac Zhang <eniac-xw.zhang@hp.com>
Signed-off-by: Alexandru Gagniuc <alexandru.gagniuc@hp.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240215154922.778394-1-alexandru.gagniuc@hp.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1513664f340289cf10402753110f3cff12a738aa upstream.
The HP ZBook Power using ALC236 codec which using 0x02 to
control mute LED and 0x01 to control micmute LED.
Therefore, add a quirk to make it works.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240122074826.1020964-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a2ed0a44d637ef9deca595054c206da7d6cbdcbc upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 086df711d9b886194481b4fbe525eb43e9ae7403 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c7de2d9bb68a5fc71c25ff96705a80a76c8436eb upstream.
Vaio VJFE-ADL is equipped with ALC269VC, and it needs
ALC298_FIXUP_SPK_VOLUME quirk to make its headset mic work.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240201122114.30080-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2468e8922d2f6da81a6192b73023eff67e3fefdd upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit efb56d84dd9c3de3c99fc396abb57c6d330038b5 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c6dce23ec993f7da7790a9eadb36864ceb60e942 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 6ef5d5b92f7117b324efaac72b3db27ae8bb3082 ]
There is a path in rt5645_jack_detect_work(), where rt5645->jd_mutex
is left locked forever. That may lead to deadlock
when rt5645_jack_detect_work() is called for the second time.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: cdba4301adda ("ASoC: rt5650: add mutex to avoid the jack detection failure")
Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Link: https://lore.kernel.org/r/1707645514-21196-1-git-send-email-khoroshilov@ispras.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e5aa6d51a2ef8c7ef7e3fe76bebe530fb68e7f08 ]
Commit 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic
CS35L56 amplifier") adds configs SND_HDA_SCODEC_CS35L56_{I2C,SPI},
which selects the non-existing config CS_DSP. Note the renaming in
commit d7cfdf17cb9d ("firmware: cs_dsp: Rename KConfig symbol CS_DSP ->
FW_CS_DSP"), though.
Select the intended config FW_CS_DSP.
This broken select command probably was not noticed as the configs also
select SND_HDA_CS_DSP_CONTROLS and this then selects FW_CS_DSP. So, the
select FW_CS_DSP could actually be dropped, but we will keep this
redundancy in place as the author originally also intended to have this
redundancy of selects in place.
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Signed-off-by: Lukas Bulwahn <lukas.bulwahn@gmail.com>
Reviewed-by: Simon Trimmer <simont@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240209082044.3981-1-lukas.bulwahn@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d7332c4a4f1a7d16f054c6357fb65c597b6a86a7 ]
With the change in the widget free logic to power down the cores only
when the scheduler widgets are freed, we need to ensure that the
scheduler widget is freed only after all the widgets associated with the
scheduler are freed. This is to ensure that the secondary core that the
scheduler is scheduled to run on is kept powered on until all widgets
that need them are in use. While this works well for dynamic pipelines,
in the case of static pipelines the current logic does not take this into
account and frees all widgets in the order they occur in the
widget_list. So, modify this to ensure that the scheduler widgets are freed
only after all other types of widgets in the widget_list are freed.
Link: https://github.com/thesofproject/linux/issues/4807
Fixes: 31ed8da1c8e5 ("ASoC: SOF: sof-audio: Modify logic for enabling/disabling topology cores")
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20240208133432.1688-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 668abe6dc7b61941fa5c724c06797efb0b87f070 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7822baa844a87cbb93308c1032c3d47d4079bb8a upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a969210066054ea109d8b7aff29a9b1c98776841 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d915a6850e27efb383cd4400caadfe47792623df upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b53cc6144a3f6c8b56afcdec89d81195c9b0dc69 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 46188db080bd1df7d2d28031b89e56f2fdbabd67 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4d0e8bdfa4a57099dc7230952a460903f2e2f8de upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c481016bb4f8a9c059c39ac06e7b65e233a61f6a upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 7aeb259086487417f0fecf66e325bee133e8813a ]
When OMTP headset plugin the headset jack of CX8070 and SN6160 sound cards,
the headset type detection circuit will recognize the headset type as CTIA.
At this point, plugout and plugin the headset will get the correct headset
type as OMTP.
The reason for the failure of headset type recognition is that the sound
card creation will enable the VREF voltage of the headset mic, which
interferes with the headset type automatic detection circuit. Plugout and
plugin the headset will restart the headset detection and get the correct
headset type.
The patch is disable the VREF voltage when the headset is not present, and
will enable the VREF voltage when the headset is present.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Link: https://lore.kernel.org/r/20240108110235.3867-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
