ASoC: Fixes for v6.15
A moderately large batch of fixes for v6.15, many driver specific
including cleanups for the enabling of the Cirrus KUnit tests and a fix
for a nasty crash on resume on AMD systems. We also have one core fix,
for an ordering issue between DAPM and DPCM which could leave things
incorrectly unpowered.
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
Both CS35L56 and CS42L43 have maximum volumes above 0dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.
In the cases where systems use the soc_sdw_* drivers, audio above the
0dB volume will likely always be distorted, therefore apply a 0dB
limit to those devices.
Stefan Binding (2):
ASoC: intel/sdw_utils: Add volume limit to cs42l43 speakers
ASoC: intel/sdw_utils: Add volume limit to cs35l56 speakers
include/sound/soc_sdw_utils.h | 1 +
sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 4 ++++
sound/soc/sdw_utils/soc_sdw_cs42l43.c | 10 ++++++++
sound/soc/sdw_utils/soc_sdw_cs_amp.c | 24 ++++++++++++++++++++
4 files changed, 39 insertions(+)
--
2.43.0
Merge series from Olivier Moysan <olivier.moysan@foss.st.com>:
This patchset adds some checks on kernel minimum rate requirements.
This avoids potential clock rate misconfiguration, when setting the
kernel frequency on STM32MP2 SoCs.
If any Soundwire manager interrupt is reported, and wake interrupt
is not reported, in this scenario irq_flag will be set to zero,
which results in interrupt handler return status as IRQ_NONE.
Add new irq flag 'wake_irq_flag' check for SoundWire wake interrupt
handling to fix incorrect irq handling return status.
Fixes: 3898b18907 ("ASoC: amd: ps: add soundwire wake interrupt handling")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://patch.msgid.link/20250430195517.3065308-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Actually check if the passed pointers are valid, before writing to them.
This also fixes a USBAN warning:
UBSAN: invalid-load in ../sound/soc/fsl/imx-card.c:687:25
load of value 255 is not a valid value for type '_Bool'
This is because playback_only is uninitialized and is not written to, as
the playback-only property is absent.
Fixes: 844de7eebe ("ASoC: audio-graph-card2: expand dai_link property part")
Signed-off-by: Alexander Stein <alexander.stein@ew.tq-group.com>
Link: https://patch.msgid.link/20250429094910.1150970-1-alexander.stein@ew.tq-group.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The volume control for cs35l56 speakers has a maximum gain of +12 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the amp or the signal path has been tuned for a particular maximum
gain in the amp.
In the case of systems which use the soc_sdw_* driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 400, which is 0 dB maximum volume inside this driver.
The volume limit should be applied to both soundwire and soundwire
bridge configurations.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The volume control for cs42l43 speakers has a maximum gain of +31.5 dB.
However, for many use cases, this can cause distorted audio, depending
various factors, such as other signal-processing elements in the chain,
for example if the audio passes through a gain control before reaching
the codec or the signal path has been tuned for a particular maximum
gain in the codec.
In the case of systems which use the soc_sdw_cs42l43 driver, audio will
likely be distorted in all cases above 0 dB, therefore add a volume
limit of 128, which is 0 dB maximum volume inside this driver.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250430103134.24579-2-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On MP2 SoCs SAI kernel clock rate is managed through
stm32_sai_set_parent_rate() function.
If the kernel clock rate was set previously to a low frequency, this
frequency may be too low to support the newly requested audio stream rate.
However the stm32_sai_rate_accurate() will only check accuracy against
the maximum kernel clock rate. The function will return leaving the kernel
clock rate unchanged.
Add a check on minimal frequency requirement, to avoid this.
Fixes: 2cfe1ff225 ("ASoC: stm32: sai: add stm32mp25 support")
Signed-off-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20250430165210.321273-3-olivier.moysan@foss.st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
the frequency of the kernel clock must be greater than or equal to the
bitclock rate. When searching for a convenient kernel clock rate in
stm32_sai_set_parent_rate() function, it is useless to continue the loop
below bitclock rate, as it will result in a invalid kernel clock rate.
Change the loop output condition.
Fixes: 2cfe1ff225 ("ASoC: stm32: sai: add stm32mp25 support")
Signed-off-by: Olivier Moysan <olivier.moysan@foss.st.com>
Link: https://patch.msgid.link/20250430165210.321273-2-olivier.moysan@foss.st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A few ASUS models use the ALC256_FIXUP_ASUS_HEADSET_MODE although they
have no built-in mic pin on NID 0x13, as found in the commit
c1732ede5e ("ALSA: hda/realtek - Fix headset and mic on several Asus
laptops with ALC256"). This was relatively harmless in the past as
NID 0x13 was assigned as the secondary mic. But since the fix for the
pin sort order, this pin became the primary one, hence user started
noticing the broken input, and we've fixed already for a few ASUS
models to switch to ALC256_FIXUP_ASUS_MIC_NO_PRESENCE.
This patch corrects the other ASUS models to use the right quirk entry
for fixing the built-in mic regression. Here we cover X541SA
(1043:12e0), X541UV (1043:12f0), Z550SA (1043:13bf0) and X555UB
(1043:1ccd).
Fixes: 3b4309546b ("ALSA: hda: Fix headset detection failure due to unstable sort")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=220058
Link: https://patch.msgid.link/20250430053210.31776-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Depending on the architecture __ffs() returns either an 'unsigned long'
or 'unsigned int' result. Compile-testing this driver on targets that
use the latter produces a warning:
sound/soc/intel/catpt/dsp.c: In function 'catpt_dsp_set_srampge':
sound/soc/intel/catpt/dsp.c:181:44: error: format '%ld' expects argument of type 'long int', but argument 4 has type 'u32' {aka 'unsigned int'} [-Werror=format=]
181 | dev_dbg(cdev->dev, "sanitize block %ld: off 0x%08x\n",
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Change the type of the local variable to match the format string and
avoid the warning on any architecture.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250429073545.3558494-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The MIDI substream name string is constructed from the combination of
the card shortname (which is taken from USB iProduct) and the USB
iJack. The problem is that some devices put the product name to the
iJack field, too. For example, aplaymidi -l output on the Lanchkey MK
49 are like:
% aplaymidi -l
Port Client name Port name
44:0 Launchkey MK4 49 Launchkey MK4 49 Launchkey MK4
44:1 Launchkey MK4 49 Launchkey MK4 49 Launchkey MK4
where the actual iJack name can't be seen because it's truncated due
to the doubly words.
For resolving those situations, this patch compares the iJack string
with the card shortname, and drops if both start with the same words.
Then the result becomes like:
% aplaymidi -l
Port Client name Port name
40:0 Launchkey MK4 49 Launchkey MK4 49 MIDI In
40:1 Launchkey MK4 49 Launchkey MK4 49 DAW In
A caveat is that there are some pre-defined names for certain
devices in the driver code, and this workaround shouldn't be applied
to them. Similarly, when the iJack isn't specified, we should skip
this check, too. The patch added those checks in addition to the
string comparison.
Suggested-by: Paul Davis <paul@linuxaudiosystems.com>
Tested-by: Paul Davis <paul@linuxaudiosystems.com>
Link: https://lore.kernel.org/CAFa_cKmEDQWcJatbYWi6A58Zg4Ma9_6Nr3k5LhqwyxC-P_kXtw@mail.gmail.com
Link: https://patch.msgid.link/20250429183626.20773-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This hardware has ALC274 codec with speaker NID 0x17 and line out
NID 0x16 for audio output. The line out is routed correctly but
the speaker is not. Thus the volume can't be controlled.
This patch removes DAC NID 0x06 (without volume control) from the
connection list for speaker NID 0x17. Routing both speaker and line
out pins to DAC NID 0x02 which controls the output volume.
Signed-off-by: Chris Chiu <chris.chiu@canonical.com>
Link: https://patch.msgid.link/20250425103618.534951-1-chris.chiu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the latest kernel versions system crashes were noticed occasionally
during suspend/resume. This occurs because the RZ SSI suspend trigger
(called from snd_soc_suspend()) is executed after rz_ssi_pm_ops->suspend()
and it accesses IP registers. After the rz_ssi_pm_ops->suspend() is
executed the IP clocks are disabled and its reset line is asserted.
Since snd_soc_suspend() is invoked through snd_soc_pm_ops->suspend(),
snd_soc_pm_ops is associated with soc_driver (defined in
sound/soc/soc-core.c), and there is no parent-child relationship between
soc_driver and rz_ssi_driver the power management subsystem does not
enforce a specific suspend/resume order between the RZ SSI platform driver
and soc_driver.
To ensure that the suspend/resume function of rz-ssi is executed after
snd_soc_suspend(), use NOIRQ_SYSTEM_SLEEP_PM_OPS().
Fixes: 1fc778f7c8 ("ASoC: renesas: rz-ssi: Add suspend to RAM support")
Cc: stable@vger.kernel.org
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Link: https://patch.msgid.link/20250410141525.4126502-1-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During initialisation of Focusrite USB audio interfaces, -EPROTO is
sometimes returned from usb_set_interface(), which sometimes prevents
the device from working: subsequent usb_set_interface() and
uac_clock_source_is_valid() calls fail.
This patch adds up to 5 retries in endpoint_set_interface(), with a
delay starting at 5ms and doubling each time. 5 retries was chosen to
allow for longer than expected waits for the interface to start
responding correctly; in testing, a single 5ms delay was sufficient to
fix the issue.
Closes: https://github.com/geoffreybennett/fcp-support/issues/2
Cc: stable@vger.kernel.org
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/Z//7s9dKsmVxHzY2@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The speaker doesn't mute when plugged headphone.
This platform support 4ch speakers.
The speaker pin 0x14 wasn't fill verb table.
After assigned model ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX.
The speaker can mute when headphone was plugged.
Fixes: aa8e3ef4fe ("ALSA: hda/realtek: Add quirks for various HP ENVY models")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/eb4c14a4d85740069c909e756bbacb0e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
This series fixes the KConfig for cs_dsp and cs-amp-lib tests so that
CONFIG_KUNIT_ALL_TESTS doesn't cause them to add modules to the build.
Issue:
When multiple audio streams share a common BE DAI, the BE DAI
widget can be powered up before its hardware parameters are configured.
This incorrect sequence leads to intermittent pcm_write errors.
For example, the below Tegra use-case throws an error:
aplay(2 streams) -> AMX(mux) -> ADX(demux) -> arecord(2 streams),
here, 'AMX TX' and 'ADX RX' are common BE DAIs.
For above usecase when failure happens below sequence is observed:
aplay(1) FE open()
- BE DAI callbacks added to the list
- BE DAI state = SND_SOC_DPCM_STATE_OPEN
aplay(2) FE open()
- BE DAI callbacks are not added to the list as the state is
already SND_SOC_DPCM_STATE_OPEN during aplay(1) FE open().
aplay(2) FE hw_params()
- BE DAI hw_params() callback ignored
aplay(2) FE prepare()
- Widget is powered ON without BE DAI hw_params() call
aplay(1) FE hw_params()
- BE DAI hw_params() is now called
Fix:
Add BE DAIs in the list if its state is either SND_SOC_DPCM_STATE_OPEN
or SND_SOC_DPCM_STATE_HW_PARAMS as well.
It ensures the widget is powered ON after BE DAI hw_params() callback.
Fixes: 0c25db3f76 ("ASoC: soc-pcm: Don't reconnect an already active BE")
Signed-off-by: Sheetal <sheetal@nvidia.com>
Link: https://patch.msgid.link/20250404105953.2784819-1-sheetal@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Fixes for v6.15
A set of small fixes, quirks and device ID additions that came in since
-rc1, none of them super stand out. There's also a change to Srini's
email address in MAINTAINERS.
On SNDRV_PCM_TRIGGER_START event, audio data pointers are not reset.
This leads to wrong data buffer usage when multiple TRIGGER_START are
received and ends to incorrect buffer usage between the user-space and
the driver. Indeed, the driver can read data that are not already set by
the user-space or the user-space and the driver are writing and reading
the same area.
Fix that resetting data pointers on each SNDRV_PCM_TRIGGER_START events.
Fixes: 075c7125b1 ("ASoC: fsl: Add support for QMC audio")
Cc: stable@vger.kernel.org
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Link: https://patch.msgid.link/20250410091643.535627-1-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Depend on SND_HDA_CIRRUS_SCODEC and GPIOLIB instead of selecting them.
KUNIT_ALL_TESTS should only build tests that have satisfied dependencies
and test components that are already being built. It must not cause
other stuff to be added to the build.
Fixes: 2144833e7b ("ALSA: hda: cirrus_scodec: Add KUnit test")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250409114520.914079-1-rf@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from srinivas.kandagatla@linaro.org:
This two patches fixes below two issues with the VI setup.
1. Only one channel gets enabled on VI feedback patch instead of two
channels
2. recording rate is hardcoded to 8K instead dyamically setting it up.
Both of these issues are fixed in these patches.
Case values introduced in commit
5f78e1fb7a ("ASoC: qcom: Add driver support for audioreach solution")
cause out of bounds access in arrays of sc7280 driver data (e.g. in case
of RX_CODEC_DMA_RX_0 in sc7280_snd_hw_params()).
Redefine LPASS_MAX_PORTS to consider the maximum possible port id for
q6dsp as sc7280 driver utilizes some of those values.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: 77d0ffef79 ("ASoC: qcom: Add macro for lpass DAI id's max limit")
Cc: stable@vger.kernel.org # v6.0+
Suggested-by: Mikhail Kobuk <m.kobuk@ispras.ru>
Suggested-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Signed-off-by: Evgeny Pimenov <pimenoveu12@gmail.com>
Link: https://patch.msgid.link/20250401204058.32261-1-pimenoveu12@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a42e988 ("ASoC: dwc: add DMA handshake control") changed the
behavior of the driver to not enable or disable i2s irqs if using DMA. This
breaks platforms such as AMD ACP. Audio playback appears to work but no
audio can be heard. Revert to the old behavior by always enabling and
disabling i2s irqs while keeping DMA handshake control.
Fixes: a42e988b62 ("ASoC: dwc: add DMA handshake control")
Signed-off-by: Brady Norander <bradynorander@gmail.com>
Link: https://patch.msgid.link/20250330130852.37881-3-bradynorander@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull soundwire fix from Vinod Koul:
- add missing config symbol CONFIG_SND_HDA_EXT_CORE required for asoc
driver CONFIG_SND_SOF_SOF_HDA_SDW_BPT
* tag 'soundwire-6.15-rc1-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire:
ASoC: SOF: Intel: Let SND_SOF_SOF_HDA_SDW_BPT select SND_HDA_EXT_CORE
timer_delete[_sync]() replaces del_timer[_sync](). Convert the whole tree
over and remove the historical wrapper inlines.
Conversion was done with coccinelle plus manual fixups where necessary.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Ingo Molnar <mingo@kernel.org>
