Age | Commit message (Collapse) | Author | Files | Lines |
|
The alc_spec.power_hook is defined only with CONFIG_PM, and the recent
fix overlooked it, resulting in a build error without CONFIG_PM.
Fix it with the simple ifdef and set __maybe_unused for the function.
We may drop the whole CONFIG_PM dependency there, but it should be
done in a separate cleanup patch later.
Fixes: 1e707769df07 ("ALSA: hda/realtek - Set GPIO3 to default at S4 state for Thinkpad with ALC1318")
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202405012104.Dr7h318W-lkp@intel.com/
Message-ID: <20240502062442.30545-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
This is much larger than is ideal, partly due to your holiday but also
due to several vendors having come in with relatively large fixes at
similar times. It's all driver specific stuff.
The meson fixes from Jerome fix some rare timing issues with blocking
operations happening in triggers, plus the continuous clock support
which fixes clocking for some platforms. The SOF series from Peter
builds to the fix to avoid spurious resets of ChainDMA which triggered
errors in cleanup paths with both PulseAudio and PipeWire, and there's
also some simple new debugfs files from Pierre which make support a lot
eaiser.
|
|
Unfortunately both Lenovo Legion Pro 7 16ARX8H and Legion 7i 16IAX7
got the very same PCI SSID while the hardware implementations are
completely different (the former is with TI TAS2781 codec while the
latter is with Cirrus CS35L41 codec). The former model got broken by
the recent fix for the latter model.
For addressing the regression, check the codec SSID and apply the
proper quirk for each model now.
Fixes: 24b6332c2d4f ("ALSA: hda: Add Lenovo Legion 7i gen7 sound quirk")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.suse.com/show_bug.cgi?id=1223462
Message-ID: <20240430163206.5200-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
There is a chance of damaging the IC when S4 resume.
Add safe mode for no stream to disable GPIO3.
Thinkpad with ALC1318 platform need to add this workaround.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/a853dc4f0a4e412381d5f60565181247@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Merge series from Jerome Brunet <jbrunet@baylibre.com>:
This patchset fixes 2 problems on TDM which both find a solution
by properly implementing the .trigger() callback for the TDM backend.
ATM, enabling the TDM formatters is done by the .prepare() callback
because handling the formatter is slow due to necessary calls to CCF.
The first problem affects the TDMIN. Because .prepare() is called on DPCM
backend first, the formatter are started before the FIFOs and this may
cause a random channel shifts if the TDMIN use multiple lanes with more
than 2 slots per lanes. Using trigger() allows to set the FE/BE order,
solving the problem.
There has already been an attempt to fix this 3y ago [1] and reverted [2]
It triggered a 'sleep in irq' error on the period IRQ. The solution is
to just use the bottom half of threaded IRQ. This is patch #1. Patch #2
and #3 remain mostly the same as 3y ago.
For TDMOUT, the problem is on pause. ATM pause only stops the FIFO and
the TDMOUT just starves. When it does, it will actually repeat the last
sample continuously. Depending on the platform, if there is no high-pass
filter on the analog path, this may translate to a constant position of
the speaker membrane. There is no audible glitch but it may damage the
speaker coil.
Properly stopping the TDMOUT in pause solves the problem. There is
behaviour change associated with that fix. Clocks used to be continuous
on pause because of the problem above. They will now be gated on pause by
default, as they should. The last change introduce the proper support for
continuous clocks, if needed.
[1]: https://lore.kernel.org/linux-amlogic/20211020114217.133153-1-jbrunet@baylibre.com
[2]: https://lore.kernel.org/linux-amlogic/20220421155725.2589089-1-narmstrong@baylibre.com
|
|
The documentation for device_get_named_child_node() mentions this
important point:
"
The caller is responsible for calling fwnode_handle_put() on the
returned fwnode pointer.
"
Add fwnode_handle_put() to avoid a leaked reference.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Fixes: 08c2a4bc9f2a ("ALSA: hda: move Intel SoundWire ACPI scan to dedicated module")
Message-ID: <20240426152731.38420-1-pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The SOF driver is selected whenever specific I2C/I2S HIDs are reported
as 'present' in the ACPI DSDT. In some cases, an HID is reported but
the hardware does not actually rely on I2C/I2S. This false positive
leads to an invalid selection of the SOF driver and as a result an
invalid topology is loaded.
This patch hardens the detection with a check that the NHLT table is
consistent with the report of an I2S-based codec in DSDT. This table
should expose at least one SSP endpoint configured for an I2S-codec
connection.
Tested on Huawei Matebook D14 (NBLB-WAX9N) using an HDaudio codec with
an invalid ES8336 ACPI HID reported:
[ 7.858249] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380
[ 7.858312] snd_hda_intel 0000:00:1f.3: snd_intel_dsp_find_config: no valid SSP found for HID ESSX8336, skipped
Reported-by: Mauro Carvalho Chehab <mchehab@kernel.org>
Tested-by: Mauro Carvalho Chehab <mchehab@kernel.org>
Closes: https://github.com/thesofproject/linux/issues/4934
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Message-ID: <20240426152818.38443-1-pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The documentation for device_get_named_child_node() mentions this
important point:
"
The caller is responsible for calling fwnode_handle_put() on the
returned fwnode pointer.
"
Add fwnode_handle_put() to avoid leaked references.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426152939.38471-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The documentation for device_get_named_child_node() mentions this
important point:
"
The caller is responsible for calling fwnode_handle_put() on the
returned fwnode pointer.
"
Add fwnode_handle_put() to avoid a leaked reference.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240426153033.38500-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Amlogic sound cards do create a lot of pcm interfaces, possibly more than
8. Some pcm interfaces are internal (like DPCM backends and c2c) and not
exposed to userspace.
Those interfaces still increase the number passed to snd_find_free_minor(),
which eventually exceeds 8 causing -EBUSY error on card registration if
CONFIG_SND_DYNAMIC_MINORS=n and the interface is exposed to userspace.
select CONFIG_SND_DYNAMIC_MINORS for Amlogic cards to avoid the problem.
Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426134150.3053741-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Some devices may need the clocks running, even while paused.
Add support for this use case.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426152946.3078805-5-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
So far, the formatters have been reset/enabled using the .prepare()
callback. This was done in this callback because walking the formatters use
a mutex. A mutex is used because formatter handling require dealing
possibly slow clock operation.
With the support of non-atomic, .trigger() callback may be used which also
allows to properly enable and disable formatters on start but also
pause/resume.
This solve a random shift on TDMIN as well repeated samples on for TDMOUT.
Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426152946.3078805-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Non atomic operations need to be performed in the trigger callback
of the TDM interfaces. Those are BEs but what matters is the nonatomic
flag of the FE in the DPCM context. Just set nonatomic for everything so,
at least, what is done is clear.
Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426152946.3078805-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
With the AXG audio subsystem, there is a possible random channel shift on
TDM capture, when the slot number per lane is more than 2, and there is
more than one lane used.
The problem has been there since the introduction of the axg audio support
but such scenario is pretty uncommon. This is why there is no loud
complains about the problem.
Solving the problem require to make the links non-atomic and use the
trigger() callback to start FEs and BEs in the appropriate order.
This was tried in the past and reverted because it caused the block irq to
sleep while atomic. However, instead of reverting, the solution is to call
snd_pcm_period_elapsed() in a non atomic context.
Use the bottom half of a threaded IRQ to do so.
Fixes: 6dc4fa179fb8 ("ASoC: meson: add axg fifo base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20240426152946.3078805-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch simply add SND_PCI_QUIRK for HP Laptop 15-da3001TU to fixed
mute led of laptop.
Signed-off-by: Aman Dhoot <amandhoot12@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAMTp=B+3NG65Z684xMwHqdXDJhY+DJK-kuSw4adn6xwnG+b5JA@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We did not delay after the second strobe signal, so another immediately
following access could potentially corrupt the written value.
This is a purely speculative fix with no supporting evidence, but after
taking out the spinlocks around the writes, it seems plausible that a
modern processor could be actually too fast. Also, it's just cleaner to
be consistent.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-7-oswald.buddenhagen@gmx.de>
|
|
A side effect of making the dock monitoring interrupt-driven was that
we'd be very quick to program a freshly connected dock. However, for
unclear reasons, the dock does not work when we do that - despite the
FPGA netlist upload going just fine. We work around this by adding a
delay before programming the dock; for safety, the value is several
times as much as was determined empirically.
Note that a badly timed dock hot-plug would have triggered the problem
even before the referenced commit - but now it would happen 100% instead
of about 3% of the time, thus making it impossible to work around by
re-plugging.
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-6-oswald.buddenhagen@gmx.de>
|
|
The FPGA access through the GPIO port does not interfere with other
sound processor register access, so there is no need to subject it to
emu_lock. And after moving all FPGA access out of the interrupt handler,
it does not need to be IRQ-safe, either.
What's more, attaching the dock causes a firmware upload, which takes
several seconds. We really don't want to disable IRQs for this long, and
even less also have someone else spin with IRQs disabled waiting for us.
Therefore, use a mutex for FPGA access locking.
This makes the code somewhat more noisy, as we need to wrap bigger
sections into the mutex, as it needs to enclose the spinlocks.
The latter has the "side effect" of fixing dock FPGA programming in a
corner case: a really badly timed mixer access right between entering
FPGA programming mode and uploading the netlist would mess up the
protocol.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-5-oswald.buddenhagen@gmx.de>
|
|
The actual event processing was already done by workqueue items. We can
move the event dispatching there as well, rather than doing it already
in the interrupt handler callback.
This change has a rather profound "side effect" on the reliability of
the FPGA programming: once we enter programming mode, we must not issue
any snd_emu1010_fpga_{read,write}() calls until we're done, as these
would badly mess up the programming protocol. But exactly that would
happen when trying to program the dock, as that triggers GPIO interrupts
as a side effect. This is mitigated by deferring the actual interrupt
handling, as workqueue items are not re-entrant.
To avoid scheduling the dispatcher on non-events, we now explicitly
ignore GPIO IRQs triggered by "uninteresting" pins, which happens a lot
as a side effect of calling snd_emu1010_fpga_{read,write}().
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-4-oswald.buddenhagen@gmx.de>
|
|
Pulled out of the next patch to improve its legibility.
As the function is now available, call it directly from
snd_emu10k1_emu1010_init(), thus making the MicroDock firmware loading
synchronous - there isn't really a reason not to. Note that this does
not affect the AudioDocks of rev1 cards, as these have no independent
power supplies, and thus come up only a while after the main card is
initialized.
As a drive-by, adjust the priorities of two messages to better reflect
their impact.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-3-oswald.buddenhagen@gmx.de>
|
|
While there are two separate IRQ status bits for dock attach and detach,
the hardware appears to mix them up more or less randomly, making them
useless for tracking what actually happened. It is much safer to check
the dock status separately and proceed based on that, as the old polling
code did.
Note that the code assumes that only the dock can be hot-plugged - if
other option card bits changed, the logic would break.
Fixes: fbb64eedf5a3 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-2-oswald.buddenhagen@gmx.de>
|
|
Volume step (dB/step) modification to fix format error
which shown in amixer control.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://lore.kernel.org/r/b1f546ad16dc4c7abb7daa7396e8345c@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Static checkers complain that the silicon_uid variable passed by
pointer to cs35l56_read_silicon_uid() could later be used
uninitialised when calling cs_amp_get_efi_calibration_data().
cs35l56_read_silicon_uid() must have succeeded to call
cs_amp_get_efi_calibration_data() and that would have populated the
variable.
However, initialise the value so we are not haunted by it forevermore.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: e1830f66f6c6 ("ASoC: cs35l56: Add helper functions for amp calibration")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240422103211.236063-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
WSA881x codecs do not retain the state while clock is stopped, so mark
this with clk_stop_mode1 flag.
Fixes: a0aab9e1404a ("ASoC: codecs: add wsa881x amplifier support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20240419140012.91384-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The conversion from MIDI2 to MIDI1 UMP messages had a leftover
artifact (superfluous bit shift), and this resulted in the bogus type
check, leading to empty outputs. Let's fix it.
Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events")
Cc: <stable@vger.kernel.org>
Link: https://github.com/alsa-project/alsa-utils/issues/262
Message-ID: <20240419100442.14806-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The Haier Boyue G42 with ALC269VC cannot detect the MIC of headset,
the line out and internal speaker until
ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Cc: <stable@vger.kernel.org>
Message-ID: <20240419082159.476879-1-aichao@kylinos.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
When using davinci-mcasp as CPU DAI with simple-card, there are some
conditions that cause simple-card to finish registering a sound card before
davinci-mcasp finishes registering all sound components. This creates a
non-working sound card from userspace with no problem indication apart
from not being able to play/record audio on a PCM stream. The issue
arises during simultaneous probe execution of both drivers. Specifically,
the simple-card driver, awaiting a CPU DAI, proceeds as soon as
davinci-mcasp registers its DAI. However, this process can lead to the
client mutex lock (client_mutex in soc-core.c) being held or davinci-mcasp
being preempted before PCM DMA registration on davinci-mcasp finishes.
This situation occurs when the probes of both drivers run concurrently.
Below is the code path for this condition. To solve the issue, defer
davinci-mcasp CPU DAI registration to the last step in the audio part of
it. This way, simple-card CPU DAI parsing will be deferred until all
audio components are registered.
Fail Code Path:
simple-card.c: probe starts
simple-card.c: simple_dai_link_of: simple_parse_node(..,cpu,..) returns EPROBE_DEFER, no CPU DAI yet
davinci-mcasp.c: probe starts
davinci-mcasp.c: devm_snd_soc_register_component() register CPU DAI
simple-card.c: probes again, finish CPU DAI parsing and call devm_snd_soc_register_card()
simple-card.c: finish probe
davinci-mcasp.c: *dma_pcm_platform_register() register PCM DMA
davinci-mcasp.c: probe finish
Cc: stable@vger.kernel.org
Fixes: 9fbd58cf4ab0 ("ASoC: davinci-mcasp: Choose PCM driver based on configured DMA controller")
Signed-off-by: Joao Paulo Goncalves <joao.goncalves@toradex.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@gmail.com>
Reviewed-by: Jai Luthra <j-luthra@ti.com>
Link: https://lore.kernel.org/r/20240417184138.1104774-1-jpaulo.silvagoncalves@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
When creating controls attached to widgets, there are a lot of rules if
they get their name prefixed with widget name or not. Due to that
controls ended up with weirdly looking names like "ssp0_fe DSP Volume",
while topology set it to "DSP Volume".
Fix this by setting no_wname_in_kcontrol_name to true in avs topology
widgets which disables unwanted behaviour.
Fixes: be2b81b519d7 ("ASoC: Intel: avs: Parse control tuples")
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20240418142621.2487478-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The headset mic requires a fixup to be properly detected/used.
As a reference, this specific model from 2021 reports
the following devices:
https://alsa-project.org/db/?f=1a5ddeb0b151db8fe051407f5bb1c075b7dd3e4a
Signed-off-by: Mauro Carvalho Chehab <mchehab@kernel.org>
Cc: <stable@vger.kernel.org>
Message-ID: <b92a9e49fb504eec8416bcc6882a52de89450102.1713370457.git.mchehab@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
change HDA & AMP configuration from ALC287_FIXUP_CS35L41_I2C_2 to
ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD for ThinkBook 16P Gen4
models to fix volumn control issue (cannot fully mute).
Signed-off-by: Huayu Zhang <zhanghuayu1233@qq.com>
Fixes: 6214e24cae9b ("ALSA: hda/realtek: Add quirks for Lenovo Thinkbook 16P laptops")
Message-ID: <tencent_37EB880C5E5BD99D21C16B288115C4545F06@qq.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Added the correct pin table for Asus GU605M and GA403U, enabling all
speakers to be controlled with the master.
Updated quirks for GU605M and GA403U by including the pin table patch
in the chain.
Co-developed-by: Luke D. Jones <luke@ljones.dev>
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Signed-off-by: Vitalii Torshyn <vitaly.torshyn@gmail.com>
Message-ID: <20240411125803.18539-1-vitaly.torshyn@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add new vendor_id and subsystem_id to support new Lenovo laptop
ThinkPad ICE-1
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240411091823.1644-1-shenghao-ding@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The patch which fixed the missing remove_late() calls missed a case
when sof_select_ipc_and_paths() could return with error and in this
case sof_init_environment() would just return with 0.
Do not ignore the error code returned by sof_select_ipc_and_paths().
Fixes: 90f8917e7a15 ("ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20240417075804.10829-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add vendor clear control register in readable register's
callback function. This prevents an access failure reported
in Intel CI tests.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Closes: https://github.com/thesofproject/linux/issues/4860
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/6a103ce9134d49d8b3941172c87a7bd4@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Currently, all ASoC systems are set to use VPMON for DSP1RX5_SRC,
however, this is required only for internal boost systems.
External boost systems require VBSTMON instead of VPMON to be the
input to DSP1RX5_SRC.
Shared Boost Active acts like Internal boost (requires VPMON).
Shared Boost Passive acts like External boost (requires VBSTMON)
All systems require DSP1RX6_SRC to be set to VBSTMON.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240411142648.650921-1-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
This chain fixes some problems with some previous patches for handling
the ASP1 config registers. The root of the problem is that the ownership
of these registers can be either with the firmware or the driver, and that
the chip has to be soft-reset after downloading the firmware.
This chain adds and uses a regmap_read_bypassed() function so that the
driver can leave the regmap in cache-only until the chip has rebooted,
but still poll a register to detect when the chip has rebooted.
Richard Fitzgerald (4):
regmap: Add regmap_read_bypassed()
ALSA: hda: cs35l56: Exit cache-only after
cs35l56_wait_for_firmware_boot()
ASoC: cs35l56: Fix unintended bus access while resetting amp
ASoC: cs35l56: Prevent overwriting firmware ASP config
drivers/base/regmap/regmap.c | 37 ++++++++++++++
include/linux/regmap.h | 8 +++
include/sound/cs35l56.h | 2 +
sound/pci/hda/cs35l56_hda.c | 4 ++
sound/soc/codecs/cs35l56-sdw.c | 2 -
sound/soc/codecs/cs35l56-shared.c | 83 ++++++++++++++++++++-----------
sound/soc/codecs/cs35l56.c | 26 +++++++++-
7 files changed, 130 insertions(+), 32 deletions(-)
--
2.39.2
|
|
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The current code will reset the ChainDMA on release unconditionally which
can result the following error when the CHainDMA is not allocated:
ipc tx : 0xe040000|0x0: GLB_CHAIN_DMA
ipc tx reply: 0x2e000007|0x0: GLB_CHAIN_DMA
FW reported error: 7 - Unsupported operation requested
ipc error for msg 0xe040000|0x0
sof_pcm_stream_free: pcm_ops hw_free failed -22
Background:
Pulseaudio and Pipewire on startup opens all available streams and
closes them without triggering a start (after probing it's capabilities).
|
|
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
We somehow missed the default path for DSP libraries for LNL, and need
to restrict support for D0i3 w/ IPC4. Also add debugfs support for
firmware profile information so that sof-test scripts can show what is
being tested.
|
|
Add vrefo settings to fix jd and headset mic recording issue.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://msgid.link/r/727219ed45d3485ba8f4646700aaa8a8@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Channel numbers of dmic supports 4 channels, modify channels_max
regarding to this issue.
Signed-off-by: Jack Yu <jack.yu@realtek.com>
Link: https://msgid.link/r/6a9b1d1fb2ea4f04b2157799f04053b1@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The codec leaves tie combo jack's sleeve/ring2 to floating status
default. It would cause electric noise while connecting the active
speaker jack during boot or shutdown.
This patch requests a gpio to control the additional jack circuit
to tie the contacts to the ground or floating.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://msgid.link/r/20240408091057.14165-1-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch adds microphone detection for the Acer 315-24p, after which a microphone appears on the device and starts working
Signed-off-by: end.to.start <end.to.start@mail.ru>
Link: https://msgid.link/r/20240408152454.45532-1-end.to.start@mail.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The SOF driver has multiple profiles to select firmware/topology
prefix/names depending on the platform and ipc_type, and each of those
fields can be overridden with kernel parameters. This results in some
cases in confusion on what configuration is actually used in a given
test.
We currently log the firmware and topology names in the kernel logs,
but there's been an ask to add the information in debugfs to simplify
test scripts used by developers and CI.
This isn't meant to be a stable ABI used by apps, changes will be
allowed as needed.
Closes: https://github.com/thesofproject/linux/issues/3867
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240408194147.28919-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Introduce a new field in struct sof_ipc_pcm_ops that can be used to
restrict DSP D0i3 during S0ix suspend to IPC3. With IPC4, all streams
must be stopped before S0ix suspend.
Reviewed-by: Uday M Bhat <uday.m.bhat@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240408194147.28919-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The commit cd6f2a2e6346 ("ASoC: SOF: Intel: Set the default firmware
library path for IPC4") added the default_lib_path field for all
platforms, but this was missed when LunarLake was later introduced.
Fixes: 64a63d9914a5 ("ASoC: SOF: Intel: LNL: Add support for Lunarlake platform")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240408194147.28919-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The ChainDMA operation differs from normal pipelines that it is only
created when the stream started, in fact a PCM using ChainDMA has no
pipelines, modules.
To reset a ChainDMA, it needs to be first allocated in firmware. When
PulseAudio/PipeWire starts, they will probe the PCMs by opening them, check
hw_params and then close the PCM without starting audio.
Unconditionally resetting the ChainDMA can result the following error:
ipc tx : 0xe040000|0x0: GLB_CHAIN_DMA
ipc tx reply: 0x2e000007|0x0: GLB_CHAIN_DMA
FW reported error: 7 - Unsupported operation requested
ipc error for msg 0xe040000|0x0
sof_pcm_stream_free: pcm_ops hw_free failed -22
Add a new chain_dma_allocated flag to sof_ipc4_pcm_stream_priv to store the
ChainDMA allocation state and use this flag to skip sending the reset if
the ChainDMA is not allocated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240409110036.9411-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Using the sof_ipc4_timestamp_info struct directly as sps->private data
is too restrictive, add a new generic sof_ipc4_pcm_stream_priv struct
containing the time_info to allow new information to be stored in a
generic way.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240409110036.9411-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The pointer to sof_ipc4_timestamp_info named most of the time as
'time_info' only to be named as 'stream_info' or 'info' in two function.
Use the consistent name of 'time_info' throughout the file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240409110036.9411-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Throughout the file the pointer for snd_sof_pcm_stream is named either
'stream' or (wrongly) 'spcm' which confuses the reader.
Use 'sps' for the pointer name as it is the most common name used in SOF
codebase.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240409110036.9411-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Only populate the ASP1 config registers in the regmap cache if the
ASP DAI is used. This prevents regcache_sync() from overwriting
these registers with their defaults when the firmware owns
control of these registers.
On a SoundWire system the ASP could be owned by the firmware to
share reference audio with the firmware on other cs35l56. Or it
can be used as a normal codec-codec interface owned by the driver.
The driver must not overwrite the registers if the firmware has
control of them.
The original implementation for this in commit 07f7d6e7a124
("ASoC: cs35l56: Fix for initializing ASP1 mixer registers") was
to still provide defaults for these registers, assuming that if
they were never reconfigured from defaults then regcache_sync()
would not write them out because they are not dirty. Unfortunately
regcache_sync() is not that smart. If the chip has not reset (so
the driver has not called regcache_mark_dirty()) a regcache_sync()
could write out registers that are not dirty.
To avoid accidental overwriting of the ASP registers, they are
removed from the table of defaults and instead are populated with
defaults only if one of the ASP DAI configuration functions is
called. So if the DAI has never been configured, the firmware is
assumed to have ownership of these registers, and the regmap cache
will not contain any entries for them.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 07f7d6e7a124 ("ASoC: cs35l56: Fix for initializing ASP1 mixer registers")
Link: https://msgid.link/r/20240408101803.43183-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Use the new regmap_read_bypassed() so that the regmap can be left
in cache-only mode while it is booting, but the driver can still
read boot-status and chip-id information during this time.
This fixes race conditions where some writes could be issued to the
silicon while it is still rebooting, before the driver has determined
that the boot is complete.
This is typically prevented by putting regmap into cache-only until the
hardware is ready. But this assumes that the driver does not need to
access device registers to determine when it is "ready". For cs35l56
this involves polling a register and the original implementation relied
on having special handlers to block racing callbacks until dsp_work()
is complete. However, some cases were missed, most notably the ASP DAI
functions.
The regmap_read_bypassed() function allows the fix for this to be
simplified to putting regmap into cache-only during the reset. The
initial boot stages (poll HALO_STATE and read the chip ID) are all done
bypassed. Only when the amp is seen to be booted is the cache-only
revoked.
Changes are:
- cs35l56_system_reset() now leaves the regmap in cache-only status.
- cs35l56_wait_for_firmware_boot() polls using regmap_read_bypassed().
- cs35l56_init() revokes cache-only either via cs35l56_hw_init() or
when firmware has rebooted after a soft reset.
- cs35l56_hw_init() exits cache-only after it has determined that the
amp has booted.
- cs35l56_sdw_init() doesn't disable cache-only, since this must be
deferred to cs35l56_init().
- cs35l56_runtime_resume_common() waits for firmware boot before exiting
cache-only.
These changes cover three situations where the registers are not
accessible:
1) SoundWire first-time enumeration. The regmap is kept in cache-only
until the chip is fully booted. The original code had to exit
cache-only to read chip status in cs35l56_init() and cs35l56_hw_init()
but this is now deferred to after the firmware has rebooted.
In this case cs35l56_sdw_probe() leaves regmap in cache-only
(unchanged behaviour) and cs35l56_hw_init() exits cache-only after the
firmware is booted and the chip identified.
2) Soft reset during first-time initialization. cs35l56_init() calls
cs35l56_system_reset(), which puts regmap into cache-only.
On I2C/SPI cs35l56_init() then flows through to call
cs35l56_wait_for_firmware_boot() and exit cache-only. On SoundWire
the re-enumeration will enter cs35l56_init() again, which then drops
down to call cs35l56_wait_for_firmware_boot() and exit cache-only.
3) Soft reset after firmware download. dsp_work() calls
cs35l56_system_reset(), which puts regmap into cache-only. After this
the flow is the same as (2).
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file")
Link: https://msgid.link/r/20240408101803.43183-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Adds calls to disable regmap cache-only after a successful return from
cs35l56_wait_for_firmware_boot().
This is to prepare for a change in the shared ASoC module that will
leave regmap in cache-only mode after cs35l56_system_reset(). This is
to prevent register accesses going to the hardware while it is
rebooting.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240408101803.43183-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The Asus T100TA quirk has been using an exact match on a product-name of
"T100TA" but there are also T100TAM variants with a slightly higher
clocked CPU and a metal backside which need the same quirk.
Sort the existing T100TA (stereo speakers) below the more specific
T100TAF (mono speaker) quirk and switch from exact matching to
substring matching so that the T100TA quirk will also match on
the T100TAM models.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://msgid.link/r/20240407191559.21596-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
DSPK configuration is wrong for 16-bit playback and this happens because
the client config is always fixed at 24-bit in hw_params(). Fix this by
updating the client config to 16-bit for the respective playback.
Fixes: 327ef6470266 ("ASoC: tegra: Add Tegra186 based DSPK driver")
Cc: stable@vger.kernel.org
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Link: https://msgid.link/r/20240405104306.551036-1-spujar@nvidia.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Recent changes addressed PAGE_SIZE ambiguity in 2/3 locations for struct
avs_icl_memwnd2. The unaddressed one causes build errors when
PAGE_SIZE != SZ_4K.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202404070100.i3t3Jf7d-lkp@intel.com/
Fixes: 275b583d047a ("ASoC: Intel: avs: ICL-based platforms support")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240408081840.1319431-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Calibrated data was written into an incorrect register, which cause
speaker protection sometimes malfuctions
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Shenghao Ding <shenghao-ding@ti.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240406132010.341-1-shenghao-ding@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add support for HP SnowWhite laptops with CS35L51 amplifiers on I2C
bus connected to Realtek codec.
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Message-ID: <20240405210635.22193-1-vitalyr@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A relatively large set of fixes here, the biggest piece of it is a
series correcting some problems with the delay reporting for Intel SOF
cards but there's a bunch of other things. Everything here is driver
specific except for a fix in the core for an issue with sign extension
handling volume controls.
|
|
In cases where the sof driver is unable to find the firmware and/or
topology file [1], it exits without releasing the i915 runtime
pm wakeref [2]. This results in dmesg warnings[3] during
suspend/resume or driver unbind. Add remove_late() to the failure path
of sof_init_environment so that i915 wakeref is released appropriately
[1]
[ 8.990366] sof-audio-pci-intel-mtl 0000:00:1f.3: SOF firmware and/or topology file not found.
[ 8.990396] sof-audio-pci-intel-mtl 0000:00:1f.3: Supported default profiles
[ 8.990398] sof-audio-pci-intel-mtl 0000:00:1f.3: - ipc type 1 (Requested):
[ 8.990399] sof-audio-pci-intel-mtl 0000:00:1f.3: Firmware file: intel/sof-ipc4/mtl/sof-mtl.ri
[ 8.990401] sof-audio-pci-intel-mtl 0000:00:1f.3: Topology file: intel/sof-ace-tplg/sof-mtl-rt711-2ch.tplg
[ 8.990402] sof-audio-pci-intel-mtl 0000:00:1f.3: Check if you have 'sof-firmware' package installed.
[ 8.990403] sof-audio-pci-intel-mtl 0000:00:1f.3: Optionally it can be manually downloaded from:
[ 8.990404] sof-audio-pci-intel-mtl 0000:00:1f.3: https://github.com/thesofproject/sof-bin/
[ 8.999088] sof-audio-pci-intel-mtl 0000:00:1f.3: error: sof_probe_work failed err: -2
[2]
ref_tracker: 0000:00:02.0@ffff9b8511b6a378 has 1/5 users at
track_intel_runtime_pm_wakeref.part.0+0x36/0x70 [i915]
__intel_runtime_pm_get+0x51/0xb0 [i915]
intel_runtime_pm_get+0x17/0x20 [i915]
intel_display_power_get+0x2f/0x70 [i915]
i915_audio_component_get_power+0x23/0x120 [i915]
snd_hdac_display_power+0x89/0x130 [snd_hda_core]
hda_codec_i915_init+0x3f/0x50 [snd_sof_intel_hda]
hda_dsp_probe_early+0x170/0x250 [snd_sof_intel_hda_common]
snd_sof_device_probe+0x224/0x320 [snd_sof]
sof_pci_probe+0x15b/0x220 [snd_sof_pci]
hda_pci_intel_probe+0x30/0x70 [snd_sof_intel_hda_common]
local_pci_probe+0x4c/0xb0
pci_device_probe+0xcc/0x250
really_probe+0x18e/0x420
__driver_probe_device+0x7e/0x170
driver_probe_device+0x23/0xa0
[3]
[ 484.105070] ------------[ cut here ]------------
[ 484.108238] thunderbolt 0000:00:0d.2: PM: pci_pm_suspend_late+0x0/0x50 returned 0 after 0 usecs
[ 484.117106] i915 0000:00:02.0: i915 raw-wakerefs=1 wakelocks=1 on cleanup
[ 484.792005] WARNING: CPU: 2 PID: 2405 at drivers/gpu/drm/i915/intel_runtime_pm.c:444 intel_runtime_pm_driver_release+0x6c/0x80
Tested-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Chaitanya Kumar Borah <chaitanya.kumar.borah@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Lucas De Marchi <lucas.demarchi@intel.com>
Link: https://github.com/thesofproject/linux/pull/4878
Signed-off-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Link: https://msgid.link/r/20240404184813.134566-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Before ACP firmware loading, DSP interrupts are not expected.
Sometimes after reboot, it's observed that before ACP firmware is loaded
false DSP interrupt is reported.
Registering the interrupt handler before acp initialization causing false
interrupts sometimes on reboot as ACP reset is not applied.
Correct the sequence by invoking acp initialization sequence prior to
registering interrupt handler.
Fixes: 738a2b5e2cc9 ("ASoC: SOF: amd: Add IPC support for ACP IP block")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240404041717.430545-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
During probe the DMIC/SSP offload is enabled and it is not reversed on
remove.
Add a remove wrapper for LNL to disable the offload for DMIC and SSP
similarly to what is done during probe.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240403111839.27259-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This v6.8 change didn't make it into the release, send it as a fix for
v6.9.
|
|
Merge series from Zhang Yi <zhangyi@everest-semi.com>:
We solved some issues related to headphone detection.And for using
the same configuration in different power conditions,we modified the
clock table
|
|
Modpost warns about missing module description, add it.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://msgid.link/r/20240402130640.3310999-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We removed the configuration of ES8326_ADC_SCALE
in es8326_jack_detect_handler because user changed
the configuration by snd_controls
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-5-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We got a headphone detection issue after suspend and resume.
And we fixed it by modifying the configuration at es8326_suspend
and invoke es8326_irq at es8326_resume.
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-4-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We got a digital microphone feature issue. And we fixed it by modifying
the clock table. Also, we changed the marco ES8326_CLK_ON declaration
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-3-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We got an error report about headphone type detection and button detection.
We fixed the headphone type detection error by adjusting the debounce timer
configuration. And we fixed the button detection error by disabling the
button detection feature when the headphone are unplugged and enabling it
when headphone are plugged in.
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240402062043.20608-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
For shutting up spurious KMSAN uninit-value warnings, just replace
kmalloc() calls with kzalloc() for the buffers used for
communications. There should be no real issue with the original code,
but it's still better to cover.
Reported-by: syzbot+7fb05ccf7b3d2f9617b3@syzkaller.appspotmail.com
Closes: https://lore.kernel.org/r/00000000000084b18706150bcca5@google.com
Message-ID: <20240402063628.26609-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fixes the realtek quirk to initialise the Cirrus amp correctly and adds
related quirk for missing DSD properties. This model laptop has slightly
updated internals compared to the previous version with Realtek Codec
ID of 0x1caf.
Signed-off-by: Luke D. Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
Message-ID: <20240402015126.21115-1-luke@ljones.dev>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
microphone
This patch addresses an issue with the Panasonic CF-SZ6's existing quirk,
specifically its headset microphone functionality. Previously, the quirk
used ALC269_FIXUP_HEADSET_MODE, which does not support the CF-SZ6's design
of a single 3.5mm jack for both mic and audio output effectively. The
device uses pin 0x19 for the headset mic without jack detection.
Following verification on the CF-SZ6 and discussions with the original
patch author, i determined that the update to
ALC269_FIXUP_ASPIRE_HEADSET_MIC is the appropriate solution. This change
is custom-designed for the CF-SZ6's unique hardware setup, which includes
a single 3.5mm jack for both mic and audio output, connecting the headset
microphone to pin 0x19 without the use of jack detection.
Fixes: 0fca97a29b83 ("ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk")
Signed-off-by: I Gede Agastya Darma Laksana <gedeagas22@gmail.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240401174602.14133-1-gedeagas22@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This fixes the sound not working from internal speakers on
Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID
have been added to cs35l41_hda_property.c and patch_realtek.c.
Signed-off-by: Christian Bendiksen <christian@bendiksen.me>
Cc: <stable@vger.kernel.org>
Message-ID: <20240401122603.6634-1-christian@bendiksen.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
As already anticipated in the original commit, playback was broken for
very short samples. I just didn't expect it to be an actual problem,
because we're talking about less than 1.5 milliseconds here. But clearly
such wavetable samples do actually exist.
The problem was that for such short samples we'd set the current
position beyond the end of the loop, so we'd run off the end of the
sample and play garbage.
This is a bigger (more audible) problem than the original one, which was
that we'd start playback with garbage (whatever was still in the cache),
which would be mostly masked by the note's attack phase.
So revert to the old behavior for now. We'll subsequently fix it
properly with a bigger patch series.
Note that this isn't a full revert - the dead code is not re-introduced,
because that would be silly.
Fixes: df335e9a8bcb ("ALSA: emu10k1: fix synthesizer sample playback position and caching")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240401145805.528794-1-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
mismatch
As described in the added code comment, a reference to .exit.text is ok
for drivers registered via module_platform_driver_probe(). Make this
explicit to prevent the following section mismatch warning
WARNING: modpost: sound/oss/dmasound/dmasound_paula: section mismatch in reference: amiga_audio_driver+0x8 (section: .data) -> amiga_audio_remove (section: .exit.text)
that triggers on an allmodconfig W=1 build.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Message-ID: <c216a129aa88f3af5c56fe6612a472f7a882f048.1711748999.git.u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
These ASUS laptops use the Realtek HDA codec combined with a number of
CS35L56 amplifiers.
The SSID of the GA403U matches a previous ASUS laptop - we can tell them
apart because they use different codecs.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Message-ID: <20240329112803.23897-1-simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
If acp_init() fails, acp pci driver probe should return error.
Add acp_init() function return value check logic.
Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20240329053815.2373979-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
Fix a set of problematic locking sequences and update error messages,
tested on SOF/SoundWire platforms.
|
|
In snd_soc_info_volsw(), mask is generated by figuring out the index of
the most significant bit set in max and converting the index to a
bitmask through bit shift 1. Unintended wraparound occurs when max is an
integer value with msb bit set. Since the bit shift value 1 is treated
as an integer type, the left shift operation will wraparound and set
mask to 0 instead of all 1's. In order to fix this, we type cast 1 as
`1ULL` to prevent the wraparound.
Fixes: 7077148fb50a ("ASoC: core: Split ops out of soc-core.c")
Signed-off-by: Stephen Lee <slee08177@gmail.com>
Link: https://msgid.link/r/20240326010131.6211-1-slee08177@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The drivers for Realtek SoundWire codecs use similar logs, which is
problematic to analyze problems reported by CI tools, e.g. "Failed to
get private value: 752001 => 0000 ret=-5". It's not uncommon to have
several Realtek devices on the same platform, having the same log
thrown makes support difficult.
This patch adds __func__ to all error logs which didn't already
include it.
No functionality change, only error logs are modified.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: a0b7c59ac1a9 ("ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 7a8735c1551e ("ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: b69de265bd0e ("ASoC: rt711: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 23adeb7056ac ("ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The disable_irq_lock protects the 'disable_irq' value, we need to lock
before testing it.
Fixes: 02fb23d72720 ("ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Chao Song <chao.song@linux.intel.com>
Link: https://msgid.link/r/20240325221817.206465-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Adding the ACPI HIDs to the match table triggers the cs35l56-hda modules
to be loaded on boot so that Serial Multi Instantiate can add the
devices to the bus and begin the driver init sequence.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Message-ID: <20240328121355.18972-1-simont@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds the existing fixup to certain TF platforms implementing
the ALC274 codec with a headset jack. It fixes/activates the inactive
microphone of the headset.
Signed-off-by: Christoffer Sandberg <cs@tuxedo.de>
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Message-ID: <20240328102757.50310-1-wse@tuxedocomputers.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
ACP PDM configuration has to be verified for all combinations.
Remove FLAG_AMD_LEGACY_ONLY_DMIC check.
Fixes: 3a94c8ad0aae ("ASoC: amd: acp: add code for scanning acp pdm controller")
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://msgid.link/r/20240327104657.3537664-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The debug message "Playback action not supported: action" is not useful,
because the action was previously printed, and the list of supported
actions are intentional.
Remove the debug statement from the default switch case.
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <8b9546db6c92dea4476a7247a88d56248c2ba8c2.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Sometimes it is useful to examine the timing of kcontrol events.
Add debug statements to each kcontrol.
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <18ff4b0caab90a2dacf907e62346fd5079a9eb1a.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The rcabin.profile_cfg_id, cur_prog, cur_conf, force_fwload_status
variables are acccessible from multiple threads and therefore require
locking.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <e35b867f6fe5fa1f869dd658a0a1f2118b737f57.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The "Speaker Digital Gain" kcontrol controls the TAS2781_DVC_LVL (0x1A)
register. Unfortunately the tas2563 does not have DVC_LVL, but has
INT_MASK0 in 0x1A, which has been misused so far.
Since commit c1947ce61ff4 ("ALSA: hda/realtek: tas2781: enable subwoofer
volume control") the volume of the tas2781 amplifiers can be controlled
by the master volume, so this digital gain kcontrol is not needed.
Remove it.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
CC: stable@vger.kernel.org
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Message-ID: <741fc21db994efd58f83e7aef38931204961e5b2.1711469583.git.soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
clang warns about what it interprets as a truncated snprintf:
sound/aoa/soundbus/i2sbus/core.c:171:6: error: 'snprintf' will always be truncated; specified size is 6, but format string expands to at least 7 [-Werror,-Wformat-truncation-non-kprintf]
The actual problem here is that it does not understand the special
%pOFn format string and assumes that it is a pointer followed by
the string "OFn", which would indeed not fit.
Slightly increasing the size of the buffer to its natural alignment
avoids the warning, as it is now long enough for the correct and
the incorrect interprations.
Fixes: b917d58dcfaa ("ALSA: aoa: Convert to using %pOFn instead of device_node.name")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Message-ID: <20240326223825.4084412-9-arnd@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The current version of delay reporting code can report incorrect
values when paired with a firmware which enables this feature.
Unfortunately there are several smaller issues that needed to be addressed
to correct the behavior:
Wrong information was used for the host side of counter
For MTL/LNL used incorrect (in a sense that it was verified only on MTL)
link side counter function.
The link side counter needs compensation logic if pause/resume is used.
The offset values were not refreshed from firmware.
Finally, not strictly connected, but the ALSA buffer size needs to be
constrained to avoid constant xrun from media players (like mpv)
The series applies cleanly for 6.9 and 6.8.y stable, but older stable
would need manual backport, but it is questionable if it is needed as
MTL/LNL is missing features.
|
|
The current code is pulling the wrong pointer causing it to disable the
wrong IRQ. Correct the code to pull the correct cs42l43 core data
pointer.
Fixes: 64353af49fec ("ASoC: cs42l43: Add system suspend ops to disable IRQ")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://msgid.link/r/20240326105434.852907-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The dreamcastcard->timer could schedule the spu_dma_work and the
spu_dma_work could also arm the dreamcastcard->timer.
When the snd_pcm_substream is closing, the aica_channel will be
deallocated. But it could still be dereferenced in the worker
thread. The reason is that del_timer() will return directly
regardless of whether the timer handler is running or not and
the worker could be rescheduled in the timer handler. As a result,
the UAF bug will happen. The racy situation is shown below:
(Thread 1) | (Thread 2)
snd_aicapcm_pcm_close() |
... | run_spu_dma() //worker
| mod_timer()
flush_work() |
del_timer() | aica_period_elapsed() //timer
kfree(dreamcastcard->channel) | schedule_work()
| run_spu_dma() //worker
... | dreamcastcard->channel-> //USE
In order to mitigate this bug and other possible corner cases,
call mod_timer() conditionally in run_spu_dma(), then implement
PCM sync_stop op to cancel both the timer and worker. The sync_stop
op will be called from PCM core appropriately when needed.
Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device")
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Duoming Zhou <duoming@zju.edu.cn>
Message-ID: <20240326094238.95442-1-duoming@zju.edu.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The original timestamp is built base on windows epoch time which is not
fit for Linux system and difficult to be used for kernel debugging. This
patch adopts syslog timestamp so that we can simply use dmesg to check
the timestamp between fw and kernel.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://msgid.link/r/20240322112703.4549-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
When a wmfw file has not been loaded the firmware control descriptions
necessary to write a stored calibration are not present. In this case
print a more descriptive error message.
The message is logged at info level because it is not fatal, and does
not necessarily imply that anything is broken.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240325144450.293630-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
SoCs with ACE architecture are tailored to use s2idle instead deep (S3)
suspend state and the IMR content is lost when the system is forced to
enter even to S3.
When waking up from S3 state the IMR boot will fail as the content is lost.
Set the skip_imr_boot flag to make sure that we don't try IMR in this case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://msgid.link/r/20240322112504.4192-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
During pause/reset or stop/start the LLP counter is not reset, which will
result broken delay reporting.
Read the LLP value on STOP/PAUSE trigger and use it in LLP reading to
normalize the LLP from the register.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-18-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch improves the delay calculation by relying on the
LLP (Linear Link Position) on the DAI side and the
LDP (Linear Data Pointer) on the host side. The LDP provides the same DMA
position as LPIB, but with a linear count instead of a position in the
ALSA ring buffer. The LDP values are provided in bytes and must be
converted to frames. The difference in units means that the host counter
will wrap earlier than the LLP. We need to wrap the LLP at the same
boundary as the host counter.
The ASoC framework relies on separate pointer and delay callback.
Measurement errors can be reduced by processing all the counter values in
the pointer callback. The delay value is stored, and will be reported to
higher levels in the delay callback.
For playback, the firmware provides a stream_start offset to handle
mixing/pause usages, where the DAI might have started earlier than the
PCM device. The delay calculation must be special-cased when the link
counter has not reached the start offset value, i.e. no valid audio has
left the DSP.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-16-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The IPC specific pointer callback can be used when additional or custom
handling is needed during the pointer calculation, like executing a delay
calculation at the same time to minimize drift between the reported pointer
and the calculated delay.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-15-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
When the final state is SOF_IPC4_PIPE_PAUSED, it is possible that the
stream will be restarted (resume or start) in which case we need to update
the offset from the firmware.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-14-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The SNDRV_PCM_TRIGGER_PAUSE_PUSH does not need to be a separate case, it
can be handled along with STOP and SUSPEND
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-13-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The sof_ipc4_timestamp_info is only used by ipc4-pcm.c internally, it
should not be in a generic header implying that it might be used elsewhere.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-12-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The get_stream_position has been replaced by get_dai_frame_counter and all
related code can be dropped form the core.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-11-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The get_stream_position has been replaced by get_dai_frame_counter, it
should not be set to allow it to be dropped from core code.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-10-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Switch to the new callback to retrieve the DAI (link) frame counter.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-9-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add implementation for reading the LDP (Linear DMA Position) to be used as
get_host_byte_counter().
The LDP is counting the number of bytes moved between the DSP and host
memory.
Set the get_dai_frame_counter to hda_dsp_get_stream_llp, which is counting
the frames on the link side of the DSP.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-8-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
For delay calculation we need two information:
Number of bytes transferred between the DSP and host memory (ALSA buffer)
Number of frames transferred between the DSP and external device
(link/codec/DMIC/etc).
The reason for the different units (bytes vs frames) on host and dai side
is that the format on the dai side is decided by the firmware and might
not be the same as on the host side, thus the expectation is that the
counter reflects the number of frames.
The kernel know the host side format and in there we have access to the
DMA position which is in bytes.
In a simplified way, the DSP caused delay is the difference between the
two counters.
The existing get_stream_position callback is defined to retrieve the frame
counter on the DAI side but it's name is too generic to be intuitive and
makes it hard to define a callback for the host side.
This patch introduces a new set of callbacks to replace the
get_stream_position and define the host side equivalent:
get_dai_frame_counter
get_host_byte_counter
Subsequent patches will remove the old callback.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Drop the MTL mtl_dsp_get_stream_hda_link_position() function and related
defines since it can only work on platforms which have 19 streams because
of the use of 0x948 as base offset for the LLP registers.
The generic hda_dsp_get_stream_hda_link_position() takes the number of
streams into consideration when reading the LLP registers for the stream
and can handle different HDA configurations.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-6-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
When the Linear Link Position is not available in firmware SRAM window we
use the host accessible position registers to read it.
The address of the PPLCLLPL/U registers depend on the number of streams
(playback+capture).
At probe time the pplc_addr is calculated for each stream and we can use
it to read the LLP without the need of address re-calculation.
Set the get_stream_position callback in sof_hda_common_ops for all
platforms:
The callback is used for IPC4 delay calculations only but the register is
a generic HDA register, not tied to any specific IPC version.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-5-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
If the PCM have the dsp_max_burst_size_in_ms set then place a constraint
to limit the minimum buffer time to avoid xruns caused by DMA bursts
spinning on the ALSA buffer.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-4-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
When setting up the pcm widget, save the DSP buffer size (in ms) for
platform code to place a constraint on playback.
On playback the DMA will fill the buffer on start and if the period
size is smaller it will immediately overrun.
On capture the DMA will move data in 1ms bursts.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The dsp_max_burst_size_in_ms can be used to save the length of the maximum
burst size in ms the host DMA will use.
Platform code can place constraint using this to avoid user space
requesting too small ALSA buffer which will result xruns.
Cc: stable@vger.kernel.org # 6.8
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240321130814.4412-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Initialization is completed before adding the component as that can
start the process of the device binding and trigger actions that check
init_done.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier")
Message-ID: <20240325145510.328378-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The system and amplifier names influence which firmware and tuning files
are downloaded to the device; log these values to aid end-user system
support.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240325142937.257869-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull more sound fixes from Takashi Iwai:
"The remaining fixes for 6.9-rc1 that have been gathered in this week.
More about ASoC at this time (one long-standing fix for compress
offload, SOF, AMD ACP, Rockchip, Cirrus and tlv320 stuff) while
another regression fix in ALSA core and a couple of HD-audio quirks as
usual are included"
* tag 'sound-fix2-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: control: Fix unannotated kfree() cleanup
ALSA: hda/realtek: Add quirks for some Clevo laptops
ALSA: hda/realtek: Add quirk for HP Spectre x360 14 eu0000
ALSA: hda/realtek: fix the hp playback volume issue for LG machines
ASoC: soc-compress: Fix and add DPCM locking
ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED
ASoC: SOF: amd: Move signed_fw_image to struct acp_quirk_entry
ASoC: amd: yc: Revert "add new YC platform variant (0x63) support"
ASoC: amd: yc: Revert "Fix non-functional mic on Lenovo 21J2"
ASoC: soc-core.c: Skip dummy codec when adding platforms
ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates
ASoC: dt-bindings: cirrus,cs42l43: Fix 'gpio-ranges' schema
ASoC: amd: yc: Fix non-functional mic on ASUS M7600RE
ASoC: tlv320adc3xxx: Don't strip remove function when driver is builtin
|
|
The endpoint in NHLT table for a SSP port could have the device type
NHLT_DEVICE_BT or NHLT_DEVICE_I2S. Use intel_nhlt_ssp_device_type()
function to retrieve the device type before querying the endpoint
blob to make sure we are always using correct device type parameter.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20231127120657.19764-3-peter.ujfalusi@linux.intel.com>
|
|
Add a helper function intel_nhlt_ssp_device_type() to detect the type
of specific SSP port. The result is nhlt_device_type enum type which
could be NHLT_DEVICE_BT or NHLT_DEVICE_I2S.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20231127120657.19764-2-peter.ujfalusi@linux.intel.com>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/tty
Pull tty / serial driver updates from Greg KH:
"Here is the big set of TTY/Serial driver updates and cleanups for
6.9-rc1. Included in here are:
- more tty cleanups from Jiri
- loads of 8250 driver cleanups from Andy
- max310x driver updates
- samsung serial driver updates
- uart_prepare_sysrq_char() updates for many drivers
- platform driver remove callback void cleanups
- stm32 driver updates
- other small tty/serial driver updates
All of these have been in linux-next for a long time with no reported
issues"
* tag 'tty-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/tty: (199 commits)
dt-bindings: serial: stm32: add power-domains property
serial: 8250_dw: Replace ACPI device check by a quirk
serial: Lock console when calling into driver before registration
serial: 8250_uniphier: Switch to use uart_read_port_properties()
serial: 8250_tegra: Switch to use uart_read_port_properties()
serial: 8250_pxa: Switch to use uart_read_port_properties()
serial: 8250_omap: Switch to use uart_read_port_properties()
serial: 8250_of: Switch to use uart_read_port_properties()
serial: 8250_lpc18xx: Switch to use uart_read_port_properties()
serial: 8250_ingenic: Switch to use uart_read_port_properties()
serial: 8250_dw: Switch to use uart_read_port_properties()
serial: 8250_bcm7271: Switch to use uart_read_port_properties()
serial: 8250_bcm2835aux: Switch to use uart_read_port_properties()
serial: 8250_aspeed_vuart: Switch to use uart_read_port_properties()
serial: port: Introduce a common helper to read properties
serial: core: Add UPIO_UNKNOWN constant for unknown port type
serial: core: Move struct uart_port::quirks closer to possible values
serial: sh-sci: Call sci_serial_{in,out}() directly
serial: core: only stop transmit when HW fifo is empty
serial: pch: Use uart_prepare_sysrq_char().
...
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.9
A bunch of fixes that came in during the merge window, probably the most
substantial thing is the DPCM locking fix for compressed audio which has
been lurking for a while.
|
|
The recent conversion to the automatic kfree() forgot to mark a
variable with __free(kfree), leading to memory leaks. Fix it.
Fixes: 1052d9882269 ("ALSA: control: Use automatic cleanup of kfree()")
Reported-by: Mirsad Todorovac <mirsad.todorovac@alu.unizg.hr>
Closes: https://lore.kernel.org/r/c1e2ef3c-164f-4840-9b1c-f7ca07ca422a@alu.unizg.hr
Message-ID: <20240320062722.31325-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add audio quirks to fix speaker output and headset detection on some new
Clevo models:
- L240TU (ALC245)
- PE60SNE-G (ALC1220)
- V350SNEQ (ALC245)
Co-authored-by: Jeremy Soller <jeremy@system76.com>
Signed-off-by: Tim Crawford <tcrawford@system76.com>
Message-ID: <20240319212726.62888-1-tcrawford@system76.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Cirrus amps support for this laptop was added in patch:
33e5e648e631 ("ALSA: hda: cs35l41: Support additional HP Envy Models")
This patch adds fixes for wrong pincfgs, wrong DAC selection and
mute/micmute LEDs.
Signed-off-by: Anthony I Gilea <i@cpp.in>
Message-ID: <e2a7aaed-e9d7-4d36-8abf-b71dfd32a0ff@cpp.in>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Two regression fixes that had been introduced in this merge window,
additional HD-audio quirks, and a further enhancement for the new
kunit"
* tag 'sound-fix-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: core: add kunitconfig
ALSA: hda/realtek: add in quirk for Acer Swift Go 16 - SFG16-71
Revert "ALSA: usb-audio: Name feature ctl using output if input is PCM"
ALSA: timer: Fix missing irq-disable at closing
ALSA: hda/realtek: Add quirk for Lenovo Yoga 9 14IMH9
|
|
Recently we tested the headphone playback on 2 LG machines, if we set
the volume to the max value or near to the max value, the sound is too
loud, it could even bring harm to listeners.
A workaround is to decrease the max volume to a reasonable value for
the headphone's amplifier, then the users couldn't set the volume
bigger than that value from the userspace.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Message-ID: <20240318011128.156023-1-hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We find mising DPCM locking inside soc_compr_set_params_fe
before calling dpcm_be_dai_hw_params() and dpcm_be_dai_prepare()
which cause lockdep assert for DPCM lock not held in
__soc_pcm_hw_params() and __soc_pcm_prepare()
Signed-off-by: Shalini Manjunatha <quic_c_shalma@quicinc.com>
Link: https://msgid.link/r/d985beeafdd32316eb45f20811eb7926da7a796e.1709720380.git.quic_c_shalma@quicinc.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
It is helpful to add .kunitconfig if we work with the tools provided by
KUnit project. The file describes the series of kernel configurations to
satisfy the dependency to build the target test.
For example:
$ ./tools/testing/kunit/kunit.py run --arch=arm64 --cross_compile=aarch64-linux-gnu- --kunitconfig=sound/core/
[11:35:13] Configuring KUnit Kernel ...
Regenerating .config ...
Populating config with:
$ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu-
[11:35:19] Building KUnit Kernel ...
Populating config with:
$ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu-
Building with:
$ make ARCH=arm64 O=.kunit --jobs=8 CROSS_COMPILE=aarch64-linux-gnu-
[11:37:35] Starting KUnit Kernel (1/1)...
[11:37:35] ============================================================
Running tests with:
$ qemu-system-aarch64 -nodefaults -m 1024 -kernel .kunit/arch/arm64/boot/Image.gz -append 'kunit.enable=1 console=ttyAMA0 kunit_shutdown=reboot' -no-reboot -nographic -serial stdio -machine virt -cpu max,pauth-impdef=on
[11:37:35] ============== sound-core-test (10 subtests) ===============
[11:37:35] [PASSED] test_phys_format_size
[11:37:35] [PASSED] test_format_width
[11:37:35] [PASSED] test_format_endianness
[11:37:35] [PASSED] test_format_signed
[11:37:35] [PASSED] test_format_fill_silence
[11:37:35] [PASSED] test_playback_avail
[11:37:35] [PASSED] test_capture_avail
[11:37:35] [PASSED] test_card_set_id
[11:37:35] [PASSED] test_pcm_format_name
[11:37:35] [PASSED] test_card_add_component
[11:37:35] ================= [PASSED] sound-core-test =================
[11:37:35] ============================================================
[11:37:35] Testing complete. Ran 10 tests: passed: 10
[11:37:35] Elapsed time: 142.333s total, 5.617s configuring, 136.047s building, 0.630s running
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Message-ID: <20240317024050.588370-1-o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Keyboard has an LED that is ON/OFF when mic is muted/active
- LED is controlled by GPIO pin
- Patch enables led to appear in /sys/class/leds/ as hda::micmute
- Enables LED when mic is MUTED
- Disables LED when mic is active
[ fixed white spaces by tiwai ]
Signed-off-by: Ian Murphy <iano200@gmail.com>
Message-ID: <20240316094157.13890-1-iano200@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This reverts commit 1601cd53c7e3197181277326dbfc131d20a74e46.
This fix is applied globally to all devices, and it may change the
existing control names. When the devices are managed with the fixed
configuration like UCM, such control name mismatch may lead to
significant regressions.
For avoiding that kind of regression, we would need to apply such
changes conditionally, but it'd take time to settle down.
While the original fix is a good thing in general, in order to address
the regression, let's revert the change for now.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218605
Reported-and-tested-by: Niklāvs Koļesņikovs <pinkflames.linux@gmail.com>
Message-ID: <20240316083744.28126-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/powerpc/linux
Pull powerpc updates from Michael Ellerman:
- Add AT_HWCAP3 and AT_HWCAP4 aux vector entries for future use
by glibc
- Add support for recognising the Power11 architected and raw PVRs
- Add support for nr_cpus=n on the command line where the
boot CPU is >= n
- Add ppcxx_allmodconfig targets for all 32-bit sub-arches
- Other small features, cleanups and fixes
Thanks to Akanksha J N, Brian King, Christophe Leroy, Dawei Li, Geoff
Levand, Greg Kroah-Hartman, Jan-Benedict Glaw, Kajol Jain, Kunwu Chan,
Li zeming, Madhavan Srinivasan, Masahiro Yamada, Nathan Chancellor,
Nicholas Piggin, Peter Bergner, Qiheng Lin, Randy Dunlap, Ricardo B.
Marliere, Rob Herring, Sathvika Vasireddy, Shrikanth Hegde, Uwe
Kleine-König, Vaibhav Jain, and Wen Xiong.
* tag 'powerpc-6.9-1' of git://git.kernel.org/pub/scm/linux/kernel/git/powerpc/linux: (71 commits)
powerpc/macio: Make remove callback of macio driver void returned
powerpc/83xx: Fix build failure with FPU=n
powerpc/64s: Fix get_hugepd_cache_index() build failure
powerpc/4xx: Fix warp_gpio_leds build failure
powerpc/amigaone: Make several functions static
powerpc/embedded6xx: Fix no previous prototype for avr_uart_send() etc.
macintosh/adb: make adb_dev_class constant
powerpc: xor_vmx: Add '-mhard-float' to CFLAGS
powerpc/fsl: Fix mfpmr() asm constraint error
powerpc: Remove cpu-as-y completely
powerpc/fsl: Modernise mt/mfpmr
powerpc/fsl: Fix mfpmr build errors with newer binutils
powerpc/64s: Use .machine power4 around dcbt
powerpc/64s: Move dcbt/dcbtst sequence into a macro
powerpc/mm: Code cleanup for __hash_page_thp
powerpc/hv-gpci: Fix the H_GET_PERF_COUNTER_INFO hcall return value checks
powerpc/irq: Allow softirq to hardirq stack transition
powerpc: Stop using of_root
powerpc/machdep: Define 'compatibles' property in ppc_md and use it
of: Reimplement of_machine_is_compatible() using of_machine_compatible_match()
...
|
|
Merge series from Cristian Ciocaltea <cristian.ciocaltea@collabora.com>:
This patch series restores audio support on Valve's Steam Deck OLED model, which
broke after the recent introduction of ACP/PSP communication for IRAM/DRAM fence
register programming.
|
|
The recent introduction of the ACP/PSP communication for IRAM/DRAM fence
register modification breaks the audio support on Valve's Steam Deck
OLED device.
It causes IPC timeout errors when trying to load DSP topology during
probing:
1707255557.688176 kernel: snd_sof_amd_vangogh 0000:04:00.5: ipc tx timed out for 0x30100000 (msg/reply size: 48/0)
1707255557.689035 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ IPC dump start ]------------
1707255557.689421 kernel: snd_sof_amd_vangogh 0000:04:00.5: dsp_msg = 0x0 dsp_ack = 0x91d14f6f host_msg = 0x1 host_ack = 0xead0f1a4 irq_stat >
1707255557.689730 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ IPC dump end ]------------
1707255557.690074 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ DSP dump start ]------------
1707255557.690376 kernel: snd_sof_amd_vangogh 0000:04:00.5: IPC timeout
1707255557.690744 kernel: snd_sof_amd_vangogh 0000:04:00.5: fw_state: SOF_FW_BOOT_COMPLETE (7)
1707255557.691037 kernel: snd_sof_amd_vangogh 0000:04:00.5: invalid header size 0xdb43fe7. FW oops is bogus
1707255557.694824 kernel: snd_sof_amd_vangogh 0000:04:00.5: unexpected fault 0x6942d3b3 trace 0x6942d3b3
1707255557.695392 kernel: snd_sof_amd_vangogh 0000:04:00.5: ------------[ DSP dump end ]------------
1707255557.695755 kernel: snd_sof_amd_vangogh 0000:04:00.5: Failed to setup widget PIPELINE.6.ACPHS1.IN
1707255557.696069 kernel: snd_sof_amd_vangogh 0000:04:00.5: error: tplg component load failed -110
1707255557.696374 kernel: snd_sof_amd_vangogh 0000:04:00.5: error: failed to load DSP topology -22
1707255557.697904 kernel: snd_sof_amd_vangogh 0000:04:00.5: ASoC: error at snd_soc_component_probe on 0000:04:00.5: -22
1707255557.698405 kernel: sof_mach nau8821-max: ASoC: failed to instantiate card -22
1707255557.701061 kernel: sof_mach nau8821-max: error -EINVAL: Failed to register card(sof-nau8821-max)
1707255557.701624 kernel: sof_mach: probe of nau8821-max failed with error -22
Introduce a new member skip_iram_dram_size_mod to struct acp_quirk_entry and
use it to skip IRAM/DRAM size modification for Vangogh Galileo device.
Fixes: 55d7bbe43346 ("ASoC: SOF: amd: Add acp-psp mailbox interface for iram-dram fence register modification")
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://msgid.link/r/20240220201623.438944-3-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The signed_fw_image member of struct sof_amd_acp_desc is used to enable
signed firmware support in the driver via the acp_sof_quirk_table.
In preparation to support additional use cases of the quirk table (i.e.
adding new flags), move signed_fw_image to a new struct acp_quirk_entry
and update all references to it accordingly.
No functional changes intended.
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Link: https://msgid.link/r/20240220201623.438944-2-cristian.ciocaltea@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The conversion to guard macro dropped the irq-disablement at closing
mistakenly, which may lead to a race. Fix it.
Fixes: beb45974dd49 ("ALSA: timer: Use guard() for locking")
Reported-by: syzbot+28c1a5a5b041a754b947@syzkaller.appspotmail.com
Closes: http://lore.kernel.org/r/0000000000000b9a510613b0145f@google.com
Message-ID: <20240315101447.18395-1-tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The speakers on the Lenovo Yoga 9 14IMH9 are similar to previous generations
such as the 14IAP7, and the bass speakers can be fixed using similar methods
with one caveat: 14IMH9 uses CS35L41 amplifiers which need to be activated
separately.
Signed-off-by: Jichi Zhang <i@jichi.ca>
Message-ID: <20240315081954.45470-3-i@jichi.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This was a relatively calm development cycle. Most of changes are
rather small device-specific fixes and enhancements. The only
significant changes in ALSA core are code refactoring with the recent
cleanup infrastructure, which should bring no functionality changes.
Some highlights below:
Core:
- Lots of cleanups in ALSA core code with automatic kfree cleanup and
locking guard macros
- New ALSA core kunit test
ASoC:
- SoundWire support for AMD ACP 6.3 systems
- Support for reporting version information for AVS firmware
- Support DSPless mode for Intel Soundwire systems
- Support for configuring CS35L56 amplifiers using EFI calibration
data
- Log which component is being operated on as part of power
management trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
HD- and USB-audio:
- More Cirrus HD-audio codec support
- TAS2781 HD-audio codec fixes
- Scarlett2 mixer fixes
Others:
- Enhancement of virtio driver for audio control supports
- Cleanups of legacy PM code with new macros
- Firewire sound updates"
* tag 'sound-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (307 commits)
ALSA: usb-audio: Stop parsing channels bits when all channels are found.
ALSA: hda/tas2781: remove unnecessary runtime_pm calls
ALSA: hda/realtek - ALC236 fix volume mute & mic mute LED on some HP models
ALSA: aaci: Delete unused variable in aaci_do_suspend
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range again
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range
ALSA: scarlett2: Fix Scarlett 4th Gen autogain status values
ALSA: scarlett2: Fix Scarlett 4th Gen 4i4 low-voltage detection
ALSA: hda/tas2781: restore power state after system_resume
ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend
ALSA: hda/tas2781: do not reset cur_* values in runtime_suspend
ALSA: hda/tas2781: add lock to system_suspend
ALSA: hda/tas2781: use dev_dbg in system_resume
ALSA: hda/realtek: fix ALC285 issues on HP Envy x360 laptops
platform/x86: serial-multi-instantiate: Add support for CS35L54 and CS35L57
ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57
ASoC: cs35l56: Add support for CS35L54 and CS35L57
ASoC: Intel: catpt: Carefully use PCI bitwise constants
ALSA: hda: hda_component: Include sound/hda_codec.h
ALSA: hda: hda_component: Add missing #include guards
...
|
|
This reverts commit 316a784839b21b122e1761cdca54677bb19a47fa,
that enabled Yellow Carp (YC) driver for PCI revision id 0x63.
Mukunda Vijendar [1] points out that revision 0x63 is Pink
Sardine platform, not Yellow Carp. The YC driver should not
be enabled for this platform. This patch prevents the YC
driver from being incorrectly enabled.
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://msgid.link/r/20240313015853.3573242-3-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This reverts commit ed00a6945dc32462c2d3744a3518d2316da66fcc,
which added a quirk entry to enable the Yellow Carp (YC)
driver for the Lenovo 21J2 laptop.
Although the microphone functioned with the YC driver, it
resulted in incorrect driver usage. The Lenovo 21J2 is not a
Yellow Carp platform, but a Pink Sardine platform, which
already has an upstreamed driver.
The microphone on the Lenovo 21J2 operates correctly with the
CONFIG_SND_SOC_AMD_PS flag enabled and does not require the
quirk entry. So this patch removes the quirk entry.
Thanks to Mukunda Vijendar [1] for pointing this out.
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Signed-off-by: Jiawei Wang <me@jwang.link>
Link: https://lore.kernel.org/linux-sound/023092e1-689c-4b00-b93f-4092c3724fb6@amd.com/ [1]
Link: https://msgid.link/r/20240313015853.3573242-2-me@jwang.link
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Merge series from Luca Ceresoli <luca.ceresoli@bootlin.com>:
This series adds a driver for the internal audio codec of the Rockchip
RK3308 SoC, along with some related patches. This codec is internally
connected to the I2S peripherals on the same chip, and it has some
peculiarities arising from that interconnection.
For proper bidirectional operation with the internal codec at any possible
combination of sampling rates, the I2S peripheral needs two clock sources
(tx and rx), while connection with an external codec commonly needs only
one.
Since v5.16 there is a driver for the I2S in
sound/soc/rockchip/rockchip_i2s_tdm.c, but in some cases it does not
configure correctly the clocks, resulting in an unnecessarily inaccurate
rate. Patch 1 fixes this.
Patches 2-4 add the codec driver along with the bindings and a new helper
macro.
Patches 5-7 add to the SoC DT file two I2S controllers (those which are
internally connected to the internal codec) and the codec itself and enable
the driver in the ARM64 defconfig.
Luca
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
---
Changes in v4:
- several cleanups in the codec probe function
- Link to v3: https://lore.kernel.org/r/20240221-rk3308-audio-codec-v3-0-dfa34abfcef6@bootlin.com
Changes in v3:
- Add the I2S clock fix patch and remove a previous fix which is now superseded
- Codec driver: fix silent playback until a given amplitude of sigital
value, seen at >= 96 kHz rate
- various other changes, listed per-patch
- Link to v2: https://lore.kernel.org/r/20231219-rk3308-audio-codec-v2-0-c70d06021946@bootlin.com
Changes in v2:
- largely rewrote the codec driver to use DAPM and lots of improvements
and cleanups
- removed the RK3308 audio card and related patches
- various other changes, listed per-patch
- Link to v1: https://lore.kernel.org/all/20220907142124.2532620-1-luca.ceresoli@bootlin.com/
---
Luca Ceresoli (7):
ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates
ASoC: dt-bindings: Add Rockchip RK3308 internal audio codec
ASoC: core: add SOC_DOUBLE_RANGE_TLV() helper macro
ASoC: codecs: Add RK3308 internal audio codec driver
arm64: defconfig: enable Rockchip RK3308 internal audio codec driver
arm64: dts: rockchip: add i2s_8ch_2 and i2s_8ch_3
arm64: dts: rockchip: add the internal audio codec
.../bindings/sound/rockchip,rk3308-codec.yaml | 98 +++
MAINTAINERS | 7 +
arch/arm64/boot/dts/rockchip/rk3308.dtsi | 56 ++
arch/arm64/configs/defconfig | 1 +
include/sound/soc.h | 12 +
sound/soc/codecs/Kconfig | 11 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/rk3308_codec.c | 974 +++++++++++++++++++++
sound/soc/codecs/rk3308_codec.h | 579 ++++++++++++
sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +-------
10 files changed, 1746 insertions(+), 346 deletions(-)
---
base-commit: dfda120c512b3edca1436f770924e91b14f93a98
change-id: 20231219-rk3308-audio-codec-a5558ba8949d
Best regards,
--
Luca Ceresoli <luca.ceresoli@bootlin.com>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm
Pull power management updates from Rafael Wysocki:
"From the functional perspective, the most significant change here is
the addition of support for Energy Models that can be updated
dynamically at run time.
There is also the addition of LZ4 compression support for hibernation,
the new preferred core support in amd-pstate, new platforms support in
the Intel RAPL driver, new model-specific EPP handling in intel_pstate
and more.
Apart from that, the cpufreq default transition delay is reduced from
10 ms to 2 ms (along with some related adjustments), the system
suspend statistics code undergoes a significant rework and there is a
usual bunch of fixes and code cleanups all over.
Specifics:
- Allow the Energy Model to be updated dynamically (Lukasz Luba)
- Add support for LZ4 compression algorithm to the hibernation image
creation and loading code (Nikhil V)
- Fix and clean up system suspend statistics collection (Rafael
Wysocki)
- Simplify device suspend and resume handling in the power management
core code (Rafael Wysocki)
- Fix PCI hibernation support description (Yiwei Lin)
- Make hibernation take set_memory_ro() return values into account as
appropriate (Christophe Leroy)
- Set mem_sleep_current during kernel command line setup to avoid an
ordering issue with handling it (Maulik Shah)
- Fix wake IRQs handling when pm_runtime_force_suspend() is used as a
driver's system suspend callback (Qingliang Li)
- Simplify pm_runtime_get_if_active() usage and add a replacement for
pm_runtime_put_autosuspend() (Sakari Ailus)
- Add a tracepoint for runtime_status changes tracking (Vilas Bhat)
- Fix section title markdown in the runtime PM documentation (Yiwei
Lin)
- Enable preferred core support in the amd-pstate cpufreq driver
(Meng Li)
- Fix min_perf assignment in amd_pstate_adjust_perf() and make the
min/max limit perf values in amd-pstate always stay within the
(highest perf, lowest perf) range (Tor Vic, Meng Li)
- Allow intel_pstate to assign model-specific values to strings used
in the EPP sysfs interface and make it do so on Meteor Lake
(Srinivas Pandruvada)
- Drop long-unused cpudata::prev_cummulative_iowait from the
intel_pstate cpufreq driver (Jiri Slaby)
- Prevent scaling_cur_freq from exceeding scaling_max_freq when the
latter is an inefficient frequency (Shivnandan Kumar)
- Change default transition delay in cpufreq to 2ms (Qais Yousef)
- Remove references to 10ms minimum sampling rate from comments in
the cpufreq code (Pierre Gondois)
- Honour transition_latency over transition_delay_us in cpufreq (Qais
Yousef)
- Stop unregistering cpufreq cooling on CPU hot-remove (Viresh Kumar)
- General enhancements / cleanups to ARM cpufreq drivers (tianyu2,
Nícolas F. R. A. Prado, Erick Archer, Arnd Bergmann, Anastasia
Belova)
- Update cpufreq-dt-platdev to block/approve devices (Richard Acayan)
- Make the SCMI cpufreq driver get a transition delay value from
firmware (Pierre Gondois)
- Prevent the haltpoll cpuidle governor from shrinking guest
poll_limit_ns below grow_start (Parshuram Sangle)
- Avoid potential overflow in integer multiplication when computing
cpuidle state parameters (C Cheng)
- Adjust MWAIT hint target C-state computation in the ACPI cpuidle
driver and in intel_idle to return a correct value for C0 (He
Rongguang)
- Address multiple issues in the TPMI RAPL driver and add support for
new platforms (Lunar Lake-M, Arrow Lake) to Intel RAPL (Zhang Rui)
- Fix freq_qos_add_request() return value check in dtpm_cpu (Daniel
Lezcano)
- Fix kernel-doc for dtpm_create_hierarchy() (Yang Li)
- Fix file leak in get_pkg_num() in x86_energy_perf_policy (Samasth
Norway Ananda)
- Fix cpupower-frequency-info.1 man page typo (Jan Kratochvil)
- Fix a couple of warnings in the OPP core code related to W=1 builds
(Viresh Kumar)
- Move dev_pm_opp_{init|free}_cpufreq_table() to pm_opp.h (Viresh
Kumar)
- Extend dev_pm_opp_data with turbo support (Sibi Sankar)
- dt-bindings: drop maxItems from inner items (David Heidelberg)"
* tag 'pm-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm: (95 commits)
dt-bindings: opp: drop maxItems from inner items
OPP: debugfs: Fix warning around icc_get_name()
OPP: debugfs: Fix warning with W=1 builds
cpufreq: Move dev_pm_opp_{init|free}_cpufreq_table() to pm_opp.h
OPP: Extend dev_pm_opp_data with turbo support
Fix cpupower-frequency-info.1 man page typo
cpufreq: scmi: Set transition_delay_us
firmware: arm_scmi: Populate fast channel rate_limit
firmware: arm_scmi: Populate perf commands rate_limit
cpuidle: ACPI/intel: fix MWAIT hint target C-state computation
PM: sleep: wakeirq: fix wake irq warning in system suspend
powercap: dtpm: Fix kernel-doc for dtpm_create_hierarchy() function
cpufreq: Don't unregister cpufreq cooling on CPU hotplug
PM: suspend: Set mem_sleep_current during kernel command line setup
cpufreq: Honour transition_latency over transition_delay_us
cpufreq: Limit resolving a frequency to policy min/max
Documentation: PM: Fix runtime_pm.rst markdown syntax
cpufreq: amd-pstate: adjust min/max limit perf
cpufreq: Remove references to 10ms min sampling rate
cpufreq: intel_pstate: Update default EPPs for Meteor Lake
...
|
|
In order to apply additional fixes that depend on the fixes merged for
v6.8 merge up the final release.
|
|
If a usb audio device sets more bits than the amount of channels
it could write outside of the map array.
Signed-off-by: Johan Carlsson <johan.carlsson@teenage.engineering>
Fixes: 04324ccc75f9 ("ALSA: usb-audio: add channel map support")
Message-ID: <20240313081509.9801-1-johan.carlsson@teenage.engineering>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The runtime_pm handling seems to have been loosely inspired by the
cs32l41 driver, but in this case the get_noresume/put sequence is not
required.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Message-ID: <20240312161217.79510-1-pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Some HP laptops have received revisions that altered their board IDs
and therefore the current patches/quirks do not apply to them.
Specifically, for my Probook 440 G8, I have a board ID of 8a74.
It is necessary to add a line for that specific model.
Signed-off-by: Valentine Altair <faetalize@proton.me>
Cc: <stable@vger.kernel.org>
Message-ID: <kOqXRBcxkKt6m5kciSDCkGqMORZi_HB3ZVPTX5sD3W1pKxt83Pf-WiQ1V1pgKKI8pYr4oGvsujt3vk2zsCE-DDtnUADFG6NGBlS5N3U4xgA=@proton.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
When pcm_runtime is adding platform components it will scan all
registered components. In case of DPCM FE/BE some DAI links will
configure dummy platform. However both dummy codec and dummy platform
are using "snd-soc-dummy" as component->name. Dummy codec should be
skipped when adding platforms otherwise there'll be overflow and UBSAN
complains.
Reported-by: Zhipeng Wang <zhipeng.wang_1@nxp.com>
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240305065606.3778642-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The sample rates set by the rockchip_i2s_tdm driver in master mode are
inaccurate up to 5% in several cases, due to the driver logic to configure
clocks and a nasty interaction with the Common Clock Framework.
To understand what happens, here is the relevant section of the clock tree
(slightly simplified), along with the names used in the driver:
vpll0 _OR_ vpll1 "mclk_root"
clk_i2s2_8ch_tx_src "mclk_parent"
clk_i2s2_8ch_tx_mux
clk_i2s2_8ch_tx "mclk" or "mclk_tx"
This is what happens when playing back e.g. at 192 kHz using
audio-graph-card (when recording the same applies, only s/tx/rx/):
0. at probe, rockchip_i2s_tdm_set_sysclk() stores the passed frequency in
i2s_tdm->mclk_tx_freq (*) which is 50176000, and that is never modified
afterwards
1. when playback is started, rockchip_i2s_tdm_hw_params() is called and
does the following two calls
2. rockchip_i2s_tdm_calibrate_mclk():
2a. selects mclk_root0 (vpll0) as a parent for mclk_parent
(mclk_tx_src), which is OK because the vpll0 rate is a good for
192000 (and sumbultiple) rates
2b. sets the mclk_root frequency based on ppm calibration computations
2c. sets mclk_tx_src to 49152000 (= 256 * 192000), which is also OK as
it is a multiple of the required bit clock
3. rockchip_i2s_tdm_set_mclk()
3a. calls clk_set_rate() to set the rate of mclk_tx (clk_i2s2_8ch_tx)
to the value of i2s_tdm->mclk_tx_freq (*), i.e. 50176000 which is
not a multiple of the sampling frequency -- this is not OK
3a1. clk_set_rate() reacts by reparenting clk_i2s2_8ch_tx_src to
vpll1 -- this is not OK because the default vpll1 rate can be
divided to get 44.1 kHz and related rates, not 192 kHz
The result is that the driver does a lot of ad-hoc decisions about clocks
and ends up in using the wrong parent at an unoptimal rate.
Step 0 is one part of the problem: unless the card driver calls set_sysclk
at each stream start, whatever rate is set in mclk_tx_freq during boot will
be taken and used until reboot. Moreover the driver does not care if its
value is not a multiple of any audio frequency.
Another part of the problem is that the whole reparenting and clock rate
setting logic is conflicting with the CCF algorithms to achieve largely the
same goal: selecting the best parent and setting the closest clock
rate. And it turns out that only calling once clk_set_rate() on
clk_i2s2_8ch_tx picks the correct vpll and sets the correct rate.
The fix is based on removing the custom logic in the driver to select the
parent and set the various clocks, and just let the Clock Framework do it
all. As a side effect, the set_sysclk() op becomes useless because we now
let the CCF compute the appropriate value for the sampling rate. It also
implies that the whole calibration logic is now dead code and so it is
removed along with the "PCM Clock Compensation in PPM" kcontrol, which has
always been broken anyway. The handling of the 4 optional clocks also
becomes dead code and is removed.
The actual rates have been tested playing 30 seconds of audio at various
sampling rates before and after this change using sox:
time play -r <sample_rate> -n synth 30 sine 950 gain -3
The time reported in the table below is the 'real' value reported by the
'time' command in the above command line.
rate before after
--------- ------ ------
8000 Hz 30.60s 30.63s
11025 Hz 30.45s 30.51s
16000 Hz 30.47s 30.50s
22050 Hz 30.78s 30.41s
32000 Hz 31.02s 30.43s
44100 Hz 30.78s 30.41s
48000 Hz 29.81s 30.45s
88200 Hz 30.78s 30.41s
96000 Hz 29.79s 30.42s
176400 Hz 27.40s 30.41s
192000 Hz 29.79s 30.42s
While the tests are running the clock tree confirms that:
* without the patch, vpll1 is always used and clk_i2s2_8ch_tx always
produces 50176000 Hz, which cannot be divided for most audio rates
except the slowest ones, generating inaccurate rates
* with the patch:
- for 192000 Hz vpll0 is used
- for 176400 Hz vpll1 is used
- clk_i2s2_8ch_tx always produces (256 * <rate>) Hz
Tested on the RK3308 using the internal audio codec.
Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller")
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240305-rk3308-audio-codec-v4-1-312acdbe628f@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The variable aaci is not used anymore and can be deleted.
Fixes: 792a6c51875c ("[ALSA] Fix PM support")
Signed-off-by: Thomas Weißschuh <thomas.weissschuh@linutronix.de>
Link: https://lore.kernel.org/r/20240312-aaci-unused-v1-1-09be643f67c2@linutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The ASUS M7600RE (Vivobook Pro 16X OLED) needs a quirks-table entry for the
internal microphone to function properly.
Signed-off-by: Mitch Cooley <m.cooley.198@gmail.com>
Link: https://msgid.link/r/CALijGznExWW4fujNWwMzmn_K=wo96sGzV_2VkT7NjvEUdkg7Gw@mail.gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v6.9
This has been quite a small release, there's a lot of driver specific
cleanups and minor enhancements but hardly anything on the core and only
one new driver. Highlights include:
- SoundWire support for AMD ACP 6.3 systems.
- Support for reporting version information for AVS firmware.
- Support DSPless mode for Intel Soundwire systems.
- Support for configuring CS35L56 amplifiers using EFI calibration
data.
- Log which component is being operated on as part of power management
trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
|
|
Merge changes related to the runtime power management of devices for
6.9-rc1:
- Simplify pm_runtime_get_if_active() usage and add a replacement for
pm_runtime_put_autosuspend() (Sakari Ailus).
- Add a tracepoint for runtime_status changes tracking (Vilas Bhat).
- Fix section title markdown in the runtime PM documentation (Yiwei
Lin).
* pm-runtime:
Documentation: PM: Fix runtime_pm.rst markdown syntax
PM: runtime: add tracepoint for runtime_status changes
PM: runtime: Add pm_runtime_put_autosuspend() replacement
PM: runtime: Simplify pm_runtime_get_if_active() usage
|
|
Using __exit for the remove function results in the remove callback
being discarded with SND_SOC_TLV320ADC3XXX=y. When such a device gets
unbound (e.g. using sysfs or hotplug), the driver is just removed
without the cleanup being performed. This results in resource leaks. Fix
it by compiling in the remove callback unconditionally.
This also fixes a W=1 modpost warning:
WARNING: modpost: sound/soc/codecs/snd-soc-tlv320adc3xxx: section mismatch in reference: adc3xxx_i2c_driver+0x10 (section: .data) -> adc3xxx_i2c_remove (section: .exit.text)
(which only happens with SND_SOC_TLV320ADC3XXX=m).
Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver")
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://msgid.link/r/20240310143852.397212-2-u.kleine-koenig@pengutronix.de
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The 4th Gen input preamp gain range is 0dB to +69dB, although the
control values range from 0 to 70. Replace SCARLETT2_MAX_GAIN with
SCARLETT2_MAX_GAIN_VALUE and SCARLETT2_MAX_GAIN_DB, and update the TLV
again.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: a45cf0a08347 ("ALSA: scarlett2: Fix Scarlett 4th Gen input gain range")
Message-ID: <Ze7OMA8ntG7KteGa@m.b4.vu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The input gain range TLV was declared as -70dB to 0dB, but the preamp
gain range is actually 0dB to +70dB. Rename SCARLETT2_GAIN_BIAS to
SCARLETT2_MAX_GAIN and update the TLV to fix.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <9168317b5ac5335943d3f14dbcd1cc2d9b2299d0.1710047969.git.g@b4.vu>
|
|
The meanings of the raw_auto_gain_status values were originally
guessed through experimentation, but the official names have now been
discovered. Update the autogain status control strings accordingly.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 0a995e38dc44 ("ALSA: scarlett2: Add support for software-controllable input gain")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <8bd12a5e7dc714801dd9887c4bc5cb35c384e27c.1710047969.git.g@b4.vu>
|
|
The value currently being read to determine the low-voltage state is
actually the front panel state. Fix the code to use the correct offset
for the low-voltage state.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: d7cfa2fdfc8a ("ALSA: scarlett2: Add power status control")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d97b7d87f43b0e54f37e1552394be2f3ae182704.1710047969.git.g@b4.vu>
|
|
After system_resume the amplifers will remain off, even if they were on
before system_suspend.
Use playback_started bool to save the playback state, and restore power
state based on it.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <1742b61901781826f6e6212ffe1d21af542d134a.1709918447.git.soyer@irl.hu>
|
|
The runtime_resume function calls prmg_load and apply_calibration
functions, but system_resume also calls them, so calling
pm_runtime_force_resume before reset is unnecessary.
For consistency, do not call the pm_runtime_force_suspend in
system_suspend, as runtime_suspend does the same.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <d0b4cc1248b9d375d59c009563da42d60d69eac3.1709918447.git.soyer@irl.hu>
|
|
The amplifier doesn't loose register state in software shutdown mode, so
there is no need to reset the cur_* values.
Without these resets, the amplifier can be turned on after
runtime_suspend without waiting for the program and
profile to be restored.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <aa27ae084150988bf6a0ead7e3403bc485d790f8.1709918447.git.soyer@irl.hu>
|
|
Add the missing lock around tasdevice_tuning_switch().
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <c666da13d4bc48cd1ab1357479e0c6096541372c.1709918447.git.soyer@irl.hu>
|
|
The system_resume function uses dev_info for tracing, but the other pm
functions use dev_dbg.
Use dev_dbg as the other pm functions.
Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver")
Signed-off-by: Gergo Koteles <soyer@irl.hu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <140f3c689c9eb5874e6eb48a570fcd8207f06a41.1709918447.git.soyer@irl.hu>
|
|
Realtek codec on HP Envy laptop series are heavily modified by vendor.
Therefore, need intervention to make it work properly. The patch fixes:
- B&O soundbar speakers (between lid and keyboard) activation
- Enable LED on mute button
- Add missing process coefficient which affects the output amplifier
- Volume control synchronization between B&O soundbar and side speakers
- Unmute headset output on several HP Envy models
- Auto-enable headset mic when plugged
This patch was tested on HP Envy x360 13-AR0107AU with Realtek ALC285
The only unsolved problem is output amplifier of all built-in speakers
is too weak, which causes volume of built-in speakers cannot be loud
as vendor's proprietary driver due to missing _DSD parameter in the
firmware. The solution is currently on research. Expected to has another
patch in the future.
Potential fix to related issues, need test before close those issues:
- https://bugzilla.kernel.org/show_bug.cgi?id=189331
- https://bugzilla.kernel.org/show_bug.cgi?id=216632
- https://bugzilla.kernel.org/show_bug.cgi?id=216311
- https://bugzilla.kernel.org/show_bug.cgi?id=213507
Signed-off-by: Athaariq Ardhiansyah <foss@athaariq.my.id>
Message-ID: <20240310140249.3695-1-foss@athaariq.my.id>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Prep for 6.9 merge.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add the HID for the CS35L54 and CS35L57 Boosted Smart Amplifiers. These
have the same control interface as the CS35L56 so are handled by the
cs35l56-hda driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-3-rf@opensource.cirrus.com>
|
|
The CS35L54 and CS35L57 are Boosted Smart Amplifiers. The CS35L54 has
I2C/SPI control and I2S/TDM audio. The CS35L57 also has SoundWire
control and audio.
The hardware differences between L54, L56 and L57 do not affect the
driver control interface so they can all be handled by the same driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240308135900.603192-2-rf@opensource.cirrus.com>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
Some more driver specific fixes for v6.8, plus one new x86 platform
quirk. All good fixes to have if you have systems that use the relevant
hardware.
|
|
PM constants for PCI devices are defined with bitwise annotation.
When used as is, sparse complains about that:
.../catpt/dsp.c:390:9: warning: restricted pci_power_t degrades to integer
.../catpt/dsp.c:414:9: warning: restricted pci_power_t degrades to integer
Force them to be u32 in the driver.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://msgid.link/r/20240307163734.3852754-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
hda_component.h uses hda_codec_dev from sound/hda_codec.h.
Include sound/hda_codec.h instead of assuming that it has already
been included by the parent .c file.
This isn't causing any problems with current code, so no need to
backport to older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240307111216.45053-2-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Add the conventional include guards around the content of the
hda_component.h header file. This prevents double-declaration of
struct hda_component if the header gets included multiple times.
This isn't causing any problems with current code, so no need to
backport to older kernels.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Message-ID: <20240307111216.45053-1-rf@opensource.cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
wm_adsp_write_ctl() must hold the pwr_lock mutex when calling
cs_dsp_get_ctl().
This was previously partially fixed by commit 781118bc2fc1
("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()")
but this only put locking around the call to cs_dsp_coeff_write_ctrl(),
missing the call to cs_dsp_get_ctl().
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 781118bc2fc1 ("ASoC: wm_adsp: Fix missing locking in wm_adsp_[read|write]_ctl()")
Link: https://msgid.link/r/20240307110227.41421-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We don't use mic1_src and mic2_src.so we delete these two members.
We changed the default value of interrupt-clk for headphone detection
Signed-off-by: Zhang Yi <zhangyi@everest-semi.com>
Link: https://msgid.link/r/20240307051222.24010-2-zhangyi@everest-semi.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Commit fc7a6209d571 ("bus: Make remove callback return void") forces
bus_type::remove be void-returned, it doesn't make much sense for any
bus based driver implementing remove callbalk to return non-void to
its caller.
This change is for macio bus based drivers.
Signed-off-by: Dawei Li <set_pte_at@outlook.com>
Acked-by: Damien Le Moal <damien.lemoal@opensource.wdc.com>
Acked-by: Jakub Kicinski <kuba@kernel.org>
Signed-off-by: Michael Ellerman <mpe@ellerman.id.au>
Link: https://msgid.link/TYCP286MB232391520CB471E7C8D6EA84CAD19@TYCP286MB2323.JPNP286.PROD.OUTLOOK.COM
|
|
Use source instead of ret, which seems to be unrelated and will always
be zero.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-5-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-2-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240306161439.1385643-1-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add the event value to the snd_soc_dapm_start and snd_soc_dapm_done trace
events to make them more informative.
Trace before:
aplay-229 [000] 250.140309: snd_soc_dapm_start: card=vscn-2046
aplay-229 [000] 250.167531: snd_soc_dapm_done: card=vscn-2046
aplay-229 [000] 251.169588: snd_soc_dapm_start: card=vscn-2046
aplay-229 [000] 251.195245: snd_soc_dapm_done: card=vscn-2046
Trace after:
aplay-214 [000] 693.290612: snd_soc_dapm_start: card=vscn-2046 event=1
aplay-214 [000] 693.315508: snd_soc_dapm_done: card=vscn-2046 event=1
aplay-214 [000] 694.537349: snd_soc_dapm_start: card=vscn-2046 event=2
aplay-214 [000] 694.563241: snd_soc_dapm_done: card=vscn-2046 event=2
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240306-improve-asoc-trace-events-v1-2-edb252bbeb10@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The snd_soc_bias_level_start and snd_soc_bias_level_done trace events
currently look like:
aplay-229 [000] 1250.140778: snd_soc_bias_level_start: card=vscn-2046 val=1
aplay-229 [000] 1250.140784: snd_soc_bias_level_done: card=vscn-2046 val=1
aplay-229 [000] 1250.140786: snd_soc_bias_level_start: card=vscn-2046 val=2
aplay-229 [000] 1250.140788: snd_soc_bias_level_done: card=vscn-2046 val=2
kworker/u8:1-21 [000] 1250.140871: snd_soc_bias_level_start: card=vscn-2046 val=1
kworker/u8:0-11 [000] 1250.140951: snd_soc_bias_level_start: card=vscn-2046 val=1
kworker/u8:0-11 [000] 1250.140956: snd_soc_bias_level_done: card=vscn-2046 val=1
kworker/u8:0-11 [000] 1250.140959: snd_soc_bias_level_start: card=vscn-2046 val=2
kworker/u8:0-11 [000] 1250.140961: snd_soc_bias_level_done: card=vscn-2046 val=2
kworker/u8:1-21 [000] 1250.167219: snd_soc_bias_level_done: card=vscn-2046 val=1
kworker/u8:1-21 [000] 1250.167222: snd_soc_bias_level_start: card=vscn-2046 val=2
kworker/u8:1-21 [000] 1250.167232: snd_soc_bias_level_done: card=vscn-2046 val=2
kworker/u8:0-11 [000] 1250.167440: snd_soc_bias_level_start: card=vscn-2046 val=3
kworker/u8:0-11 [000] 1250.167444: snd_soc_bias_level_done: card=vscn-2046 val=3
kworker/u8:1-21 [000] 1250.167497: snd_soc_bias_level_start: card=vscn-2046 val=3
kworker/u8:1-21 [000] 1250.167506: snd_soc_bias_level_done: card=vscn-2046 val=3
There are clearly multiple calls, one per component, but they cannot be
discriminated from each other.
Change the ftrace events to also print the component name, to make it clear
which part of the code is involved. This requires changing the passed value
from a struct snd_soc_card, where the DAPM context is not kwown, to a
struct snd_soc_dapm_context where it is obviously known but the a card
pointer is also available.
With this change, the resulting trace becomes:
aplay-247 [000] 1436.357332: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=1
aplay-247 [000] 1436.357338: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=1
aplay-247 [000] 1436.357340: snd_soc_bias_level_start: card=vscn-2046 component=(none) val=2
aplay-247 [000] 1436.357343: snd_soc_bias_level_done: card=vscn-2046 component=(none) val=2
kworker/u8:4-215 [000] 1436.357437: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=1
kworker/u8:5-231 [000] 1436.357518: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=1
kworker/u8:5-231 [000] 1436.357523: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=1
kworker/u8:5-231 [000] 1436.357526: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=2
kworker/u8:5-231 [000] 1436.357528: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=2
kworker/u8:4-215 [000] 1436.383217: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=1
kworker/u8:4-215 [000] 1436.383221: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=2
kworker/u8:4-215 [000] 1436.383231: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=2
kworker/u8:5-231 [000] 1436.383468: snd_soc_bias_level_start: card=vscn-2046 component=ff320000.i2s val=3
kworker/u8:5-231 [000] 1436.383472: snd_soc_bias_level_done: card=vscn-2046 component=ff320000.i2s val=3
kworker/u8:4-215 [000] 1436.383503: snd_soc_bias_level_start: card=vscn-2046 component=ff560000.codec val=3
kworker/u8:4-215 [000] 1436.383513: snd_soc_bias_level_done: card=vscn-2046 component=ff560000.codec val=3
Signed-off-by: Luca Ceresoli <luca.ceresoli@bootlin.com>
Link: https://msgid.link/r/20240306-improve-asoc-trace-events-v1-1-edb252bbeb10@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
intel-mid.h is providing some core parts of the South Complex PM,
which are usually are not used by individual drivers. In particular,
this driver doesn't use it, so simply remove the unused header.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Link: https://msgid.link/r/20240305160723.1363534-1-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The HP EliteBook 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/20240304134033.773348-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
ASoC machine driver can use snd_soc_{of_}get_dlc() (A) to get DAI name
for dlc (snd_soc_dai_link_component). In this function call
dlc->dai_name is parsed via snd_soc_dai_name_get() (B).
(A) int snd_soc_get_dlc(...)
{
...
(B) dlc->dai_name = snd_soc_dai_name_get(dai);
...
}
(B) has a priority to return dai->name as dlc->dai_name. In most cases
card can probe successfully. However it has an issue that ASoC tries to
rebind card. Here is a simplified flow for example:
| a) Card probes successfully at first
| b) One of the component bound to this card is removed for some
| reason the component->dev is released
| c) That component is re-registered
v d) ASoC calls snd_soc_try_rebind_card()
a) points dlc->dai_name to dai->name. b) releases all resource of the
old DAI. c) creates new DAI structure. In result d) can not use
dlc->dai_name to add new created DAI.
So it's reasonable that prefer to return dai->driver->name in
snd_soc_dai_name_get() because dai->driver is a pre-defined global
variable. Also update snd_soc_is_matching_dai() for alignment.
Signed-off-by: Chancel Liu <chancel.liu@nxp.com>
Link: https://msgid.link/r/20240304072128.2845432-1-chancel.liu@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Add a KUnit test for the cs-amp-lib library. This has test cases
for cs_amp_get_efi_calibration_data() and cs_amp_write_cal_coeffs().
A KUNIT_STATIC_STUB_REDIRECT() has been added to
cs_amp_get_efi_variable() and cs_amp_write_cal_coeff() so that the
KUnit test can redirect these to test harness functions.
Much of the testing involves invoking the same function with different
parameters, i.e. the number of amps and the amp index within the array.
This uses parameterization rather than looping. The idea is to avoid
looping over configurations within one test case as that has a higher
chance of having a bug that doesn't actually test all the expected cases.
Having the test run exactly one configuration, and then tear-down, is less
prone to accidentally skipped configurations.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://msgid.link/r/20240304143705.26362-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The HP Pavilion Aero Laptop 13-be2xxx(8BD6) requires a quirk entry for its internal microphone to function.
Signed-off-by: Al Raj Hassain <alrajhassain@gmail.com>
Reviewed-by: Mario Limonciello <mario.limonciello@amd.com>
Link: https://msgid.link/r/20240304103924.13673-1-alrajhassain@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Timing select registers for SRC and CMD are by default
referring to the corresponding SSI word select.
The calculation rule from HW spec skips SSI8, which has
no clock connection.
>From section 43.2.18 CMD Output Timing Select Register (CMDOUT_TIMSEL),
of R-Car Series, 3rd Generation Hardware User’s Manual Rev.2.20:
CMD0_OUT_DIVCLK_ Output Timing
SEL [4:0] Signal Select
B'0 0110: ssi_ws0
B'0 0111: ssi_ws1
B'0 1000: ssi_ws2
B'0 1001: ssi_ws3
B'0 1010: ssi_ws4
B'0 1011: ssi_ws5
B'0 1100: ssi_ws6
B'0 1101: ssi_ws7
<GAP>
B'0 1110: ssi_ws9
B'0 1111: Setting prohibited
Fix the erroneous prohibited setting of timsel value 1111 (0xf) for SSI9
by using timsel value 1110 (0xe) instead. This is possible because SSI8
is not connected as shown by <GAP> in the table above.
[21.695055] rcar_sound ec500000.sound: b adg[0]-CMDOUT_TIMSEL (32):00000f00/00000f1f
Correct the timsel assignment.
Fixes: 629509c5bc478c ("ASoC: rsnd: add Gen2 SRC and DMAEngine support")
Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Andreas Pape <Andreas.Pape4@bosch.com>
Signed-off-by: Yeswanth Rayapati <yeswanth.rayapati@in.bosch.com>
Tested-by: Yeswanth Rayapati <yeswanth.rayapati@in.bosch.com>
[erosca: massage commit description]
Signed-off-by: Eugeniu Rosca <eugeniu.rosca@bosch.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://msgid.link/r/20240301085003.3057-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
There was one overlooked place to be replaced with
snd_ctl_find_id_mixer() for code simplification.
No functional change, only code refactoring.
Link: https://lore.kernel.org/r/20240304082158.8583-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Use the macro to improve readability.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-6-cezary.rojewski@intel.com
|
|
Be cohesive and use same pattern in each error message.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-5-cezary.rojewski@intel.com
|
|
HDMI codecs which are present and functional from audio perspective lack
i915 support on drm side what results in -ENODEV during the probing
sequence. There is no reason to perform recovery procedure e.g.: reset
the HDAudio controller if this is the case.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-4-cezary.rojewski@intel.com
|
|
If i915 does not support given platform but the hardware i.e.: HDAudio
codec is still there, the codec-probing procedure will succeed for such
device but the follow up initialization will always end up with -ENODEV.
While bus could filter out address '2' which Intel's HDMI/DP codecs
always enumerate on, more robust approach is to check for i915 presence
before registering display codecs.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-3-cezary.rojewski@intel.com
|
|
Commit 78f613ba1efb ("drm/i915: finish removal of CNL") and its friends
removed support for i915 for all CNL-based platforms. HDAudio library,
however, still treats such platforms as valid candidates for i915
binding. Update query mechanism to reflect changes made in drm tree.
At the same time, i915 support for LKF-based platforms has not been
provided so remove them from valid binding candidates.
Link: https://lore.kernel.org/all/20210728215946.1573015-1-lucas.demarchi@intel.com/
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20240226124432.1203798-2-cezary.rojewski@intel.com
|
|
When building feature controls from a unit without a name, we try to
derive a name first from the feature unit's input, then fall back to the
output terminal.
If a feature unit connects directly to a "USB Streaming" input terminal
rather than a mixer or other virtual type, the control receives the
somewhat meaningless name "PCM", even if the output had a descriptive
type such as "Headset" or "Speaker".
Here is an example of such AudioControl descriptor from a USB headset
which ends up named "PCM Playback" and is therefore not recognized as
headphones by userspace:
AudioControl Interface Descriptor:
bLength 12
bDescriptorType 36
bDescriptorSubtype 2 (INPUT_TERMINAL)
bTerminalID 4
wTerminalType 0x0101 USB Streaming
bAssocTerminal 5
bNrChannels 2
wChannelConfig 0x0003
Left Front (L)
Right Front (R)
iChannelNames 0
iTerminal 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 3 (OUTPUT_TERMINAL)
bTerminalID 5
wTerminalType 0x0402 Headset
bAssocTerminal 4
bSourceID 6
iTerminal 0
AudioControl Interface Descriptor:
bLength 13
bDescriptorType 36
bDescriptorSubtype 6 (FEATURE_UNIT)
bUnitID 6
bSourceID 4
bControlSize 2
bmaControls(0) 0x0002
Volume Control
bmaControls(1) 0x0000
bmaControls(2) 0x0000
iFeature 0
Other headsets and DACs I tried that used their output terminal for
naming only did so due to their input being an unnamed sidetone mixer.
Instead of always starting with the input terminal, check the type of it
first. If it seems uninteresting, invert the order and use the output
terminal first for naming.
This makes userspace recognize headsets with simple controls as
headphones, and leads to more consistent naming of playback devices
based on their outputs irrespective of sidetone mixers.
Signed-off-by: Kenny Levinsen <kl@kl.wtf>
Link: https://lore.kernel.org/r/20240301231107.42679-1-kl@kl.wtf
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Whilst this laptop contains _DSD inside the BIOS, there is an error in
this configuration. Override the _DSD in the BIOS with the correct
configuration for this laptop.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-4-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
These models use 2 CS35L41 amps with HDA using I2C.
Both models have _DSD support inside cs35l41_hda_property.c.
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218437
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-3-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Adds sound support for 2 Lenovo Thinkbook 16P laptops using CS35L41
HDA with External Boost.
SSIDs:
- 17AA38A9
- 17AA38AB
Closes: https://bugzilla.kernel.org/show_bug.cgi?id=218437
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20240301160154.158398-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It will be enable headset Mic for Acer NB platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/fe0eb6661ca240f3b7762b5b3257710d@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
snd_hwdep_control_ioctl()
Clang prior to 17.0.0 has a bug in its asm goto jump scope analysis to
determine that no variables with the cleanup attribute are skipped by an
indirect jump. Instead of only checking the scope of each label that is
a possible target of each asm goto statement, it checks the scope of
every label, which can cause an error when a variable with the cleanup
attribute is used between two asm goto statements with different scopes,
even if they have completely different label targets:
sound/core/hwdep.c:273:8: error: cannot jump from this asm goto statement to one of its possible targets
if (get_user(device, (int __user *)arg))
^
arch/powerpc/include/asm/uaccess.h:295:5: note: expanded from macro 'get_user'
__get_user(x, _gu_addr) : \
^
arch/powerpc/include/asm/uaccess.h:283:2: note: expanded from macro '__get_user'
__get_user_size_allowed(__gu_val, __gu_addr, __gu_size, __gu_err); \
^
arch/powerpc/include/asm/uaccess.h:199:3: note: expanded from macro '__get_user_size_allowed'
__get_user_size_goto(x, ptr, size, __gus_failed); \
^
arch/powerpc/include/asm/uaccess.h:187:10: note: expanded from macro '__get_user_size_goto'
case 1: __get_user_asm_goto(x, (u8 __user *)ptr, label, "lbz"); break; \
^
arch/powerpc/include/asm/uaccess.h:158:2: note: expanded from macro '__get_user_asm_goto'
asm_volatile_goto( \
^
include/linux/compiler_types.h:366:33: note: expanded from macro 'asm_volatile_goto'
#define asm_volatile_goto(x...) asm goto(x)
^
sound/core/hwdep.c:291:9: note: possible target of asm goto statement
if (put_user(device, (int __user *)arg))
^
arch/powerpc/include/asm/uaccess.h:66:5: note: expanded from macro 'put_user'
__put_user(x, _pu_addr) : -EFAULT; \
^
arch/powerpc/include/asm/uaccess.h:52:9: note: expanded from macro '__put_user'
\
^
sound/core/hwdep.c:276:4: note: jump bypasses initialization of variable with __attribute__((cleanup))
scoped_guard(mutex, ®ister_mutex) {
^
include/linux/cleanup.h:169:20: note: expanded from macro 'scoped_guard'
for (CLASS(_name, scope)(args), \
To avoid this issue, move the put_user() call out of the scoped_guard()
scope, which allows the asm goto scope analysis to see that the variable
with the cleanup attribute will never be skipped by the asm goto
statements.
There should be no functional change because prior to the refactoring,
put_user() was not called under register_mutex, so this call does not
even need to be in the scoped_guard() in the first place.
Fixes: e6684d08cc19 ("ALSA: hwdep: Use guard() for locking")
Closes: https://github.com/ClangBuiltLinux/linux/issues/2003
Signed-off-by: Nathan Chancellor <nathan@kernel.org>
Link: https://lore.kernel.org/r/20240301-fix-snd-hwdep-guard-v1-1-6aab033f3f83@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In azx_probe_codecs function, when bus->codec_mask is becomes to 0(no codecs),
execute azx_init_chip, bus->codec_mask will be initialized to a value again,
this causes snd_hda_codec_new function to run, the process is as follows:
-->snd_hda_codec_new
-->snd_hda_codec_device_init
-->snd_hdac_device_init---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_VENDOR_ID) 2s
---snd_hdac_read_parm(...AC_PAR_SUBSYSTEM_ID) 2s
---snd_hdac_read_parm(...AC_PAR_REV_ID) 2s
---snd_hdac_read_parm(...AC_PAR_NODE_COUNT) 2s
when no codecs, read communication is error, each command will be polled for
2 second, a total of 10s, it is easy to some problem.
like this:
2 [ 14.833404][ 6] [ T164] hda 0006:00: Codec #0 probe error; disabling it...
3 [ 14.844178][ 6] [ T164] hda 0006:00: codec_mask = 0x1
4 [ 14.880532][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
5 [ 15.891988][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0000
6 [ 16.978090][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0001
7 [ 18.140895][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0002
8 [ 19.135516][ 6] [ T164] hda 0006:00: too slow response, last cmd=0x0f0004
10 [ 19.900086][ 6] [ T164] hda 0006:00: no codecs initialized
11 [ 45.573398][ 2] [ C2] watchdog: BUG: soft lockup - CPU#2 stuck for 22s! [kworker/2:0:25]
Here, when bus->codec_mask is 0, use a direct break to avoid execute snd_hda_codec_new function.
Signed-off-by: songxiebing <songxiebing@kylinos.cn>
Link: https://lore.kernel.org/r/20240301011841.7247-1-soxiebing@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
platform
Headset Mic will no show at resume back.
This patch will fix this issue.
Fixes: d7f32791a9fc ("ALSA: hda/realtek - Add headset Mic support for Lenovo ALC897 platform")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/4713d48a372e47f98bba0c6120fd8254@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The bios version can differ depending if it is a dual-boot variant of the tablet.
Therefore another DMI match is required.
Signed-off-by: Alban Boyé <alban.boye@protonmail.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20240228192807.15130-1-alban.boye@protonmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Fix a typo in the shift value used in madera_set_fll_clks.
Fixes: 3863857dd5ca3 ("ASoC: madera: Enable clocks for input pins when used for the FLL")
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Link: https://msgid.link/r/20240229114637.352098-1-stuarth@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.8
A few small fixes, some driver specific and one slightly larger one
from Richard which adds a new core helper and updates a small clutch of
drivers to deal with the fact that they were using a helper which
requires that the lock for the list of controls without holding that
lock. We also have some quirks for new AMD based Lenovo systems.
|