More driver specific fixes, the firmware change is part of fixing the
race conditions in the Cirrus driver.
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Merge tag 'asoc-fix-v6.14-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.14
More driver specific fixes, the firmware change is part of fixing the
race conditions in the Cirrus driver.
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>
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
Currently, we assume that the PCH DMIC pins are pin-muxed with SoundWire
links. However, we do see a HW design that use PCH DMIC along with 3
SoundWire links. Remove the check and add warning to let users know that
SoundWire MIC and PCH DMIC are both present and they could overwrite it
with kernel params.
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>
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>
Typically, SoundWire MIC and PCH DMIC will not coexist. However, we may
want to use both of them in some special cases. Add a warning to let
users know that SoundWire MIC and PCH DMIC are both present and they
could overwrite it with kernel params.
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>
Link: https://patch.msgid.link/20250225093716.67240-3-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, we assume that the PCH DMIC pins are pin-muxed with SoundWire
links. However, we do see a HW design that use PCH DMIC along with 3
SoundWire links. Remove the check now.
With this change the PCM DMIC will be presented if it is reported by the
BIOS irrespective of whether there are SDW links present or not.
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>
Link: https://patch.msgid.link/20250225093716.67240-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
A few fixes I and James Calligero picked out of the Asahi tree.
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Merge tag 'asoc-fix-v6.14-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.14
A few fixes I and James Calligero picked out of the Asahi tree.
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>
TX launch polarity needs to be the opposite of RX capture polarity, to
generate the right bit slot alignment.
Reviewed-by: Neal Gompa <neal@gompa.dev>
Signed-off-by: Hector Martin <marcan@marcan.st>
Signed-off-by: James Calligeros <jcalligeros99@gmail.com>
Link: https://patch.msgid.link/20250218-apple-codec-changes-v2-28-932760fd7e07@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
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>
This patch reduces the resume time by half and introduces an option to
include a delay after a single write operation before continuing.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://patch.msgid.link/20250214162354.2675652-2-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
More fixes and deviec quirks, most of them driver specific including a
few SOF robustness fixes. Nothing super remarkable individually.
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Merge tag 'asoc-fix-v6.14-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.14
More fixes and deviec quirks, most of them driver specific including a
few SOF robustness fixes. Nothing super remarkable individually.
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>
Because firmware issue of platform, found spi device is not stable,
so add status check before firmware download, and remove some
operations which is not must in current stage.
Signed-off-by: Baojun Xu <baojun.xu@ti.com>
Fixes: bb5f86ea50 ("ALSA: hda/tas2781: Add tas2781 hda SPI driver")
Link: https://patch.msgid.link/20250211083941.5574-1-baojun.xu@ti.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conditional involving sdev->first_boot in acp_sof_ipc_irq_thread()
will succeed only once, i.e. during the very first run of the
DSP firmware.
Use the unlikely() annotation to help improve branch prediction
accuracy.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-4-67824c1e4c9a@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In some cases, e.g. during resuming from suspend, there is a possibility
that some IPC reply messages get received by the host while the DSP
firmware has not yet reached the complete boot state.
Detect when this happens and do not attempt to process the unexpected
replies from DSP. Instead, provide proper debugging support.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-3-67824c1e4c9a@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove all the includes for headers which are not (directly) used from
the Vangogh SOF driver sources.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Reviewed-by: Venkata Prasad Potturu <venkataprasad.potturu@amd.com>
Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-2-67824c1e4c9a@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stress testing resume from suspend on Valve Steam Deck OLED (Galileo)
revealed that the DSP firmware could enter an unrecoverable faulty
state, where the kernel ring buffer is flooded with IPC related error
messages:
[ +0.017002] snd_sof_amd_vangogh 0000:04:00.5: acp_sof_ipc_send_msg: Failed to acquire HW lock
[ +0.000054] snd_sof_amd_vangogh 0000:04:00.5: ipc3_tx_msg_unlocked: ipc message send for 0x30100000 failed: -22
[ +0.000005] snd_sof_amd_vangogh 0000:04:00.5: Failed to setup widget PIPELINE.6.ACPHS1.IN
[ +0.000004] snd_sof_amd_vangogh 0000:04:00.5: Failed to restore pipeline after resume -22
[ +0.000003] snd_sof_amd_vangogh 0000:04:00.5: PM: dpm_run_callback(): pci_pm_resume returns -22
[ +0.000009] snd_sof_amd_vangogh 0000:04:00.5: PM: failed to resume async: error -22
[...]
[ +0.002582] PM: suspend exit
[ +0.065085] snd_sof_amd_vangogh 0000:04:00.5: ipc tx error for 0x30130000 (msg/reply size: 12/0): -22
[ +0.000499] snd_sof_amd_vangogh 0000:04:00.5: error: failed widget list set up for pcm 1 dir 0
[ +0.000011] snd_sof_amd_vangogh 0000:04:00.5: error: set pcm hw_params after resume
[ +0.000006] snd_sof_amd_vangogh 0000:04:00.5: ASoC: error at snd_soc_pcm_component_prepare on 0000:04:00.5: -22
[...]
A system reboot would be necessary to restore the speakers
functionality.
However, by delaying a bit any host to DSP transmission right after
the firmware boot completed, the issue could not be reproduced anymore
and sound continued to work flawlessly even after performing thousands
of suspend/resume cycles.
Introduce the post_fw_run_delay ACP quirk to allow providing the
aforementioned delay via the snd_sof_dsp_ops->post_fw_run() callback for
the affected devices.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://patch.msgid.link/20250207-sof-vangogh-fixes-v1-1-67824c1e4c9a@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
hrtimer_setup() takes the callback function pointer as argument and
initializes the timer completely.
Replace hrtimer_init() and the open coded initialization of
hrtimer::function with the new setup mechanism.
Patch was created by using Coccinelle.
Acked-by: Zack Rusin <zack.rusin@broadcom.com>
Signed-off-by: Nam Cao <namcao@linutronix.de>
Cc: Takashi Iwai <tiwai@suse.com>
Link: https://patch.msgid.link/598031332ce738c82286a158cb66eb7e735b2e79.1738746904.git.namcao@linutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PTL-H uses the same configuration as PTL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250210081730.22916-4-peter.ujfalusi@linux.intel.com
Use same recipes as PTL for PTL-H.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250210081730.22916-3-peter.ujfalusi@linux.intel.com
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The Nullity of sps->cstream needs to be checked in sof_ipc_msg_data() and not
assume that it is not NULL.
The sps->stream must be cleared to NULL on close since this is used as a check
to see if we have active PCM stream.
Report from internal ticket, priv->cali_data.data devm_kzalloc twice,
drop the first one, it is the unnecessary one.
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Link: https://patch.msgid.link/20250206123808.1590-1-shenghao-ding@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In enviornment without KMOD requesting module may fail to load
snd-hda-codec-hdmi, resulting in HDMI audio not usable.
Add softdep to loading HDMI codec module first to ensure we can load it
correctly.
Signed-off-by: Terry Cheong <htcheong@chromium.org>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Johny Lin <lpg76627@gmail.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250206094723.18013-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
For systems which load firmware on the cs35l41 which use ACPI, the
_SUB value is used to differentiate firmware and tuning files for the
individual systems. In the case where a system does not have a _SUB
defined in ACPI node for cs35l41, there needs to be a fallback to
allow the files for that system to be differentiated. Since all
ACPI nodes for cs35l41 should have a HID defined, the HID should be a
safe option.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: André Almeida <andrealmeid@igalia.com>
Tested-by: André Almeida <andrealmeid@igalia.com>
Link: https://patch.msgid.link/20250205164806.414020-1-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Using `fsleep` instead of `msleep` resolves some customer complaints
regarding the precision of up/down DAPM event timing. `fsleep()`
automatically selects the appropriate sleep function, making the delay
time more predictable.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Link: https://patch.msgid.link/20250205160849.500306-1-vitalyr@opensource.cirrus.com
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
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>
Current rsnd driver supports Synchronous SRC Mode, but HW allow to update
rate only within 1% from current rate. Adjust to it.
Becially, this feature is used to fine-tune subtle difference that occur
during sampling rate conversion in SRC. So, it should be called within 1%
margin of rate difference.
If there was difference over 1%, it will apply with 1% increments by using
loop without indicating error message.
Cc: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Tested-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Link: https://patch.msgid.link/871pwd2qe8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_kctrl_accept_runtime() (1) is used for runtime convert rate
(= Synchronous SRC Mode). Now, rsnd driver has 2 kctrls for it
(A): "SRC Out Rate Switch"
(B): "SRC Out Rate" // it calls (1)
(A): can be called anytime
(B): can be called only runtime, and will indicate warning if it was used
at non-runtime.
To use runtime convert rate (= Synchronous SRC Mode), user might uses
command in below order.
(X): > amixer set "SRC Out Rate" on
> aplay xxx.wav &
(Y): > amixer set "SRC Out Rate" 48010 // convert rate to 48010Hz
(Y): calls B
(X): calls both A and B.
In this case, when user calls (X), it calls both (A) and (B), but it is not
yet start running. So, (B) will indicate warning.
This warning was added by commit b5c0886898 ("ASoC: rsnd: add warning
message to rsnd_kctrl_accept_runtime()"), but the message sounds like the
operation was not correct. Let's update warning message.
The message is very SRC specific, implement it in src.c
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Link: https://patch.msgid.link/8734gt2qed.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It will indicate "unsupported clock rate" when setup clock failed.
But it is unclear what kind of rate was failed. Indicate it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh@renesas.com>
Link: https://patch.msgid.link/874j192qej.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
