commit c6557ccf80 upstream.
This fixes a regression introduced a few weeks ago in stable kernels
6.12.14 and 6.13.3. The internal microphone on ASUS Vivobook N705UD /
X705UD laptops is broken: the microphone appears in userspace (e.g.
Gnome settings) but no sound is detected.
I bisected it to commit 3b4309546b ("ALSA: hda: Fix headset detection
failure due to unstable sort").
I figured out the cause:
1. The initial pins enabled for the ALC256 driver are:
cfg->inputs == {
{ pin=0x19, type=AUTO_PIN_MIC,
is_headset_mic=1, is_headphone_mic=0, has_boost_on_pin=1 },
{ pin=0x1a, type=AUTO_PIN_MIC,
is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 } }
2. Since 2017 and commits c1732ede5e ("ALSA: hda/realtek - Fix headset
and mic on several ASUS laptops with ALC256") and 28e8af8a16 ("ALSA:
hda/realtek: Fix mic and headset jack sense on ASUS X705UD"), the
quirk ALC256_FIXUP_ASUS_MIC is also applied to ASUS X705UD / N705UD
laptops.
This added another internal microphone on pin 0x13:
cfg->inputs == {
{ pin=0x13, type=AUTO_PIN_MIC,
is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 },
{ pin=0x19, type=AUTO_PIN_MIC,
is_headset_mic=1, is_headphone_mic=0, has_boost_on_pin=1 },
{ pin=0x1a, type=AUTO_PIN_MIC,
is_headset_mic=0, is_headphone_mic=0, has_boost_on_pin=1 } }
I don't know what this pin 0x13 corresponds to. To the best of my
knowledge, these laptops have only one internal microphone.
3. Before 2025 and commit 3b4309546b ("ALSA: hda: Fix headset
detection failure due to unstable sort"), the sort function would let
the microphone of pin 0x1a (the working one) *before* the microphone
of pin 0x13 (the phantom one).
4. After this commit 3b4309546b, the fixed sort function puts the
working microphone (pin 0x1a) *after* the phantom one (pin 0x13). As
a result, no sound is detected anymore.
It looks like the quirk ALC256_FIXUP_ASUS_MIC is not needed anymore for
ASUS Vivobook X705UD / N705UD laptops. Without it, everything works
fine:
- the internal microphone is detected and records actual sound,
- plugging in a jack headset is detected and can record actual sound
with it,
- unplugging the jack headset makes the system go back to internal
microphone and can record actual sound.
Cc: stable@vger.kernel.org
Cc: Kuan-Wei Chiu <visitorckw@gmail.com>
Cc: Chris Chiu <chris.chiu@canonical.com>
Fixes: 3b4309546b ("ALSA: hda: Fix headset detection failure due to unstable sort")
Tested-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Adrien Vergé <adrienverge@gmail.com>
Link: https://patch.msgid.link/20250226135515.24219-1-adrienverge@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9af3b4f2d8 upstream.
Re-add the sample-rate quirk for the Pioneer DJM-900NXS2. This
device does not work without setting sample-rate.
Signed-off-by: Dmitry Panchenko <dmitry@d-systems.ee>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250220161540.3624660-1-dmitry@d-systems.ee
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9e7c6779e3 ]
ASUS VivoBook 15 with SSID 1043:1460 took an incorrect quirk via the
pin pattern matching for ASUS (ALC256_FIXUP_ASUS_MIC), resulting in
the two built-in mic pins (0x13 and 0x1b). This had worked without
problems casually in the past because the right pin (0x1b) was picked
up as the primary device. But since we fixed the pin enumeration for
other bugs, the bogus one (0x13) is picked up as the primary device,
hence the bug surfaced now.
For addressing the regression, this patch explicitly specifies the
quirk entry with ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, which sets up only
the headset mic pin.
Fixes: 3b4309546b ("ALSA: hda: Fix headset detection failure due to unstable sort")
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219807
Link: https://patch.msgid.link/20250225154540.13543-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 769c1b7929 ]
When SPI is used for control, the driver must hold the SPI bus lock
while issuing the sequence of writes to perform a soft reset.
>From the time the driver writes the SYSTEM_RESET command until the
driver does a write to terminate the reset, there must not be any
activity on the SPI bus lines. If there is any SPI activity during the
soft-reset, another soft-reset will be triggered. The state of the SPI
chip select is irrelevant.
A repeated soft-reset does not in itself cause any problems, and it is
not an infinite loop. The problem is a race between these resets and
the driver polling for boot completion. There is a time window between
soft resets where the driver could read HALO_STATE as 2 (fully booted)
while the chip is actually soft-resetting. Although this window is
small, it is long enough that it is possible to hit it in normal
operation.
To prevent this race and ensure the chip really is fully booted, the
driver calls spi_bus_lock() to prevent other activity while resetting.
It then issues the SYSTEM_RESET mailbox command. After allowing
sufficient time for reset to take effect, the driver issues a PING
mailbox command, which will force completion of the full soft-reset
sequence. The SPI bus lock can then be released. The mailbox is
checked for any boot or wakeup response from the firmware, before the
value in HALO_STATE will be trusted.
This does not affect SoundWire or I2C control.
Fixes: 8a731fd37f ("ASoC: cs35l56: Move utility functions to shared file")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://patch.msgid.link/20250225131843.113752-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0da83ab025 ]
If stream names of DAI driver are duplicated there'll be warnings when
machine driver tries to add widgets on a route:
[ 8.831335] fsl-asoc-card sound-wm8960: ASoC: sink widget CPU-Playback overwritten
[ 8.839917] fsl-asoc-card sound-wm8960: ASoC: source widget CPU-Capture overwritten
Use different stream names to avoid such warnings.
DAI names in AUDMIX are also updated accordingly.
Fixes: 15c9583904 ("ASoC: fsl_sai: Add separate DAI for transmitter and receiver")
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Acked-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Link: https://patch.msgid.link/20250217010437.258621-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5b0c02f9b8 ]
The ES8328 codec driver, which is also used for the ES8388 chip that
appears to have an identical register map, claims that the output can
either take the route from DAC->Mixer->Output or through DAC->Output
directly. To the best of what I could find, this is not true, and
creates problems.
Without DACCONTROL17 bit index 7 set for the left channel, as well as
DACCONTROL20 bit index 7 set for the right channel, I cannot get any
analog audio out on Left Out 2 and Right Out 2 respectively, despite the
DAPM routes claiming that this should be possible. Furthermore, the same
is the case for Left Out 1 and Right Out 1, showing that those two don't
have a direct route from DAC to output bypassing the mixer either.
Those control bits toggle whether the DACs are fed (stale bread?) into
their respective mixers. If one "unmutes" the mixer controls in
alsamixer, then sure, the audio output works, but if it doesn't work
without the mixer being fed the DAC input then evidently it's not a
direct output from the DAC.
ES8328/ES8388 are seemingly not alone in this. ES8323, which uses a
separate driver for what appears to be a very similar register map,
simply flips those two bits on in its probe function, and then pretends
there is no power management whatsoever for the individual controls.
Fair enough.
My theory as to why nobody has noticed this up to this point is that
everyone just assumes it's their fault when they had to unmute an
additional control in ALSA.
Fix this in the es8328 driver by removing the erroneous direct route,
then get rid of the playback switch controls and have those bits tied to
the mixer's widget instead, which until now had no register to play
with.
Fixes: 567e4f9892 ("ASoC: add es8328 codec driver")
Signed-off-by: Nicolas Frattaroli <nicolas.frattaroli@collabora.com>
Link: https://patch.msgid.link/20250222-es8328-route-bludgeoning-v1-1-99bfb7fb22d9@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a3bdd8f5c2 ]
We fixed the UAF issue in USB MIDI code by canceling the pending work
at closing each MIDI output device in the commit below. However, this
assumed that it's the only one that is tied with the endpoint, and it
resulted in unexpected data truncations when multiple devices are
assigned to a single endpoint and opened simultaneously.
For addressing the unexpected MIDI message drops, simply replace
cancel_work_sync() with flush_work(). The drain callback should have
been already invoked before the close callback, hence the port->active
flag must be already cleared. So this just assures that the pending
work is finished before freeing the resources.
Fixes: 0125de3812 ("ALSA: usb-audio: Cancel pending work at closing a MIDI substream")
Reported-and-tested-by: John Keeping <jkeeping@inmusicbrands.com>
Closes: https://lore.kernel.org/20250217111647.3368132-1-jkeeping@inmusicbrands.com
Link: https://patch.msgid.link/20250218114024.23125-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 46c7b901e2 upstream.
The spcm->stream[substream->stream].substream is set during open and was
left untouched. After the first PCM stream it will never be NULL and we
have code which checks for substream NULLity as indication if the stream is
active or not.
For the compressed cstream pointer the same has been done, this change will
correct the handling of PCM streams.
Fixes: 090349a9fe ("ASoC: SOF: Add support for compress API for stream data/offset")
Cc: stable@vger.kernel.org
Reported-by: Curtis Malainey <cujomalainey@chromium.org>
Closes: https://github.com/thesofproject/linux/pull/5214
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://patch.msgid.link/20250205135232.19762-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6d1f86610f upstream.
Allows the LED on the dedicated mute button on the HP ProBook 450 G4
laptop to change colour correctly.
Signed-off-by: John Veness <john-linux@pelago.org.uk>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/2fb55d48-6991-4a42-b591-4c78f2fad8d7@pelago.org.uk
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 822b7ec657 upstream.
Check the return value of snd_ctl_rename_id() in
snd_hda_create_dig_out_ctls(). Ensure that failures
are properly handled.
[ Note: the error cannot happen practically because the only error
condition in snd_ctl_rename_id() is the missing ID, but this is a
rename, hence it must be present. But for the code consistency,
it's safer to have always the proper return check -- tiwai ]
Fixes: 5c219a3408 ("ALSA: hda: Fix kctl->id initialization")
Cc: stable@vger.kernel.org # 6.4+
Signed-off-by: Wentao Liang <vulab@iscas.ac.cn>
Link: https://patch.msgid.link/20250213074543.1620-1-vulab@iscas.ac.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a8c9a45338 upstream.
If 'micfil->quality' received from micfil_quality_set() somehow ends
up with an unpredictable value, switch() operator will fail to
initialize local variable qsel before regmap_update_bits() tries
to utilize it.
While it is unlikely, play it safe and enable a default case that
returns -EINVAL error.
Found by Linux Verification Center (linuxtesting.org) with static
analysis tool SVACE.
Fixes: bea1d61d58 ("ASoC: fsl_micfil: rework quality setting")
Cc: stable@vger.kernel.org
Signed-off-by: Nikita Zhandarovich <n.zhandarovich@fintech.ru>
Link: https://patch.msgid.link/20250116142436.22389-1-n.zhandarovich@fintech.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d8d99c3b5c upstream.
The nullity of sps->cstream should be checked similarly as it is done in
sof_set_stream_data_offset() function.
Assuming that it is not NULL if sps->stream is NULL is incorrect and can
lead to NULL pointer dereference.
Fixes: 090349a9fe ("ASoC: SOF: Add support for compress API for stream data/offset")
Cc: stable@vger.kernel.org
Reported-by: Curtis Malainey <cujomalainey@chromium.org>
Closes: https://github.com/thesofproject/linux/pull/5214
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://patch.msgid.link/20250205135232.19762-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit e77aa4b2ea ]
When a destination client is a user client in the legacy MIDI mode and
it sets the no-UMP-conversion flag, currently the all UMP events are
still passed as-is. But this may confuse the user-space, because the
event packet size is different from the legacy mode.
Since we cannot handle UMP events in user clients unless it's running
in the UMP client mode, we should filter out those events instead of
accepting blindly. This patch addresses it by slightly adjusting the
conditions for UMP event handling at the event delivery time.
Fixes: 329ffe11a0 ("ALSA: seq: Allow suppressing UMP conversions")
Link: https://lore.kernel.org/b77a2cd6-7b59-4eb0-a8db-22d507d3af5f@gmail.com
Link: https://patch.msgid.link/20250217170034.21930-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 08b613b9e2 ]
This patch corrects the full-scale volume setting logic. On certain
platforms, the full-scale volume bit is required. The current logic
mistakenly sets this bit and incorrectly clears reserved bit 0, causing
the headphone output to be muted.
Fixes: 342b6b610a ("ALSA: hda/cs8409: Fix Full Scale Volume setting for all variants")
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://patch.msgid.link/20250214210736.30814-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 571b69f2f9 ]
When defer probe happens, there may be below error:
platform 59820000.sai: Resources present before probing
The cpu_mclk clock is from the cpu dai device, if it is not released,
then the cpu dai device probe will fail for the second time.
The cpu_mclk is used to get rate for rate constraint, rate constraint
may be specific for each platform, which is not necessary for machine
driver, so remove it.
Fixes: b86ef53677 ("ASoC: fsl: Add Audio Mixer machine driver")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250213070518.547375-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6fd60136d2 ]
Other, non DAI copier widgets could have the same stream name (sname) as
the ALH copier and in that case the copier->data is NULL, no alh_data is
attached, which could lead to NULL pointer dereference.
We could check for this NULL pointer in sof_ipc4_prepare_copier_module()
and avoid the crash, but a similar loop in sof_ipc4_widget_setup_comp_dai()
will miscalculate the ALH device count, causing broken audio.
The correct fix is to harden the matching logic by making sure that the
1. widget is a DAI widget - so dai = w->private is valid
2. the dai (and thus the copier) is ALH copier
Fixes: a150345aa7 ("ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support")
Reported-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Link: https://github.com/thesofproject/sof/pull/9652
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250206084642.14988-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6b24e67b40 ]
Commit 2f45a4e289 ("ASoC: rockchip: i2s_tdm: Fixup config for
SND_SOC_DAIFMT_DSP_A/B") applied a partial change to fix the
configuration for DSP A and DSP B formats.
The shift control also needs updating to set the correct offset for
frame data compared to LRCK. Set the correct values.
Fixes: 081068fd64 ("ASoC: rockchip: add support for i2s-tdm controller")
Signed-off-by: John Keeping <jkeeping@inmusicbrands.com>
Link: https://patch.msgid.link/20250204161311.2117240-1-jkeeping@inmusicbrands.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 82a0a3e6f8 ]
My static checker rule complains about this code. The concern is that
if "sample_space" is negative then the "sample_space >= runtime->channels"
condition will not work as intended because it will be type promoted to a
high unsigned int value.
strm->fifo_sample_size is SSI_FIFO_DEPTH (32). The SSIFSR_TDC_MASK is
0x3f. Without any further context it does seem like a reasonable warning
and it can't hurt to add a check for negatives.
Cc: stable@vger.kernel.org
Fixes: 03e786bd43 ("ASoC: sh: Add RZ/G2L SSIF-2 driver")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://patch.msgid.link/e07c3dc5-d885-4b04-a742-71f42243f4fd@stanley.mountain
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 541011dc2d ]
The stop trigger invokes rz_ssi_stop() and rz_ssi_stream_quit().
- The purpose of rz_ssi_stop() is to disable TX/RX, terminate DMA
transactions, and set the controller to idle.
- The purpose of rz_ssi_stream_quit() is to reset the substream-specific
software data by setting strm->running and strm->substream appropriately.
The function rz_ssi_is_stream_running() checks if both strm->substream and
strm->running are valid and returns true if so. Its implementation is as
follows:
static inline bool rz_ssi_is_stream_running(struct rz_ssi_stream *strm)
{
return strm->substream && strm->running;
}
When the controller is configured in full-duplex mode (with both playback
and capture active), the rz_ssi_stop() function does not modify the
controller settings when called for the first substream in the full-duplex
setup. Instead, it simply sets strm->running = 0 and returns if the
companion substream is still running. The following code illustrates this:
static int rz_ssi_stop(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm)
{
strm->running = 0;
if (rz_ssi_is_stream_running(&ssi->playback) ||
rz_ssi_is_stream_running(&ssi->capture))
return 0;
// ...
}
The controller settings, along with the DMA termination (for the last
stopped substream), are only applied when the last substream in the
full-duplex setup is stopped.
While applying the controller settings only when the last substream stops
is not problematic, terminating the DMA operations for only one substream
causes failures when starting and stopping full-duplex operations multiple
times in a loop.
To address this issue, call dmaengine_terminate_async() for both substreams
involved in the full-duplex setup when the last substream in the setup is
stopped.
Fixes: 4f8cd05a43 ("ASoC: sh: rz-ssi: Add full duplex support")
Cc: stable@vger.kernel.org
Reviewed-by: Biju Das <biju.das.jz@bp.renesas.com>
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://patch.msgid.link/20241210170953.2936724-5-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6917192378 ]
The Vexia EDU ATLA 10 tablet comes in 2 different versions with
significantly different mainboards. The only outward difference is that
the charging barrel on one is marked 5V and the other is marked 9V.
The 5V version mostly works with the BYTCR defaults, except that it is
missing a CHAN package in its ACPI tables and the default of using
SSP0-AIF2 is wrong, instead SSP0-AIF1 must be used. That and its jack
detect signal is not inverted as it usually is.
Add a DMI quirk for the 5V version to fix sound not working.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://patch.msgid.link/20250123132507.18434-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 3b4309546b upstream.
The auto_parser assumed sort() was stable, but the kernel's sort() uses
heapsort, which has never been stable. After commit 0e02ca29a5
("lib/sort: optimize heapsort with double-pop variation"), the order of
equal elements changed, causing the headset to fail to work.
Fix the issue by recording the original order of elements before
sorting and using it as a tiebreaker for equal elements in the
comparison function.
Fixes: b9030a005d ("ALSA: hda - Use standard sort function in hda_auto_parser.c")
Reported-by: Austrum <austrum.lab@gmail.com>
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219158
Tested-by: Austrum <austrum.lab@gmail.com>
Cc: stable@vger.kernel.org
Signed-off-by: Kuan-Wei Chiu <visitorckw@gmail.com>
Link: https://patch.msgid.link/20250128165415.643223-1-visitorckw@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0f3a822ae2 upstream.
The recent cleanup of the quirk table entries with the codec ID
matching caused a regression on some Lenovo Legion 7 models with PCI
SSID 17aa:386f: it assumed wrongly as if the codec SSID on the machine
were also 17aa:386f, but in this case, it was 17aa:38a8. This made
the binding with a wrong sub-codec, instead of TAS2781, the Cirrus
codec was bound.
For addressing the regression, correct the quirk entry to the right
value 17aa:38a8.
Note that this makes the entry appearing in an unsorted position.
This exception is needed because the entry must match before the PCI
SSID 17aa:386f.
Also there is another entry for 17aa:38a8, but the latter is for PCI
SSID matching while the new entry is for the codec SSID matching.
Fixes: 504f052aa3 ("ALSA: hda/realtek: Use codec SSID matching for Lenovo devices")
Reported-and-tested-by: Samantha Glocker <iam@anislandsomewhere.com>
Closes: https://lore.kernel.org/CAGPQRHYd48U__UKYj2jJnT4+dnNNoWRBi+wj6zPRn=JpNMBUrg@mail.gmail.com
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250125120519.16420-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1aec3ed2e3 upstream.
Positivo C6400 is equipped with ALC269VB, and it needs
ALC269VB_FIXUP_ASUS_ZENBOOK quirk to make its headset mic work.
Also must to limits the microphone boost.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250114170619.11510-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 7e24ec93ae ]
As reported by the kernel test robot, the following error occurs:
sound/soc/amd/yc/acp6x-mach.c: In function 'acp6x_probe':
>> sound/soc/amd/yc/acp6x-mach.c:573:15: error: implicit declaration of function 'acpi_evaluate_integer'; did you mean 'acpi_evaluate_object'? [-Werror=implicit-function-declaration]
573 | ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status);
| ^~~~~~~~~~~~~~~~~~~~~
| acpi_evaluate_object
cc1: some warnings being treated as errors
The function 'acpi_evaluate_integer' and its prototype in 'acpi_bus.h'
are only available when 'CONFIG_ACPI' is enabled. Add a 'depends on ACPI'
directive in Kconfig to ensure proper compilation.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202501090345.pBIDRTym-lkp@intel.com/
Signed-off-by: Yu-Chun Lin <eleanor15x@gmail.com>
Link: https://patch.msgid.link/20250109171547.362412-1-eleanor15x@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 301c26a018 ]
commit 1f56643514 ("ASoC: lower "no backend DAIs enabled for ... Port"
log severity") ignores -EINVAL error message on common soc_pcm_ret().
It is used from many functions, ignoring -EINVAL is over-kill.
The reason why -EINVAL was ignored was it really should only be used
upon invalid parameters coming from userspace and in that case we don't
want to log an error since we do not want to give userspace a way to do
a denial-of-service attack on the syslog / diskspace.
So don't use soc_pcm_ret() on .prepare callback is better idea.
Link: https://lore.kernel.org/r/87v7vptzap.wl-kuninori.morimoto.gx@renesas.com
Cc: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87bjxg8jju.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7662f0e5d5 ]
In addition to changing the DMI match to examine the product name rather
than the SKU, this adds the quirk to inform the machine driver to not
bind in the cs42l43 microphone DAI link.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20241206075903.195730-5-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 569922b82c ]
Each cpu DAI should associate with a widget. However, the topology might
not create the right number of DAI widgets for aggregated amps. And it
will cause NULL pointer deference.
Check that the DAI widget associated with the CPU DAI is valid to prevent
NULL pointer deference due to missing DAI widgets in topologies with
aggregated amps.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Link: https://patch.msgid.link/20241203104853.56956-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit d85fc52cbb upstream.
Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
usb 1-1: New USB device found, idVendor=2fc6, idProduct=f0b7
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
usb 1-1: Product: iBasso DC07 Pro
usb 1-1: Manufacturer: iBasso
usb 1-1: SerialNumber: CTUA171130B
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/TYUPR06MB62174A48D04E09A37996DF84D2ED2@TYUPR06MB6217.apcprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 3ff53862c3 ]
On error path, function acp_i2s_set_tdm_slot returns without releasing
the lock and this could result in potential deadlocks in the future.
Error reported by sparse:
sound/soc/amd/acp/acp-i2s.c:95:12: error: context imbalance in
'acp_i2s_set_tdm_slot' - different lock contexts for basic block
Fixes: cd60dec899 ("ASoC: amd: acp: Refactor TDM slots selction based on acp revision id")
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.dev>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250127083422.20406-1-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5323186e2e ]
In commit
9e2ab4b18e ("ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates"),
the set_sysclk callback was removed as considered unused as the mclk rate
can be set in the hw_params callback.
The difference between hw_params and set_sysclk is that the former is
called with the audio sampling rate set in the params (e.g.: 48000 Hz)
while the latter is called with a clock rate already computed with
sampling_rate * mclk-fs (e.g.: 48000 * 256)
For HDMI audio using the Rockchip I2S TDM driver, the mclk-fs value must
be set to 128 instead of the default 256, and that value is set in the
device tree at the machine driver level (like a simple-audio-card
compatible node).
Therefore, the i2s_tdm driver has no idea that another mclk-fs value can
be configured and simply computes the mclk rate in the hw_params callback
with DEFAULT_MCLK_FS * params_rate(params), which is wrong for HDMI
audio.
Re-add the set_sysclk callback so that the mclk rate is computed by the
machine driver which has the correct mclk-fs value set in its device tree
node.
Fixes: 9e2ab4b18e ("ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates")
Signed-off-by: Detlev Casanova <detlev.casanova@collabora.com>
Link: https://patch.msgid.link/20250117163102.65807-1-detlev.casanova@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 17615e4216 ]
Update the DMI match for a Lenovo laptop to a new DMI identifier.
This laptop ships with a different DMI identifier to what was expected
and now has three match entries. It also has the DMICs connected to the
host rather than the cs42l43 codec.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 83c062ae81 ("ASoC: Intel: sof_sdw: Add quirks for some new Lenovo laptops")
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250102123335.256698-3-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c9e05763f3 ]
Update the DMI match for a Lenovo laptop to the new DMI identifier.
This laptop ships with a different DMI identifier to what was expected,
and also has the DMICs connected to the host rather than the cs42l43
codec.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 83c062ae81 ("ASoC: Intel: sof_sdw: Add quirks for some new Lenovo laptops")
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://patch.msgid.link/20250102123335.256698-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7579790915 ]
The snd_hdac_adsp_xxx() wrap snd_hdac_reg_xxx() helpers to simplify
register access for AudioDSP drivers e.g.: the avs-driver. Byte- and
word-variants of said helps do not expand to bare readx/writex()
operations but functions instead and, due to pointer type
incompatibility, cause compilation to fail.
As the macros are utilized by the avs-driver alone, relocate the code
introduced with commit c19bd02e90 ("ALSA: hda: Add helper macros for
DSP capable devices") into the avs/ directory and update it to operate
on 'adev' i.e.: the avs-driver-context directly to fix the issue.
Fixes: c19bd02e90 ("ALSA: hda: Add helper macros for DSP capable devices")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250110113326.3809897-2-cezary.rojewski@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cf4d74256f ]
While 'stack_dump_size' is a u32 bitfield of 16 bits, u32 has a bigger
upper bound than the type u16 of loop counter 'offset' what in theory
may lead to infinite loop condition.
Found out by Coverity static analyzer.
Fixes: c8c960c109 ("ASoC: Intel: avs: APL-based platforms support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250109122216.3667847-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit dbda5c35b8 ]
For few TGL-based platforms the minor version number for AudioDSP
firmware is incorrect forcing users to utilize ignore_fw_version module
parameter.
Fixes: 5acb19ecd1 ("ASoC: Intel: avs: TGL-based platforms support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250109122216.3667847-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9001d51544 ]
CONFIG_SND_SEQ_UMP_CLIENT is a Kconfig for a sequencer client
corresponding to the UMP rawmidi, while we have another major knob
CONFIG_SND_SEQ_UMP that specifies whether the sequencer core supports
UMP packets or not. Strictly speaking both of them are independent,
but practically seen, it makes no sense to enable
CONFIG_SND_SEQ_UMP_CLIENT without UMP support itself.
This patch makes such an implicit dependency clearer. Now
CONFIG_SND_SEQ_UMP_CLIENT depends on both CONFIG_SND_UMP and
CONFIG_SND_SEQ_UMP. Meanwhile, CONFIG_SND_SEQ_UMP is enabled as
default when CONFIG_SND_UMP is set.
Fixes: 81fd444aa3 ("ALSA: seq: Bind UMP device")
Link: https://patch.msgid.link/20250101125548.25961-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 55c209cd43 ]
There is no need to populate the ckdv[] with invalid dividers as that
part will not be indexed anyway. The ssi->audio_mck/bclk_rate should
always be >= 0. While at it, change the ckdv type as u8, as the divider
128 was previously using the s8 sign bit.
Signed-off-by: Claudiu Beznea <claudiu.beznea.uj@bp.renesas.com>
Fixes: 03e786bd43 ("ASoC: sh: Add RZ/G2L SSIF-2 driver")
Reviewed-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://patch.msgid.link/20241210170953.2936724-6-claudiu.beznea.uj@bp.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0a2319308d ]
There have been intermittent issues with the SPDIF output on H3
and H2+ devices which has been fixed by setting the s_clk to 4
times the audio pll.
Add a quirk for the clock multiplier as not every supported SoC
requires it. Without the multiplier, the audio at normal sampling
rates was distorted and did not play at higher sampling rates.
Fixes: 1bd92af877 ("ASoC: sun4i-spdif: Add support for the H3 SoC")
Signed-off-by: George Lander <lander@jagmn.com>
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Link: https://patch.msgid.link/20241111165600.57219-2-codekipper@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4ab80a2961 ]
mach_params->dmic_num will be used to set the cfg-mics value of
card->components string which should be the dmic channels. However
dmic_num is dmic link number and could be set due to the SOC_SDW_PCH_DMIC
quirk. Set mach_params->dmic_num to the default value if the dmic link
is created due to the SOC_SDW_PCH_DMIC quirk.
Fixes: 7db9f63611 ("ASoC: Intel: sof_sdw: overwrite mach_params->dmic_num")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://patch.msgid.link/20241206075903.195730-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 582057d223 ]
We should use *-y instead of *-objs in Makefile for the module
objects. *-objs is used rather for host programs.
Fixes: 313e978df7 ("ASoC: codecs: wcd937x: add audio routing and Kconfig")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241203141823.22393-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e9d2a2f492 ]
We should use *-y instead of *-objs in Makefile for the module
objects. *-objs is used rather for host programs.
Fixes: 5bbfdad8cf ("ASoC: mediatek: Add MT8365 support")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241203141823.22393-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3f0b8d367d ]
We should use *-y instead of *-objs in Makefile for the module
objects. *-objs is used rather for host programs.
Fixes: c486def5b3 ("ASoC: cs40l50: Support I2S streaming to CS40L50")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20241203141823.22393-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit ad5b205f9e upstream.
Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
usb 1-1: New USB device found, idVendor=0d8c, idProduct=0014
usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0
usb 1-1: Product: USB Audio Device
usb 1-1: Manufacturer: C-Media Electronics Inc.
Signed-off-by: Lianqin Hu <hulianqin@vivo.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/TYUPR06MB6217E94D922B9BF422A73F32D2192@TYUPR06MB6217.apcprd06.prod.outlook.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
