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commit d18ca8635db2f88c17acbdf6412f26d4f6aff414 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2e93a29b48a017c777d4fcbfcc51aba4e6a90d38 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2d5af3ab9e6f1cf1468b2a5221b5c1f7f46c3333 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit c4e51e424e2c772ce1836912a8b0b87cd61bc9d5 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c158cf914713efc3bcdc25680c7156c48c12ef6a ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6db26f9ea4edd8a17d39ab3c20111e3ccd704aef ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bf5e4887eeddb48480568466536aa08ec7f179a5 ]
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 so it could not be done in .trigger(), which is atomic by default.
It turns out there is a problem on capture path of the AXG series.
The FIFO may get out of sync with the TDM decoder if the IP are not enabled
in a specific order. The FIFO must be enabled before the formatter starts
producing data. IOW, we must deal with FE before the BE. The .prepare()
callback is called on the BEs before the FE so it is not OK for the AXG.
The .trigger() callback order can be configured, and it deals with the FE
before the BEs by default. To solve our problem, we just need to start and
stop the formatters from the .trigger() callback. It is OK do so now that
the links have been made 'nonatomic' in the card driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211020114217.133153-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e138233e56e9829e65b6293887063a1a3ccb2d68 ]
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, it is clear.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20211020114217.133153-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b11d26660dff8d7430892008616452dc8e5fb0f3 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9e6f39535c794adea6ba802a52c722d193c28124 ]
Use FIELD_GET() and FIELD_PREP() helpers instead of doing it manually.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240227150826.573581-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: b11d26660dff ("ASoC: meson: axg-fifo: use threaded irq to check periods")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 23fb6bc2696119391ec3a92ccaffe50e567c515e ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ]
Miglia Harmony Audio (OXFW970) has a quirk to put the number of
accumulated quadlets in CIP payload into the dbc field of CIP header.
This commit handles the quirk in the packet processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 1576f263ee2147dc395531476881058609ad3d38 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit fc563aa900659a850e2ada4af26b9d7a3de6c591 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit aae86cfd8790bcc7693a5a0894df58de5cb5128c ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ee287771644394d071e6a331951ee8079b64f9a7 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 310a5caa4e861616a27a83c3e8bda17d65026fa8 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 051e0840ffa8ab25554d6b14b62c9ab9e4901457 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d397b6e56151099cf3b1f7bfccb204a6a8591720 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit a39d51ff1f52cd0b6fe7d379ac93bd8b4237d1b7 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c062166995c9e57d5cd508b332898f79da319802 ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 59c6a3a43b221cc2a211181b1298e43b2c2df782 ]
According to Amlogic datasheets for the SoCs supported by this driver, the
maximum bit clock rate is 100MHz.
The tdm interface allows the rates listed by the DAI driver, regardless of
the number slots or their width. However, these will impact the bit clock
rate.
Hitting the 100MHz limit is very unlikely for most use cases but it is
possible.
For example with 32 slots / 32 bits wide, the maximum rate is no longer
384kHz but ~96kHz.
Add the constraint accordingly if the component is not already active.
If it is active, the rate is already constrained by the first stream rate.
Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e3741a8d28a1137f8b19ae6f3d6e3be69a454a0a ]
By default, when mclk-fs is not provided, the tdm-interface driver
requests an MCLK that is 4x the bit clock, SCLK.
However there is no justification for this:
* If the codec needs MCLK for its operation, mclk-fs is expected to be set
according to the codec requirements.
* If the codec does not need MCLK the minimum is 2 * SCLK, because this is
minimum the divider between SCLK and MCLK can do.
Multiplying by 4 may cause problems because the PLL limit may be reached
sooner than it should, so use 2x instead.
Fixes: d60e4f1e4be5 ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://msgid.link/r/20240223175116.2005407-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5ad992c71b6a8e8a547954addc7af9fbde6ca10a ]
clang-16 warns about casting functions to incompatible types, as is done
here to call clk_disable_unprepare:
sound/soc/meson/t9015.c:274:4: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict]
274 | (void(*)(void *))clk_disable_unprepare,
The pattern of getting, enabling and setting a disable callback for a
clock can be replaced with devm_clk_get_enabled(), which also fixes
this warning.
Fixes: 33901f5b9b16 ("ASoC: meson: add t9015 internal DAC driver")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Justin Stitt <justinstitt@google.com>
Link: https://msgid.link/r/20240213215807.3326688-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 98ac85a00f31d2e9d5452b825a9ed0153d934043 ]
clang-16 warns about casting functions to incompatible types, as is done
here to call clk_disable_unprepare:
sound/soc/meson/aiu.c:243:12: error: cast from 'void (*)(struct clk *)' to 'void (*)(void *)' converts to incompatible function type [-Werror,-Wcast-function-type-strict]
243 | (void(*)(void *))clk_disable_unprepare,
The pattern of getting, enabling and setting a disable callback for a
clock can be replaced with devm_clk_get_enabled(), which also fixes
this warning.
Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Reviewed-by: Justin Stitt <justinstitt@google.com>
Link: https://msgid.link/r/20240213215807.3326688-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2ff4e003e8e105fb65c682c876a5cb0e00f854bf ]
Use the dev_err_probe() helper, instead of open-coding the same
operation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20211214020843.2225831-17-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Stable-dep-of: 98ac85a00f31 ("ASoC: meson: aiu: fix function pointer type mismatch")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d7bf73809849463f76de42aad62c850305dd6c5d ]
clang-16 points out a control flow integrity (kcfi) issue when event
callbacks get converted to incompatible types:
sound/core/seq/seq_midi.c:135:30: error: cast from 'int (*)(struct snd_rawmidi_substream *, const char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
135 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)dump_midi, substream);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/core/seq/seq_virmidi.c:83:31: error: cast from 'int (*)(struct snd_rawmidi_substream *, const unsigned char *, int)' to 'snd_seq_dump_func_t' (aka 'int (*)(void *, void *, int)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
83 | snd_seq_dump_var_event(ev, (snd_seq_dump_func_t)snd_rawmidi_receive, vmidi->substream);
| ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
For addressing those errors, introduce wrapper functions that are used
for callbacks and bridge to the actual function call with pointer
cast.
The code was originally added with the initial ALSA merge in linux-2.5.4.
[ the patch description shamelessly copied from Arnd's original patch
-- tiwai ]
Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20240213101020.459183-1-arnd@kernel.org
Link: https://lore.kernel.org/r/20240213135343.16411-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 96e202f8c52ac49452f83317cf3b34cd1ad81e18 ]
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: Sasha Levin <sashal@kernel.org>
[ Upstream commit f8b0127aca8c60826e7354e504a12d4a46b1c3bb ]
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>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b34bf65838f7c6e785f62681605a538b73c2808c ]
It had pop noise from Headphone port when system reboot state.
If NID 58h Index 0x0 to fill default value, it will reduce pop noise.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/7493e207919a4fb3a0599324fd010e3e@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 551539a8606e28cb2a130f8ef3e9834235b456c4 ]
The DMI strings used for the LattePanda board DMI quirks are very generic.
Using the dmidecode database from https://linux-hardware.org/ shows
that the chosen DMI strings also match the following 2 laptops
which also have a rt5645 codec:
Insignia NS-P11W7100 https://linux-hardware.org/?computer=E092FFF8BA04
Insignia NS-P10W8100 https://linux-hardware.org/?computer=AFB6C0BF7934
All 4 hw revisions of the LattePanda board have "S70CR" in their BIOS
version DMI strings:
DF-BI-7-S70CR100-*
DF-BI-7-S70CR110-*
DF-BI-7-S70CR200-*
LP-BS-7-S70CR700-*
See e.g. https://linux-hardware.org/?computer=D98250A817C0
Add a partial (non exact) DMI match on this string to make the LattePanda
board DMI match more precise to avoid false-positive matches.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://msgid.link/r/20240211212736.179605-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 668abe6dc7b61941fa5c724c06797efb0b87f070 ]
The quirk table entries should be put in the USB ID order, but some
entries have been put in random places. Re-sort them.
Fixes: bf990c102319 ("ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue")
Fixes: fd28941cff1c ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless")
Fixes: dfd5fe19db7d ("ALSA: usb-audio: Add FIXED_RATE quirk for JBL Quantum610 Wireless")
Fixes: 4a63e68a2951 ("ALSA: usb-audio: Fix microphone sound on Nexigo webcam.")
Fixes: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Fixes: 4fb7c24f69c4 ("ALSA: usb-audio: Add quirk for Fiero SC-01")
Fixes: 2307a0e1ca0b ("ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0)")
Link: https://lore.kernel.org/r/20240124155307.16996-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 03a8b0df757f1beb21ba1626e23ca7412e48b525 ]
Fix following coccicheck error:
./sound/usb/endpoint.c:1671:8-10: ERROR: reference preceded by free on line 1671.
Here should be 'cp' rather than 'ip'.
Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock")
Signed-off-by: Wan Jiabing <wanjiabing@vivo.com>
Link: https://lore.kernel.org/r/20220518021617.10114-1-wanjiabing@vivo.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7822baa844a87cbb93308c1032c3d47d4079bb8a ]
The RODE NT-USB+ is marketed as a professional usb microphone, however the
usb audio interface is a mess:
[ 1.130977] usb 1-5: new full-speed USB device number 2 using xhci_hcd
[ 1.503906] usb 1-5: config 1 has an invalid interface number: 5 but max is 4
[ 1.503912] usb 1-5: config 1 has no interface number 4
[ 1.519689] usb 1-5: New USB device found, idVendor=19f7, idProduct=0035, bcdDevice= 1.09
[ 1.519695] usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ 1.519697] usb 1-5: Product: RØDE NT-USB+
[ 1.519699] usb 1-5: Manufacturer: RØDE
[ 1.519700] usb 1-5: SerialNumber: 1D773A1A
[ 8.327495] usb 1-5: 1:1: cannot get freq at ep 0x82
[ 8.344500] usb 1-5: 1:2: cannot get freq at ep 0x82
[ 8.365499] usb 1-5: 2:1: cannot get freq at ep 0x2
Add QUIRK_FLAG_GET_SAMPLE_RATE to work around the broken sample rate get.
I have asked Rode support to fix it, but they show no interest.
Signed-off-by: Sean Young <sean@mess.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240124151524.23314-1-sean@mess.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a63e68a295187ae3c1cb3fa0c583c96a959714f ]
I own an external usb Webcam, model NexiGo N930AF, which had low mic volume and
inconsistent sound quality. Video works as expected.
(snip)
[ +0.047857] usb 5-1: new high-speed USB device number 2 using xhci_hcd
[ +0.003406] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17
[ +0.000007] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam
[ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD
[ +0.000004] usb 5-1: SerialNumber: 20201217011
[ +0.003900] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283)
[ +0.025726] usb 5-1: 3:1: cannot get usb sound sample rate freq at ep 0x86
[ +0.071482] usb 5-1: 3:2: cannot get usb sound sample rate freq at ep 0x86
[ +0.004679] usb 5-1: 3:3: cannot get usb sound sample rate freq at ep 0x86
[ +0.051607] usb 5-1: Warning! Unlikely big volume range (=4096), cval->res is probably wrong.
[ +0.000005] usb 5-1: [7] FU [Mic Capture Volume] ch = 1, val = 0/4096/1
Set up quirk cval->res to 16 for 256 levels,
Set GET_SAMPLE_RATE quirk flag to stop trying to get the sample rate.
Confirmed that happened anyway later due to the backoff mechanism, after 3 failures
All audio stream on device interfaces share the same values,
apart from wMaxPacketSize and tSamFreq :
(snip)
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 3
bAlternateSetting 3
bNumEndpoints 1
bInterfaceClass 1 Audio
bInterfaceSubClass 2 Streaming
bInterfaceProtocol 0
iInterface 0
AudioStreaming Interface Descriptor:
bLength 7
bDescriptorType 36
bDescriptorSubtype 1 (AS_GENERAL)
bTerminalLink 8
bDelay 1 frames
wFormatTag 0x0001 PCM
AudioStreaming Interface Descriptor:
bLength 11
bDescriptorType 36
bDescriptorSubtype 2 (FORMAT_TYPE)
bFormatType 1 (FORMAT_TYPE_I)
bNrChannels 1
bSubframeSize 2
bBitResolution 16
bSamFreqType 1 Discrete
tSamFreq[ 0] 44100
Endpoint Descriptor:
bLength 9
bDescriptorType 5
bEndpointAddress 0x86 EP 6 IN
bmAttributes 5
Transfer Type Isochronous
Synch Type Asynchronous
Usage Type Data
wMaxPacketSize 0x005c 1x 92 bytes
bInterval 4
bRefresh 0
bSynchAddress 0
AudioStreaming Endpoint Descriptor:
bLength 7
bDescriptorType 37
bDescriptorSubtype 1 (EP_GENERAL)
bmAttributes 0x01
Sampling Frequency
bLockDelayUnits 0 Undefined
wLockDelay 0x0000
(snip)
Based on the usb data about manufacturer, SPCA2281B3 is the most likely controller IC
Manufacturer does not provide link for datasheet nor detailed specs.
No way to confirm if the firmware supports any other way of getting the sample rate.
Testing patch provides consistent good sound recording quality and volume range.
(snip)
[ +0.045764] usb 5-1: new high-speed USB device number 2 using xhci_hcd
[ +0.106290] usb 5-1: New USB device found, idVendor=1bcf, idProduct=2283, bcdDevice=12.17
[ +0.000006] usb 5-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3
[ +0.000004] usb 5-1: Product: NexiGo N930AF FHD Webcam
[ +0.000003] usb 5-1: Manufacturer: SHENZHEN AONI ELECTRONIC CO., LTD
[ +0.000004] usb 5-1: SerialNumber: 20201217011
[ +0.043700] usb 5-1: set resolution quirk: cval->res = 16
[ +0.002585] usb 5-1: Found UVC 1.00 device NexiGo N930AF FHD Webcam (1bcf:2283)
Signed-off-by: Christos Skevis <xristos.thes@gmail.com>
Link: https://lore.kernel.org/r/20231006155330.399393-1-xristos.thes@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fd28941cff1cd9d8ffa59fe11eb64148e09b6ed6 ]
It seems that the firmware is broken and does not accept
the UAC_EP_CS_ATTR_SAMPLE_RATE URB. There is only one rate (48000Hz)
available in the descriptors for the output endpoint.
Create a new quirk QUIRK_FLAG_FIXED_RATE to skip the rate setup
when only one rate is available (fixed).
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216798
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20221215153037.1163786-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 67df411db3f0209e4bb5227d4dd9d41b21368b9d ]
Tascam's Model 12 is a mixer which can also operate as a USB audio
interface. The audio interface uses explicit feedback but it seems that
it does not correctly handle missing isochronous frames.
When injecting an xrun (or doing anything else that pauses the playback
stream) the feedback rate climbs (for example, at 44,100Hz nominal, I
see a stable rate around 44,099 but xrun injection sees this peak at
around 44,135 in most cases) and glitches are heard in the audio stream
for several seconds - this is significantly worse than the single glitch
expected for an underrun.
While the stream does normally recover and the feedback rate returns to
a stable value, I have seen some occurrences where this does not happen
and the rate continues to increase while no audio is heard from the
output. I have not found a solid reproduction for this.
This misbehaviour can be avoided by totally resetting the stream state
by switching the interface to alt 0 and back before restarting the
playback stream.
Add a new quirk flag which forces the endpoint and interface to be
reconfigured whenever the stream is stopped, and use this for the Tascam
Model 12.
Separate interfaces are used for the playback and capture endpoints, so
resetting the playback interface here will not affect the capture stream
if it is running. While there are two endpoints on the interface,
these are the OUT data endpoint and the IN explicit feedback endpoint
corresponding to it and these are always stopped and started together.
Signed-off-by: John Keeping <john@metanate.com>
Link: https://lore.kernel.org/r/20221129130100.1257904-1-john@metanate.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1045f5f1ff0751423aeb65648e5e1abd7a7a8672 ]
After splitting to snd_usb_endpoint_set_params() and *_prepare(), the
skip of each function should be checked with different flags, while we
still use ep->need_setup as the single one. Introduce
ep->need_prepare for indicating the need of prepare, and also add the
missing check of ep->need_setup at the set_params.
Fixes: 2be79d586454 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)")
Link: https://lore.kernel.org/r/20221009104212.18877-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9355b60e401d825590d37f04ea873c58efe9b7bf ]
snd_usb_endpoint_set_params() should return zero for a success, but
currently it returns the sample rate. Correct it.
Fixes: 2be79d586454 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)")
Link: https://lore.kernel.org/r/20221009104212.18877-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a74f8d0aa902ca494676b79226e0b5a1747b81d4 ]
The protection with chip->mutex was lost after splitting
snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare().
Apply the same mutex again to the former function.
Fixes: 2be79d586454 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)")
Link: https://lore.kernel.org/r/20221009104212.18877-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 9a737e7f8b371e97eb649904276407cee2c9cf30 ]
We fixed the bug introduced by the patch for managing the shared
clocks at the commit 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock
rate at closing EP"), but it was merely a workaround. By this change,
the clock reference rate is cleared at each EP close, hence the still
remaining EP may need a re-setup of rate unnecessarily.
This patch introduces the proper refcounting for the clock reference
object so that the clock setup is done only when needed.
Fixes: 809f44a0cc5a ("ALSA: usb-audio: Clear fixed clock rate at closing EP")
Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock")
Link: https://lore.kernel.org/r/20220920181126.4912-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2be79d58645465351af5320eb14c70a94724c5ef ]
This is a second attempt to fix the bug appearing on Android with the
recent kernel; the first try was ff878b408a03 and reverted at commit
79764ec772bc.
The details taken from the v1 patch:
One of the former changes for the endpoint management was the more
consistent setup of endpoints at hw_params.
snd_usb_endpoint_configure() is a single function that does the full
setup, and it's called from both PCM hw_params and prepare callbacks.
Although the EP setup at the prepare phase is usually skipped (by
checking need_setup flag), it may be still effective in some cases
like suspend/resume that requires the interface setup again.
As it's a full and single setup, the invocation of
snd_usb_endpoint_configure() includes not only the USB interface setup
but also the buffer release and allocation. OTOH, doing the buffer
release and re-allocation at PCM prepare phase is rather superfluous,
and better to be done only in the hw_params phase.
For those optimizations, this patch splits the endpoint setup to two
phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(),
to be called from hw_params and from prepare, respectively.
Note that this patch changes the driver operation slightly,
effectively moving the USB interface setup again to PCM prepare stage
instead of hw_params stage, while the buffer allocation and such
initializations are still done at hw_params stage.
And, the change of the USB interface setup timing (moving to prepare)
gave an interesting "fix", too: it was reported that the recent
kernels caused silent output at the beginning on playbacks on some
devices on Android, and this change casually fixed the regression.
It seems that those devices are picky about the sample rate change (or
the interface change?), and don't follow the too immediate rate
changes.
Meanwhile, Android operates the PCM in the following order:
- open, then hw_params with the possibly highest sample rate
- close without prepare
- re-open, hw_params with the normal sample rate
- prepare, and start streaming
This procedure ended up the hw_params twice with different rates, and
because the recent kernel did set up the sample rate twice one and
after, it screwed up the device. OTOH, the earlier kernels didn't set
up the USB interface at hw_params, hence this problem didn't appear.
Now, with this patch, the USB interface setup is again back to the
prepare phase, and it works around the problem automagically.
Although we should address the sample rate problem in a more solid
way in future, let's keep things working as before for now.
***
What's new in the take#2 patch:
- The regression caused by the v1 patch (bko#216500) was due to the
missing check of need_setup flag at hw_params. Now the check is
added, and the snd_usb_endpoint_set_params() call is skipped when
the running EP is re-opened.
- There was another bug in v1 where the clock reference rate wasn't
updated at hw_params phase, which may lead to a lack of the proper
hw constraints when an application doesn't issue the prepare but
only the hw_params call. This patch fixes it as well by tracking
the clock rate change in the prepare callback with a new flag
"need_update" for the clock reference object, just like others.
- The configure_endpoints() are simplified and folded back into
snd_usb_pcm_prepare().
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Fixes: ff878b408a03 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare")
Reported-by: chihhao chen <chihhao.chen@mediatek.com>
Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@mediatek.com
Link: https://lore.kernel.org/r/20220901124136.4984-1-tiwai@suse.de
Link: https://bugzilla.kernel.org/show_bug.cgi?id=216500
Link: https://lore.kernel.org/r/20220920181106.4894-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 809f44a0cc5ad4b1209467a6287f8ac0eb49d393 ]
The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple
accesses on the single clock") tries to manage the clock rate shared
by several endpoints. This was intended for avoiding the unmatched
rate by a different endpoint, but unfortunately, it introduced a
regression for PulseAudio and pipewire, too; those applications try to
probe the multiple possible rates (44.1k and 48kHz) and setting up the
normal rate fails but only the last rate is applied.
The cause is that the last sample rate is still left to the clock
reference even after closing the endpoint, and this value is still
used at the next open. It happens only when applications set up via
PCM prepare but don't start/stop the stream; the rate is reset when
the stream is stopped, but it's not cleared at close.
This patch addresses the issue above, simply by clearing the rate set
in the clock reference at the last close of each endpoint.
Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock")
Reported-by: Jason A. Donenfeld <Jason@zx2c4.com>
Tested-by: Jason A. Donenfeld <Jason@zx2c4.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/
Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620
Link: https://lore.kernel.org/r/20220907100421.6443-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c11117b634f4f832c4420d3cf41c44227f140ce1 ]
When a clock source is connected to multiple nodes / endpoints, the
current USB-audio driver tries to set up at each time one of them is
configured. Although it reads the current rate and updates only if it
differs, some devices seem unhappy with this behavior and spew the
errors when reading/updating the rate unnecessarily.
This patch tries to reduce the redundant clock setup by introducing a
refcount for each clock source. When the stream is actually running,
a clock rate is "locked", and it bypasses the clock and/or refuse to
change any longer.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934
Link: https://lore.kernel.org/r/20220516104807.16482-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 7822baa844a8 ("ALSA: usb-audio: add quirk for RODE NT-USB+")
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 77ce96543b03f437c6b45f286d8110db2b6622a3 upstream.
The local helper function to compare the given pair of cycle count
evaluates them. If the left value is less than the right value, the
function returns negative value.
If the safe cycle is less than the current cycle, it is the case of
cycle lost. However, it is not currently handled properly.
This commit fixes the bug.
Cc: <stable@vger.kernel.org>
Fixes: 705794c53b00 ("ALSA: firewire-lib: check cycle continuity")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20240218033026.72577-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 4df49712eb54141be00a9312547436d55677f092 ]
We forgot to remove the line for snd-rtctimer from Makefile while
dropping the functionality. Get rid of the stale line.
Fixes: 34ce71a96dcb ("ALSA: timer: remove legacy rtctimer")
Link: https://lore.kernel.org/r/20240221092156.28695-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
