ALSA: hda: cs35l56: Fix lifetime of cs_dsp instance
The cs_dsp instance is initialized in the driver probe() so it
should be freed in the driver remove(). Also fix a missing call
to cs_dsp_remove() in the error path of cs35l56_hda_common_probe().
The call to cs_dsp_remove() was being done in the component unbind
callback cs35l56_hda_unbind(). This meant that if the driver was
unbound and then re-bound it would be using an uninitialized cs_dsp
instance.
It is best to initialize the cs_dsp instance in probe() so that it
can return an error if it fails. The component binding API doesn't
have any error handling so there's no way to handle a failure if
cs_dsp was initialized in the bind.
ALSA: hda: hda_component: Initialize shared data during bind callback
Move the initialization of the shared struct hda_component array into
hda_component_manager_bind().
The purpose of the manager bind() callback is to allow it to perform
initialization before binding in the component drivers. This is the
correct place to initialize the shared data.
The original implementation initialized the shared data in
hda_component_manager_init(). This is only done once during probe()
of the manager driver. So if the component binding was unbound and
then rebound, the shared data would not be re-initialized.
ALSA: hda/cs_dsp_ctl: Use private_free for control cleanup
Use the control private_free callback to free the associated data
block. This ensures that the memory won't leak, whatever way the
control gets destroyed.
The original implementation didn't actually remove the ALSA
controls in hda_cs_dsp_control_remove(). It only freed the internal
tracking structure. This meant it was possible to remove/unload the
amp driver while leaving its ALSA controls still present in the
soundcard. Obviously attempting to access them could cause segfaults
or at least dereferencing stale pointers.
Takashi Iwai [Tue, 7 May 2024 13:55:09 +0000 (15:55 +0200)]
ALSA: misc: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:08 +0000 (15:55 +0200)]
ALSA: aoa: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:07 +0000 (15:55 +0200)]
ALSA: firewire: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:06 +0000 (15:55 +0200)]
ALSA: drivers: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:05 +0000 (15:55 +0200)]
ALSA: usb: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:04 +0000 (15:55 +0200)]
ALSA: isa: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:03 +0000 (15:55 +0200)]
ALSA: hda: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:02 +0000 (15:55 +0200)]
ALSA: pci: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Takashi Iwai [Tue, 7 May 2024 13:55:01 +0000 (15:55 +0200)]
ALSA: core: Use *-y instead of *-objs in Makefile
*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Jaroslav Kysela [Mon, 6 May 2024 07:54:19 +0000 (09:54 +0200)]
selftests/alsa: make dump_config_tree() as void function
dump_config_tree() is declared to return an int, but the compiler cannot
prove that it always returns any value at all. This leads to a clang
warning, when building via:
make LLVM=1 -C tools/testing/selftests
Suggested-by: John Hubbard <jhubbard@nvidia.com> Cc: Mark Brown <broonie@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Reviewed-by: Mark Brown <broonie@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20240506075419.301780-1-perex@perex.cz
Wolfram Sang [Tue, 30 Apr 2024 12:10:27 +0000 (14:10 +0200)]
ALSA: aoa: soundbus: i2sbus: pcm: use 'time_left' variable with wait_for_completion_timeout()
There is a confusing pattern in the kernel to use a variable named 'timeout' to
store the result of wait_for_completion_timeout() causing patterns like:
timeout = wait_for_completion_timeout(...)
if (!timeout) return -ETIMEDOUT;
with all kinds of permutations. Use 'time_left' as a variable to make the code
self explaining.
Fix to the proper variable type 'unsigned long' while here.
ALSA: usb-audio: Add sampling rates support for Mbox3
This adds support for all sample rates supported by the
hardware,Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000}
Fixes syncing clock issues that presented as pops. To test this, without
this patch playing 440hz tone produces pops.
Clock is now synced between playback and capture interfaces so no more
latency drift issue when using pipewire pro-profile.
(https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900)
Stefan Binding [Mon, 29 Apr 2024 15:48:52 +0000 (16:48 +0100)]
ALSA: hda: cs35l41: Ignore errors when configuring IRQs
IRQs used for CS35L41 HDA are used to detect and attempt to recover
from errors. Without these interrupts, the driver should behave as
normal.
For laptops which contain a bad configuration for the interrupt in the
BIOS, the current behaviour of failing when trying to configure the
interrupt means the probe fails, and audio is broken.
It is better for the user experience if the driver instead warns that
no interrupt is configured rather than simply failing.
The drawback is that if an error occurs without the interrupt, we
firstly would not be able to trace the issue, and secondly would not
be able to attempt to recover from the issue, but this is better than
failing immediately.
Takashi Iwai [Mon, 6 May 2024 16:13:45 +0000 (18:13 +0200)]
ALSA: hda: codec: Reduce CONFIG_PM dependencies
CONFIG_PM is almost mandatory nowadays for real systems, but we have
lots of CONFIG_PM dependent code in snd-hda-codec helper code.
Let's reduce the dependencies of CONFIG_PM now. The only visible
drawback would be a couple of superfluous trace entries for runtime
PM, but we can live with that.
Takashi Iwai [Mon, 6 May 2024 16:13:44 +0000 (18:13 +0200)]
ALSA: hda: intel: Reduce CONFIG_PM dependencies
snd-hda-intel contains lots of CONFIG_PM dependent code although
CONFIG_PM is almost mandatory nowadays, and it makes the code
unnecessarily complex.
Let's reduce the dependencies of CONFIG_PM in snd-hda-intel driver
code. I left a few module options to be dependent on CONFIG_PM (which
are visible to users), but other places are either enabled or
optimized by compiler automatically.
Takashi Iwai [Thu, 2 May 2024 06:24:42 +0000 (08:24 +0200)]
ALSA: hda/realtek: Fix build error without CONFIG_PM
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>
Takashi Iwai [Wed, 1 May 2024 16:05:13 +0000 (18:05 +0200)]
Merge tag 'asoc-fix-v6.9-rc6' of 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.
Kailang Yang [Tue, 30 Apr 2024 09:15:53 +0000 (17:15 +0800)]
ALSA: hda/realtek - Set GPIO3 to default at S4 state for Thinkpad with ALC1318
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.
Mark Brown [Tue, 30 Apr 2024 14:36:23 +0000 (23:36 +0900)]
ASoC: meson: tdm fixes
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.
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
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.
ASoC: meson: axg-tdm-interface: manage formatters in trigger
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.
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.
ASoC: meson: axg-fifo: use threaded irq to check periods
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.
Firstly, it is pointless to explicitly disable the power to the dock
prior to resetting the FPGA, as the latter will do the former anyway.
Secondly, it doesn't make much sense to check whether the FPGA is
already programmed. It's much simpler to just presume it is, and issue
the self-reset command. If it isn't, the effect isn't worse than the
checks themselves. As a side effect, we lose the info if the reset
fails, but there is no plausible way how that could happen unless the
card burns out while operating, and in that case we'll detect a firmware
upload failure a bit later anyway.
ALSA: emu10k1: make E-MU FPGA writes potentially more reliable
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.
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.
ALSA: emu10k1: use mutex for E-MU FPGA access locking
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.
ALSA: emu10k1: move the whole GPIO event handling to the workqueue
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}().
ALSA: emu10k1: factor out snd_emu1010_load_dock_firmware()
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.
ALSA: emu10k1: fix E-MU card dock presence monitoring
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.
Colin Ian King [Thu, 25 Apr 2024 16:07:54 +0000 (17:07 +0100)]
ALSA: kunit: make read-only array buf_samples static const
Don't populate the read-only array buf_samples on the stack at
run time, instead make it static const.
Signed-off-by: Colin Ian King <colin.i.king@gmail.com> Acked-by: Ivan Orlov <ivan.orlov0322@gmail.com> Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240425160754.114716-1-colin.i.king@gmail.com>
Simon Trimmer [Mon, 22 Apr 2024 10:32:11 +0000 (11:32 +0100)]
ASoC: cs35l56: Avoid static analysis warning of uninitialised variable
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>
ALSA: scarlett2: Zero initialize ret in scarlett2_ag_target_ctl_get()
Clang warns (or errors with CONFIG_WERROR):
sound/usb/mixer_scarlett2.c:3697:6: error: variable 'err' is used uninitialized whenever 'if' condition is false [-Werror,-Wsometimes-uninitialized]
3697 | if (private->autogain_updated) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~
sound/usb/mixer_scarlett2.c:3707:9: note: uninitialized use occurs here
3707 | return err;
| ^~~
sound/usb/mixer_scarlett2.c:3697:2: note: remove the 'if' if its condition is always true
3697 | if (private->autogain_updated) {
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/usb/mixer_scarlett2.c:3688:9: note: initialize the variable 'err' to silence this warning
3688 | int err;
| ^
| = 0
1 error generated.
Initialize ret to zero to ensure ret is initialized in all paths within
scarlett2_ag_target_ctl_get(), which matches the style of other
functions in this driver.
Although the purpose of dummy seq client is a direct pass-through,
it's sometimes helpful for debugging if it can convert to a certain
UMP MIDI version. This patch adds an option to specify the UMP event
conversion. As default, it skips the conversion and does
passthrough, while user can pass ump=1 or ump=2 to enforce the
conversion to UMP MIDI1 or MIDI2 format.
ALSA: seq: ump: Fix conversion from MIDI2 to MIDI1 UMP messages
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.
Ai Chao [Fri, 19 Apr 2024 08:21:59 +0000 (16:21 +0800)]
ALSA: hda/realtek - Enable audio jacks of Haier Boyue G42 with ALC269VC
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.
ASoC: ti: davinci-mcasp: Fix race condition during probe
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>
ASoC: Intel: avs: Set name of control as in topology
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.
Pavel Hofman [Tue, 16 Apr 2024 12:17:25 +0000 (14:17 +0200)]
ALSA: pcm: add support for 705.6kHz and 768kHz sample rates
Many modern codecs support 705.6kHz and 768kHz sample rates. Current HW
params fail to set 705.6kHz and 768kHz sample rates as these are not in the
known-rates list.
Add these new rates to the known-rates list to allow them.
Also add defines in pcm.h so that drivers can use it.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-3-pavel.hofman@ivitera.com>
Pavel Hofman [Tue, 16 Apr 2024 12:17:24 +0000 (14:17 +0200)]
ALSA: aloop: add DSD formats
The snd-aloop loopback driver does not modify or access the actual samples
in any way, defines no volume or mute controls, it's strictly bitperfect.
Therefore DSD formats can be supported without any modification.
Add all DSD formats to the list of supported formats.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com> Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240416121726.628679-2-pavel.hofman@ivitera.com>
ALSA: hda/realtek: Fix volumn control of ThinkBook 16P Gen4
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).
ALSA: hda/realtek: Fixes for Asus GU605M and GA403U sound
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>
Stefan Binding [Thu, 11 Apr 2024 11:08:13 +0000 (12:08 +0100)]
ALSA: hda: cs35l41: Remove Speaker ID for Lenovo Legion slim 7 16ARHA7
These laptops do not have _DSD and must be added by configuration
table, however, the initial entries for them are incorrect:
Neither laptop contains a Speaker ID GPIO.
This issue would not affect audio playback, but may affect which files
are loaded when loading firmware.
ALSA: hda: cs35l41: Remove redundant argument to cs35l41_request_firmware_file()
In every case the 'dir' argument to cs35l41_request_firmware_file() is passed
the string "cirrus/", so this is a redundant argument and can be removed.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-7-sbinding@opensource.cirrus.com>
Stefan Binding [Thu, 11 Apr 2024 11:08:11 +0000 (12:08 +0100)]
ALSA: hda: cs35l41: Use shared cs-amp-lib to apply calibration
The original mechanism for applying calibration assumed that the
calibration data would be ordered the same as the amp instances.
However, for some 4 amp laptops, this is not the case.
To ensure that the correct calibration is applied to the correct amp,
the calibration data contains a unique id, which matches a unique id
inside the CS35L41. This can be used to match to the correct data
entry. This mechanism is available inside the shared module cs-amp-lib.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-6-sbinding@opensource.cirrus.com>
Stefan Binding [Thu, 11 Apr 2024 11:08:10 +0000 (12:08 +0100)]
ALSA: hda: cs35l41: Update DSP1RX5/6 Sources for DSP config
Currently, all PC systems are set to use VBSTMON for DSP1RX5_SRC,
however, this is required only for external boost systems.
Internal boost systems require VPMON instead of VBSTMON to be the
input to DSP1RX5_SRC.
All systems require DSP1RX6_SRC to be set to VBSTMON.
Also fix incorrect comment for DACPCM1_SRC to use DSP1TX1.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-5-sbinding@opensource.cirrus.com>
Stefan Binding [Thu, 11 Apr 2024 11:08:07 +0000 (12:08 +0100)]
ALSA: hda: cs35l41: Set the max PCM Gain using tuning setting
Some systems requires different max PCM Gains settings than the default.
The current default value, when running firmware is 17.5 dB, which is
used for all systems. Some systems require lower values.
Value when running without firmware is 4.5 dB and remains unchanged.
Since the gain value is dependent on Tuning and Firmware, it can
change, so it cannot be saved in _DSD. Instead we can store it inside
a configuration binary file alongside the Firmware and Tuning files.
The gain value increments in steps of 1 dB, with value 0 representing
0.5 dB. The max value is 20, which corresponds to 20.5 dB.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240411110813.330483-2-sbinding@opensource.cirrus.com>
projects / thirdparty / linux.git / log
