IF YOU WOULD LIKE TO GET AN ACCOUNT, please write an
email to Administrator. User accounts are meant only to access repo
and report issues and/or generate pull requests.
This is a purpose-specific Git hosting for
BaseALT
projects. Thank you for your understanding!
Только зарегистрированные пользователи имеют доступ к сервису!
Для получения аккаунта, обратитесь к администратору.
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.
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: 738a2b5e2c ("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>
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
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>
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>
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: 0fca97a29b ("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>
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: df335e9a8b ("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>
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>
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: 7077148fb5 ("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>
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: 73cfbfa9ca ("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>
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>
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 c1947ce61f ("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: 5be27f1e3e ("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: b917d58dcf ("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 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: 198de43d75 ("[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>
