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.
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>
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.
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.
Richard Fitzgerald [Thu, 11 Apr 2024 11:08:12 +0000 (12:08 +0100)]
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>
Peter Ujfalusi [Tue, 9 Apr 2024 08:38:10 +0000 (11:38 +0300)]
ALSA: pci: hda: hda_controller: Add support for use_pio_for_commands mode
Set the use_pio_for_commands flag in case AZX_DCAPS_PIO_COMMANDS quirk is
enabled.
When the PIO command mode is used we can re-use the existing
azx_single_send_cmd() / azx_single_get_response() functions safely as the
CORB DMA is not going to be enabled in snd_hdac_bus_init_cmd_io().
Peter Ujfalusi [Tue, 9 Apr 2024 08:38:09 +0000 (11:38 +0300)]
ALSA: hda: hdac_controller: Implement support for use_pio_for_commands mode
In case the use_pio_for_commands flag is set we must not enable the
CORB DMA to make sure that it is not interfering with the immediate
command mode.
Convert the snd_hdac_bus_send_cmd/snd_hdac_bus_get_response as wrappers to
call either the PIO or CORB based command handling depending on the
use_pio_for_commands flag.
Peter Ujfalusi [Tue, 9 Apr 2024 08:38:08 +0000 (11:38 +0300)]
ALSA: hda: Introduce flags to force commands via PIO instead of CORB
Add AZX_DCAPS_PIO_COMMANDS quirk (bit 31) and use_pio_for_commands flag to
be able to select PIO mode as alternative for CORB based command sending
while retaining the RIRB functionality to receive unsolicited responses.
This mode differs from the azx single_cmd mode when RIRB is disabled.
The mixed mode is needed on Lunar Lake family because it is recommended to
use Immediate Command Response (PIO mode) instead of CORB for HDA commands.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:37:56 +0000 (05:07 +1030)]
ALSA: scarlett2: Add autogain target controls
The Scarlett 4th Gen and Vocaster interfaces allow the autogain target
dBFS value(s) to be configured. Add Mean and Peak Target controls for
4th Gen, and a Hot Target control for Vocaster.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:37:12 +0000 (05:07 +1030)]
ALSA: scarlett2: Add DSP controls
Add filter and compressor DSP controls for the Vocaster interfaces.
Mark scarlett2_notify_input_dsp() as __always_unused until it gets
used when the Vocaster callback function array is added.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:36:23 +0000 (05:06 +1030)]
ALSA: scarlett2: Add input mute controls
Add controls for the input mute switches that the Vocaster interfaces
have. Mark scarlett2_notify_input_mute() as __always_unused until it
gets used when the Vocaster callback function array is added.
The 4th Gen Scarlett interfaces added software-controllable input gain
along with channel select, channel link, auto-gain, and "safe" mode.
Vocaster has software-controllable input gain and auto-gain but not
channel select, channel link, or safe mode.
Add a device info field safe_input_count to indicate how many channels
have a safe mode control, and use the presence of the input select and
input link switch configuration parameters to determine if those
controls should be created.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:35:15 +0000 (05:05 +1030)]
ALSA: scarlett2: Add pbuf field to struct scarlett2_config
scarlett2_usb_set_config() was using size = 0 as a signal to use the
parameter buffer. Replace that with an explicit indication (pbuf = 1),
as the upcoming Vocaster support has a config item written via the
parameter buffer with size = 1 rather than the implicit size of 8.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:34:14 +0000 (05:04 +1030)]
ALSA: scarlett2: Implement handling of the ACK notification
After scarlett2_usb() sends a command, it seems that we should wait
for an ACK before attempting to read the response. Not doing that
didn't seem necessary previously but seems to be causing occasional
issues with 4th Gen devices.
Geoffrey D. Bennett [Tue, 12 Mar 2024 18:33:14 +0000 (05:03 +1030)]
ALSA: scarlett2: Move initialisation code lower in the source
So that more forward declarations won't be required when we add
handling of the ACK notification, move the initialisation functions to
after the notification functions.
Both drivers provide both sample_new and sample_free, and it makes no
sense to pretend that they could not. In fact, load_data() would already
crash if sample_new was null. So remove the remaining null checks.
Contrary to that, the emu10k1 driver actually has a null sample_reset,
though I'm not convinced that this inconsistency is justified.
ALSA: emu10k1: shrink blank space in front of wavetable samples
There is no need for it to be 32 samples - 3 will do just fine (which is
the interpolator's epsilon). The old size was presumably meant to
compensate for the cache's presence, but we're now handling that
properly.
ALSA: emu10k1: fix wavetable playback position and caching, take 2
Compensate for the cache lag of 64 frames, and actually populate the
cache. Without these, the playback would start with garbage (which
would be (mostly?) masqueraded by the note's attack phase).
Note that we set the starting address only 61 frames ahead, to
compensate for the interpolator's epsilon. Unlike for PCM playback, we
don't even need to manually silence-fill the first frames in the cache,
because we insert some silence in front of each sample anyway.
A challenge are extremely short samples with a loop end below the cache
size, because a) we'd have to wrap the current address to be within the
loop and b) automatic pre-filling of the cache with the right data does
not work in this case.
We could pre-fill the cache manually, but that's slow, requires
additional code for each sample width, and is made even more complex by
the driver's virtual address space having no contiguous mapping for the
CPU.
We could have the engine fill the cache piece-wise (which is really what
happens when playback is running), but that would also be complex, and
we'd need to wait for the engine to handle each piece, so it wouldn't be
that much faster than the manual fill.
For the case of requiring only one loop iteration prior to reaching the
cache size, we could leverage the engine's looping mechanism around
CCR_CACHELOOPFLAG, but this special case doesn't seem worth the
complexity.
So we just unroll the loop as far as necessary to be able to play back
the sample without any fiddling.
Pedantically, this would be incorrect for loop-until-release samples
with a low loop end which are released very quickly, but that would be
relatively harmless, is not a plausible use case in the first place, and
SoundFont sample mode 3 isn't actually implemented anyway (it's
conflated with mode 1, infinite looping).
ALSA: emu10k1: move patch loader assertions into low-level functions
Convert some checks in snd_emu10k1_sample_new() back into assertions (as
they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*,
2008-08-08)), and move them into the low-level memory access functions
they protect.
ALSA: emux: centralize & improve patch info validation
This does several closely related things:
- Move the code from the drivers into the SoundFont loader, which
de-duplicates it.
- Sort of explain the weird "recalculate address offset" feature. Note
that I don't think it actually makes any sense - the calling user
space code should do that. The background is certainly that the source
data (the SoundFont format) uses pointers into a single wave block
(and the API allows doing the same for on-board ROM), but the API
expects the wave data from user space to be pre-chopped into
individual patches anyway.
- Make sure that the specified offsets actually lie within the supplied
wave data. Note that we don't validate ROM offsets, so one can play
back anything within the sound card's address space.
- In load_guspatch(), don't call the sample_new callback anymore when
the patch size is zero, as was already the case in load_data(). The
callbacks would instantly return in that case anyway; these checks are
now removed.
ALSA: emu10k1: prune vestiges of SNDRV_SFNT_SAMPLE_{BIDIR,REVERSE}_LOOP support
This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in
the GUS patch loader. It has not worked on emu10k1 since before ALSA hit
mainline, yet nobody appears to have complained. And as it isn't super
easy to implement, just admit defeat and clean up the code.
If somebody wanted to resurrect the feature, the emu8k driver could
serve as a template, but the code would be quite different. But
arguably, this should be done in user space in the first place, as this
doesn't represent a hardware feature (somewhat ironically, the actual
GUS driver has no synth support, and therefore no GUS patch loader).
Note that instead of properly rejecting affected samples, we continue to
just pretend that the feature wasn't requested. This is extremely
questionable behavior, but avoids that possibly unused instruments
suddenly prevent loading the entire file, which would break backwards
compatibility. But at least we log a warning now.
ALSA: emux: fix init of patch_info.truesize in load_data()
The field is explicitly documented to be initialized by the driver
(which it actually is). Also, using patch_info.size would be actually
wrong for 16-bit data, as one field counts samples, while the other
counts bytes.
Merge tag 'asoc-fix-v6.9-rc2' of 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.
ASoC: SOF: Core: Add remove_late() to sof_init_environment failure path
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
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.
Zhang Yi [Tue, 2 Apr 2024 06:20:42 +0000 (14:20 +0800)]
ASoC: codecs: ES8326: Solve a headphone detection issue after suspend and resume
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.
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.
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.
Luke D. Jones [Tue, 2 Apr 2024 01:51:26 +0000 (14:51 +1300)]
ALSA: hda/realtek: cs35l41: Support ASUS ROG G634JYR
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>
I Gede Agastya Darma Laksana [Mon, 1 Apr 2024 17:46:02 +0000 (00:46 +0700)]
ALSA: hda/realtek: Update Panasonic CF-SZ6 quirk to support headset with 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.
Revert "ALSA: emu10k1: fix synthesizer sample playback position and caching"
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.
Uwe Kleine-König [Fri, 29 Mar 2024 21:54:42 +0000 (22:54 +0100)]
OSS: dmasound/paula: Mark driver struct with __refdata to prevent section 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
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