Age | Commit message (Collapse) | Author |
|
[ Upstream commit 690aa09b1845c0d5c3c29dabd50a9d0488c97c48 ]
Currently wrong bit depth is exposed in hw params, causing clipped
volume during playback. Expose correct parameters.
Fixes: a126750fc865 ("ASoC: Intel: catpt: PCM operations")
Reported-by: Andy Shevchenko <andriy.shevchenko@intel.com>
Tested-by: Andy Shevchenko <andriy.shevchenko@intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Message-ID: <20250909092829.375953-1-amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 35fc531a59694f24a2456569cf7d1a9c6436841c ]
The dev_err message is reporting an error about capture streams however
it is using the incorrect variable num_playback instead of num_capture.
Fix this by using the correct variable num_capture.
Fixes: a1d1e266b445 ("ASoC: SOF: Intel: Add Intel specific HDA stream operations")
Signed-off-by: Colin Ian King <colin.i.king@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://patch.msgid.link/20250902120639.2626861-1-colin.i.king@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 9b17d3724df55ecc2bc67978822585f2b023be48 ]
Using a single value of 22500000 for both 48000Hz and 44100Hz audio
will sometimes result in returning wrong dividers due to rounding.
Update the code to use the actual value for both.
Fixes: 51b2bb3f2568 ("ASoC: wm8974: configure pll and mclk divider automatically")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://patch.msgid.link/20250821082639.1301453-4-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit b4799520dcd6fe1e14495cecbbe9975d847cd482 ]
Fixes: 0b5e92c5e020 ("ASoC WM8940 Driver")
Reported-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Link: https://patch.msgid.link/20250821082639.1301453-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit d05afb53c683ef7ed1228b593c3360f4d3126c58 ]
Using a single value of 22500000 for both 48000Hz and 44100Hz audio
will sometimes result in returning wrong dividers due to rounding.
Update the code to use the actual value for both.
Fixes: 294833fc9eb4 ("ASoC: wm8940: Rewrite code to set proper clocks")
Reported-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Ankur Tyagi <ankur.tyagi85@gmail.com>
Link: https://patch.msgid.link/20250821082639.1301453-2-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
commit d33c3471047fc54966621d19329e6a23ebc8ec50 upstream.
This laptop uses the ALC236 codec with COEF 0x7 and idx 1 to
control the mute LED. Enable the existing quirk for this device.
Signed-off-by: Praful Adiga <praful.adiga@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 33b55b94bca904ca25a9585e3cd43d15f0467969 upstream.
The q6i2s_set_fmt() function was defined but never linked into the
I2S DAI operations, resulting DAI format settings is being ignored
during stream setup. This change fixes the issue by properly linking
the .set_fmt handler within the DAI ops.
Fixes: 30ad723b93ade ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Cc: stable@vger.kernel.org
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>
Message-ID: <20250908053631.70978-3-mohammad.rafi.shaik@oss.qualcomm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
failed
commit 68f27f7c7708183e7873c585ded2f1b057ac5b97 upstream.
If earlier opening of source graph fails (e.g. ADSP rejects due to
incorrect audioreach topology), the graph is closed and
"dai_data->graph[dai->id]" is assigned NULL. Preparing the DAI for sink
graph continues though and next call to q6apm_lpass_dai_prepare()
receives dai_data->graph[dai->id]=NULL leading to NULL pointer
exception:
qcom-apm gprsvc:service:2:1: Error (1) Processing 0x01001002 cmd
qcom-apm gprsvc:service:2:1: DSP returned error[1001002] 1
q6apm-lpass-dais 30000000.remoteproc:glink-edge:gpr:service@1:bedais: fail to start APM port 78
q6apm-lpass-dais 30000000.remoteproc:glink-edge:gpr:service@1:bedais: ASoC: error at snd_soc_pcm_dai_prepare on TX_CODEC_DMA_TX_3: -22
Unable to handle kernel NULL pointer dereference at virtual address 00000000000000a8
...
Call trace:
q6apm_graph_media_format_pcm+0x48/0x120 (P)
q6apm_lpass_dai_prepare+0x110/0x1b4
snd_soc_pcm_dai_prepare+0x74/0x108
__soc_pcm_prepare+0x44/0x160
dpcm_be_dai_prepare+0x124/0x1c0
Fixes: 30ad723b93ad ("ASoC: qdsp6: audioreach: add q6apm lpass dai support")
Cc: stable@vger.kernel.org
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Message-ID: <20250904101849.121503-2-krzysztof.kozlowski@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 5f1af203ef964e7f7bf9d32716dfa5f332cc6f09 upstream.
Fix missing lpaif_type configuration for the I2S interface.
The proper lpaif interface type required to allow DSP to vote
appropriate clock setting for I2S interface.
Fixes: 25ab80db6b133 ("ASoC: qdsp6: audioreach: add module configuration command helpers")
Cc: stable@vger.kernel.org
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Mohammad Rafi Shaik <mohammad.rafi.shaik@oss.qualcomm.com>
Message-ID: <20250908053631.70978-2-mohammad.rafi.shaik@oss.qualcomm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
supported
[ Upstream commit aea3493246c474bc917d124d6fb627663ab6bef0 ]
The ALSA HwDep character device of the firewire-motu driver incorrectly
returns EPOLLOUT in poll(2), even though the driver implements no operation
for write(2). This misleads userspace applications to believe write() is
allowed, potentially resulting in unnecessarily wakeups.
This issue dates back to the driver's initial code added by a commit
71c3797779d3 ("ALSA: firewire-motu: add hwdep interface"), and persisted
when POLLOUT was updated to EPOLLOUT by a commit a9a08845e9ac ('vfs: do
bulk POLL* -> EPOLL* replacement("").').
This commit fixes the bug.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://patch.msgid.link/20250829233749.366222-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 829ee558f3527fd602c6e2e9f270959d1de09fe0 ]
ASUS VivoBook X515UA with PCI SSID 1043:106f had a default quirk
pickup via pin table that applies ALC256_FIXUP_ASUS_MIC, but this adds
a bogus built-in mic pin 0x13 enabled. This was no big problem
because the pin 0x13 was assigned as the secondary mic, but the recent
fix made the entries sorted, hence this bogus pin appeared now as the
primary input and it broke.
For fixing the bug, put the right quirk entry for this device pointing
to ALC256_FIXUP_ASUS_MIC_NO_PRESENCE.
Fixes: 3b4309546b48 ("ALSA: hda: Fix headset detection failure due to unstable sort")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=219897
Link: https://patch.msgid.link/20250324153233.21195-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
commit 051b02b17a8b383ee033db211f90f24b91ac7006 upstream.
Add a PCI quirk to enable microphone detection on the headphone jack of
TongFang X6AR5xxY and X6FR5xxY devices.
Signed-off-by: Aaron Erhardt <aer@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250826141054.1201482-1-aer@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit bcd6659d4911c528381531472a0cefbd4003e29e upstream.
It was reported that HP EliteDesk 800 G4 DM 65W (SSID 103c:845a) needs
the similar quirk for enabling HDMI outputs, too. This patch adds the
corresponding quirk entry.
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250901115009.27498-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 9c6182843b0d02ca04cc1d946954a65a2286c7db upstream.
Applying the quirk of that, the lowest Playback mixer volume setting
mutes the audio output, on more devices.
Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/merge_requests/2514
Cc: <stable@vger.kernel.org>
Tested-by: Guoli An <anguoli@uniontech.com>
Signed-off-by: Cryolitia PukNgae <cryolitia@uniontech.com>
Link: https://patch.msgid.link/20250822-mixer-quirk-v1-1-b19252239c1c@uniontech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
[ Upstream commit 43e0da37d5cfb23eec6aeee9422f84d86621ce2b ]
We already have a component driver named "RX-MACRO", which is
lpass-rx-macro.c. The tx macro component driver's name should
be "TX-MACRO" accordingly. Fix it.
Cc: Srinivas Kandagatla <srini@kernel.org>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Reviewed-by: Neil Armstrong <neil.armstrong@linaro.org>
Link: https://patch.msgid.link/20250806140030.691477-1-alexey.klimov@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 8410fe81093ff231e964891e215b624dabb734b0 ]
The entry of the validators table for UAC3 feature unit is defined
with a wrong sub-type UAC_FEATURE (= 0x06) while it should have been
UAC3_FEATURE (= 0x07). This patch corrects the entry value.
Fixes: 57f8770620e9 ("ALSA: usb-audio: More validations of descriptor units")
Link: https://patch.msgid.link/20250821150835.8894-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 5003a65790ed66be882d1987cc2ca86af0de3db1 ]
In the snd_utimer_create() function, if the kasprintf() function return
NULL, snd_utimer_put_id() will be called, finally use ida_free()
to free the unallocated id 0.
the syzkaller reported the following information:
------------[ cut here ]------------
ida_free called for id=0 which is not allocated.
WARNING: CPU: 1 PID: 1286 at lib/idr.c:592 ida_free+0x1fd/0x2f0 lib/idr.c:592
Modules linked in:
CPU: 1 UID: 0 PID: 1286 Comm: syz-executor164 Not tainted 6.15.8 #3 PREEMPT(lazy)
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.16.3-4.fc42 04/01/2014
RIP: 0010:ida_free+0x1fd/0x2f0 lib/idr.c:592
Code: f8 fc 41 83 fc 3e 76 69 e8 70 b2 f8 (...)
RSP: 0018:ffffc900007f79c8 EFLAGS: 00010282
RAX: 0000000000000000 RBX: 1ffff920000fef3b RCX: ffffffff872176a5
RDX: ffff88800369d200 RSI: 0000000000000000 RDI: ffff88800369d200
RBP: 0000000000000000 R08: ffffffff87ba60a5 R09: 0000000000000000
R10: 0000000000000001 R11: 0000000000000000 R12: 0000000000000000
R13: 0000000000000002 R14: 0000000000000000 R15: 0000000000000000
FS: 00007f6f1abc1740(0000) GS:ffff8880d76a0000(0000) knlGS:0000000000000000
CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033
CR2: 00007f6f1ad7a784 CR3: 000000007a6e2000 CR4: 00000000000006f0
Call Trace:
<TASK>
snd_utimer_put_id sound/core/timer.c:2043 [inline] [snd_timer]
snd_utimer_create+0x59b/0x6a0 sound/core/timer.c:2184 [snd_timer]
snd_utimer_ioctl_create sound/core/timer.c:2202 [inline] [snd_timer]
__snd_timer_user_ioctl.isra.0+0x724/0x1340 sound/core/timer.c:2287 [snd_timer]
snd_timer_user_ioctl+0x75/0xc0 sound/core/timer.c:2298 [snd_timer]
vfs_ioctl fs/ioctl.c:51 [inline]
__do_sys_ioctl fs/ioctl.c:907 [inline]
__se_sys_ioctl fs/ioctl.c:893 [inline]
__x64_sys_ioctl+0x198/0x200 fs/ioctl.c:893
do_syscall_x64 arch/x86/entry/syscall_64.c:63 [inline]
do_syscall_64+0x7b/0x160 arch/x86/entry/syscall_64.c:94
entry_SYSCALL_64_after_hwframe+0x76/0x7e
[...]
The utimer->id should be set properly before the kasprintf() function,
ensures the snd_utimer_put_id() function will free the allocated id.
Fixes: 37745918e0e75 ("ALSA: timer: Introduce virtual userspace-driven timers")
Signed-off-by: Dewei Meng <mengdewei@cqsoftware.com.cn>
Link: https://patch.msgid.link/20250821014317.40786-1-mengdewei@cqsoftware.com.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 89f0addeee3cb2dc49837599330ed9c4612f05b0 ]
The "p" pointer is void so sizeof(*p) is 1. The intent was to check
sizeof(*cs_desc), which is 3, instead.
Fixes: ecfd41166b72 ("ALSA: usb-audio: Validate UAC3 cluster segment descriptors")
Signed-off-by: Dan Carpenter <dan.carpenter@linaro.org>
Link: https://patch.msgid.link/aKL5kftC1qGt6lpv@stanley.mountain
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
commit eafae0fdd115a71b3a200ef1a31f86da04bac77f upstream.
The HP EliteBook x360 830 G6 and HP EliteBook 830 G6 have
Realtek HDA codec ALC215. It needs the ALC285_FIXUP_HP_GPIO_LED
quirk to enable the mute LED.
Cc: <stable@vger.kernel.org>
Signed-off-by: Evgeniy Harchenko <evgeniyharchenko.dev@gmail.com>
Link: https://patch.msgid.link/20250815095814.75845-1-evgeniyharchenko.dev@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit eb3bb145280b6c857a748731a229698e4a7cf37b upstream.
Replace GFP_ATOMIC with GFP_KERNEL for dma_alloc_coherent() calls. This
change improves memory allocation reliability during firmware loading,
particularly during system resume when memory pressure is high. Because
of using GFP_KERNEL, reclaim can happen which can reduce the probability
of failure.
Fixes memory allocation failures observed during system resume with
fragmented memory conditions.
snd_sof_amd_vangogh 0000:04:00.5: error: failed to load DSP firmware after resume -12
Fixes: 145d7e5ae8f4e ("ASoC: SOF: amd: add option to use sram for data bin loading")
Fixes: 7e51a9e38ab20 ("ASoC: SOF: amd: Add fw loader and renoir dsp ops to load firmware")
Cc: stable@vger.kernel.org
Signed-off-by: Muhammad Usama Anjum <usama.anjum@collabora.com>
Link: https://patch.msgid.link/20250725190254.1081184-1-usama.anjum@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
[ Upstream commit 0e270f32975fd21874185ba53653630dd40bf560 ]
Use the regmap_write() for software reset in fsl_sai_config_disable would
cause the FSL_SAI_CSR_BCE bit to be cleared. Refer to
commit 197c53c8ecb34 ("ASoC: fsl_sai: Don't disable bitclock for i.MX8MP")
FSL_SAI_CSR_BCE should not be cleared. So need to use regmap_update_bits()
instead of regmap_write() for these bit operations.
Fixes: dc78f7e59169d ("ASoC: fsl_sai: Force a software reset when starting in consumer mode")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250807020318.2143219-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 11f74f48c14c1f4fe16541900ea5944c42e30ccf ]
If pcim_request_all_regions() fails, error path operates on
uninitialized 'bus' pointer. Found out by Coverity static analyzer.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250730124906.351798-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit dbe05428c4e54068a86e7e02405f3b30b1d2b3dd ]
Several months ago, Joshua Grisham submitted a patch [1]
for several ALC298 based sound cards.
The entry for the LG gram 16 in the alc269_fixup_tbl only matches the
Subsystem ID for the 16Z90R-Q and 16Z90R-K models [2]. My 16Z90R-A has a
different Subsystem ID [3]. I'm not sure why these IDs differ, but I
speculate it's due to the NVIDIA GPU included in the 16Z90R-A model that
isn't present in the other models.
I applied the patch to the latest Arch Linux kernel and the card was
initialized as expected.
[1]: https://lore.kernel.org/linux-sound/20240909193000.838815-1-josh@joshuagrisham.com/
[2]: https://linux-hardware.org/?id=pci:8086-51ca-1854-0488
[3]: https://linux-hardware.org/?id=pci:8086-51ca-1854-0489
Signed-off-by: Thomas Croft <thomasmcft@gmail.com>
Link: https://patch.msgid.link/20250804151457.134761-2-thomasmcft@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 8167f4f42572818fa8153be2b03e4c2120846603 ]
Qcom lpass is using component->id to keep DAI ID (A).
(S) static int lpass_platform_pcmops_open(
sruct snd_soc_component *component,
struct snd_pcm_substream *substream)
{ ^^^^^^^^^(B0)
...
(B1) struct snd_soc_pcm_runtime *soc_runtime = snd_soc_substream_to_rtd(substream);
(B2) struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_runtime, 0);
...
(B3) unsigned int dai_id = cpu_dai->driver->id;
(A) component->id = dai_id;
...
}
This driver can get dai_id from substream (B0 - B3).
In this driver, below functions get dai_id from component->id (A).
(X) lpass_platform_pcmops_suspend()
(Y) lpass_platform_pcmops_resume()
(Z) lpass_platform_copy()
Here, (Z) can get it from substream (B0 - B3), don't need to use
component->id (A). On suspend/resume (X)(Y), dai_id can only be obtained
from component->id (A), because there is no substream (B0) in function
parameter.
But, component->id (A) itself should not be used for such purpose.
It is intilialized at snd_soc_component_initialize(), and parsed its ID
(= component->id) from device name (a).
int snd_soc_component_initialize(...)
{
...
if (!component->name) {
(a) component->name = fmt_single_name(dev, &component->id);
... ^^^^^^^^^^^^^
}
...
}
Unfortunately, current code is broken to start with.
There are many regmaps that the driver cares about, however its only
managing one (either dp or i2s) in component suspend/resume path.
I2S regmap is mandatory however other regmaps are setup based on flags
like "hdmi_port_enable" and "codec_dma_enable".
Correct thing for suspend/resume path to handle is by checking these
flags, instead of using component->id.
Signed-off-by: Srinivas Kandagatla <srini@kernel.org>
Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87a56ouuob.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 19f971057b2d7b99c80530ec1052b45de236a8da ]
To be more resilient to codec-detection failures when the hardware
powers on slowly, add retry mechanism to the device verification check.
Similar pattern is found throughout a number of Realtek codecs. Our
tests show that 60ms delay is sufficient to address readiness issues on
rt5640 chip.
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Xinxin Wan <xinxin.wan@intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://patch.msgid.link/20250530142120.2944095-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit fd3ab72e42e9871a9902b945a2bf8bb87b49c718 ]
Fix all macro related issues identified by checkpatch.pl:
CHECK: Macro argument 'x' may be better as '(x)' to avoid precedence issues
Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250526-dualsense-alsa-jack-v1-3-1a821463b632@collabora.com
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 92f59aeb13252265c20e7aef1379a8080c57e0a2 ]
At the time being recalculate_boundary() is implemented with a
loop which shows up as costly in a perf profile, as depicted by
the annotate below:
0.00 : c057e934: 3d 40 7f ff lis r10,32767
0.03 : c057e938: 61 4a ff ff ori r10,r10,65535
0.21 : c057e93c: 7d 49 50 50 subf r10,r9,r10
5.39 : c057e940: 7d 3c 4b 78 mr r28,r9
2.11 : c057e944: 55 29 08 3c slwi r9,r9,1
3.04 : c057e948: 7c 09 50 40 cmplw r9,r10
2.47 : c057e94c: 40 81 ff f4 ble c057e940 <snd_pcm_ioctl+0xee0>
Total: 13.2% on that simple loop.
But what the loop does is to multiply the boundary by 2 until it is
over the wanted border. This can be avoided by using fls() to get the
boundary value order and shift it by the appropriate number of bits at
once.
This change provides the following profile:
0.04 : c057f6e8: 3d 20 7f ff lis r9,32767
0.02 : c057f6ec: 61 29 ff ff ori r9,r9,65535
0.34 : c057f6f0: 7d 5a 48 50 subf r10,r26,r9
0.23 : c057f6f4: 7c 1a 50 40 cmplw r26,r10
0.02 : c057f6f8: 41 81 00 20 bgt c057f718 <snd_pcm_ioctl+0xf08>
0.26 : c057f6fc: 7f 47 00 34 cntlzw r7,r26
0.09 : c057f700: 7d 48 00 34 cntlzw r8,r10
0.22 : c057f704: 7d 08 38 50 subf r8,r8,r7
0.04 : c057f708: 7f 5a 40 30 slw r26,r26,r8
0.35 : c057f70c: 7c 0a d0 40 cmplw r10,r26
0.13 : c057f710: 40 80 05 f8 bge c057fd08 <snd_pcm_ioctl+0x14f8>
0.00 : c057f714: 57 5a f8 7e srwi r26,r26,1
Total: 1.7% with that loopless alternative.
Signed-off-by: Christophe Leroy <christophe.leroy@csgroup.eu>
Link: https://patch.msgid.link/4836e2cde653eebaf2709ebe30eec736bb8c67fd.1749202237.git.christophe.leroy@csgroup.eu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit a409c60111e6bb98fcabab2aeaa069daa9434ca0 ]
The 'sprintf' call in 'add_tuning_control' may exceed the 44-byte
buffer if either string argument is too long. This triggers a compiler
warning.
Replaced 'sprintf' with 'snprintf' to limit string lengths to prevent
overflow.
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202506100642.95jpuMY1-lkp@intel.com/
Signed-off-by: Lucy Thrun <lucy.thrun@digital-rabbithole.de>
Link: https://patch.msgid.link/20250610175012.918-3-lucy.thrun@digital-rabbithole.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 2d91cb261cac6d885954b8f5da28b5c176c18131 ]
snd_soc_remove_pcm_runtime() might be called with rtd == NULL which will
leads to null pointer dereference.
This was reproduced with topology loading and marking a link as ignore
due to missing hardware component on the system.
On module removal the soc_tplg_remove_link() would call
snd_soc_remove_pcm_runtime() with rtd == NULL since the link was ignored,
no runtime was created.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://patch.msgid.link/20250619084222.559-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 6b3cb7f4341cbf62d41ccf6ea906dbe66be8aa3d ]
Parsing the dapm_widget_tokens is also needed for DSPless mode as it is
setting the snd_soc_dapm_widget.no_wname_in_kcontrol_name flag for the
kcontrol creation from DAPM widgets.
Without that flag set, the following warnings might appear because of long
control names:
ALSA: Control name 'eqiir.2.1 Post Mixer Analog Playback IIR Eq bytes' truncated to 'eqiir.2.1 Post Mixer Analog Playback IIR Eq'
ALSA: Control name 'eqfir.2.1 Post Mixer Analog Playback FIR Eq bytes' truncated to 'eqfir.2.1 Post Mixer Analog Playback FIR Eq'
ALSA: Control name 'drc.2.1 Post Mixer Analog Playback DRC bytes' truncated to 'drc.2.1 Post Mixer Analog Playback DRC byte'
ALSA: Control name 'drc.2.1 Post Mixer Analog Playback DRC switch' truncated to 'drc.2.1 Post Mixer Analog Playback DRC swit'
ALSA: Control name 'gain.15.1 Pre Mixer Deepbuffer HDA Analog Volume' truncated to 'gain.15.1 Pre Mixer Deepbuffer HDA Analog V'
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://patch.msgid.link/20250619102640.12068-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 87aafc8580acf87fcaf1a7e30ed858d8c8d37d81 ]
code mistakenly used a hardcoded index (codec[1]) instead of
iterating, over the codec array using the loop variable i.
Use codec[i] instead of codec[1] to match the loop iteration.
Signed-off-by: Alok Tiwari <alok.a.tiwari@oracle.com>
Link: https://patch.msgid.link/20250621185233.4081094-1-alok.a.tiwari@oracle.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit c4ca928a6db1593802cd945f075a7e21dd0430c1 ]
We currently log parse failures for ELD data and some disconnection events
as errors without rate limiting. These log messages can be triggered very
frequently in some situations, especially ELD parsing when there is nothing
connected to a HDMI port which will generate:
hdmi-audio-codec hdmi-audio-codec.1.auto: HDMI: Unknown ELD version 0
While there's doubtless work that could be done on reducing the number of
connection notification callbacks it's possible these may be legitimately
generated by poor quality physical connections so let's use rate limiting
to mitigate the log spam for the parse errors and lower the severity for
disconnect logging to debug level.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://patch.msgid.link/20250613-asoc-hdmi-eld-logging-v1-1-76d64154d969@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 1adcbdf54f76e1004bdf71df4eb1888c26e7ad06 ]
Although the jack polling is canceled at shutdown in
snd_hda_codec_shutdown(), it might be still re-triggered when the work
is being processed at cancel_delayed_work_sync() call. This may
result in the unexpected hardware access that should have been already
disabled.
For assuring to stop the jack polling, clear codec->jackpoll_interval
at shutdown.
Reported-by: Joakim Zhang <joakim.zhang@cixtech.com>
Closes: https://lore.kernel.org/20250619020844.2974160-4-joakim.zhang@cixtech.com
Tested-by: Joakim Zhang <joakim.zhang@cixtech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250623131437.10670-2-tiwai@suse.de
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 5f7e54b23e4d253eff3b10b12d6fa92d28d7dddc ]
We used to call directly hda_jackpoll_work() from a couple of places
for updating the jack and notify to user-space, but this makes rather
the code flow fragile. Namely, because of those direct calls,
hda_jackpoll_work() uses snd_hda_power_up_pm() and *_down_pm() calls
instead of the standard snd_hda_power_up() and *_down() calls. The
latter pair assures the runtime PM resume sync, so it can avoid the
race against the PM callbacks gracefully, while the former pair may
continue if called concurrently, hence it may race (by design).
In this patch, we change the call pattern of hda_jackpoll_work(); now
all callers are replaced with the standard snd_hda_jack_report_sync()
and the additional schedule_delayed_work().
Since hda_jackpoll_work() is called only from the associated work,
it's always outside the PM code path, and we can safely use
snd_hda_power_up() and *_down() there instead. This allows us to
remove the racy check of power-state in hda_jackpoll_work(), as well
as the tricky cancel_delayed_work() and rescheduling at
hda_codec_runtime_suspend().
Reported-by: Joakim Zhang <joakim.zhang@cixtech.com>
Closes: https://lore.kernel.org/20250619020844.2974160-1-joakim.zhang@cixtech.com
Tested-by: Joakim Zhang <joakim.zhang@cixtech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250623131437.10670-4-tiwai@suse.de
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit f40ecc2743652c0b0f19935f81baf57c601eb7f0 ]
ASoC has 2 functions to set bias level.
(A) snd_soc_dapm_force_bias_level()
(B) snd_soc_dapm_set_bias_level()
snd_soc_dapm_force_bias_level() (A) will set dapm->bias_level (a) if
successed.
(A) int snd_soc_dapm_force_bias_level(...)
{
...
if (ret == 0)
(a) dapm->bias_level = level;
...
}
snd_soc_dapm_set_bias_level() (B) is also a function that sets bias_level.
It will call snd_soc_dapm_force_bias_level() (A) inside, but doesn't
set dapm->bias_level by itself. One note is that (A) might not be called.
(B) static int snd_soc_dapm_set_bias_level(...)
{
...
ret = snd_soc_card_set_bias_level(...);
...
if (dapm != &card->dapm)
(A) ret = snd_soc_dapm_force_bias_level(...);
...
ret = snd_soc_card_set_bias_level_post(...);
...
}
dapm->bias_level will be set if (A) was called, but might not be set
if (B) was called, even though it calles set_bias_level() function.
We should set dapm->bias_level if we calls
snd_soc_dapm_set_bias_level() (B), too.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/87qzyn4g4h.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
commit 0db77eccd964b11ab2b757031d1354fcc5a025ea upstream.
Framework Laptop 13 (AMD Ryzen AI 300) requires the same quirk for
headset detection as other Framework 13 models.
Signed-off-by: Christopher Eby <kreed@kreed.org>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250810030006.9060-1-kreed@kreed.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit b26e2afb3834d4a61ce54c8484ff6014bef0b4b7 upstream.
Add a PCI quirk to enable microphone input on the headphone jack on
the HONOR BRB-X M1010 laptop.
Signed-off-by: Vasiliy Kovalev <kovalev@altlinux.org>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250811132716.45076-1-kovalev@altlinux.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit ecfd41166b72b67d3bdeb88d224ff445f6163869 upstream.
UAC3 class segment descriptors need to be verified whether their sizes
match with the declared lengths and whether they fit with the
allocated buffer sizes, too. Otherwise malicious firmware may lead to
the unexpected OOB accesses.
Fixes: 11785ef53228 ("ALSA: usb-audio: Initial Power Domain support")
Reported-and-tested-by: Youngjun Lee <yjjuny.lee@samsung.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250814081245.8902-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit d832ccbc301fbd9e5a1d691bdcf461cdb514595f upstream.
UAC3 power domain descriptors need to be verified with its variable
bLength for avoiding the unexpected OOB accesses by malicious
firmware, too.
Fixes: 9a2fe9b801f5 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Reported-and-tested-by: Youngjun Lee <yjjuny.lee@samsung.com>
Cc: <stable@vger.kernel.org>
Link: https://patch.msgid.link/20250814081245.8902-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit a9dec0963187d05725369156a5e0e14cd3487bfb upstream.
My friend have Victus 16-d1xxx with board ID 8A26, the existing quirk
for Victus 16-d1xxx wasn't working because of different board ID
Tested on Victus 16-d1015nt Laptop. The LED behaviour works
as intended.
Cc: <stable@vger.kernel.org>
Signed-off-by: Edip Hazuri <edip@medip.dev>
Link: https://patch.msgid.link/20250729181848.24432-4-edip@medip.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 956048a3cd9d2575032e2c7ca62803677357ae18 upstream.
The mute led on this laptop is using ALC245 but requires a quirk to work
This patch enables the existing quirk for the device.
Tested on Victus 16-S0063NT Laptop. The LED behaviour works
as intended.
Cc: <stable@vger.kernel.org>
Signed-off-by: Edip Hazuri <edip@medip.dev>
Link: https://patch.msgid.link/20250729181848.24432-2-edip@medip.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit bd7814a4c0fd883894bdf9fe5eda24c9df826e4c upstream.
The mute led on this laptop is using ALC245 but requires a quirk to work
This patch enables the existing quirk for the device.
Tested on Victus 16-r1xxx Laptop. The LED behaviour works
as intended.
Cc: <stable@vger.kernel.org>
Signed-off-by: Edip Hazuri <edip@medip.dev>
Link: https://patch.msgid.link/20250725151436.51543-2-edip@medip.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 8a15ca0ca51399b652b1bbb23b590b220cf03d62 upstream.
During communication with Focusrite Scarlett Gen 2/3/4 USB audio
interfaces, -EPROTO is sometimes returned from scarlett2_usb_tx(),
snd_usb_ctl_msg() which can cause initialisation and control
operations to fail intermittently.
This patch adds up to 5 retries in scarlett2_usb(), with a delay
starting at 5ms and doubling each time. This follows the same approach
as the fix for usb_set_interface() in endpoint.c (commit f406005e162b
("ALSA: usb-audio: Add retry on -EPROTO from usb_set_interface()")),
which resolved similar -EPROTO issues during device initialisation,
and is the same approach as in fcp.c:fcp_usb().
Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Closes: https://github.com/geoffreybennett/linux-fcp/issues/41
Cc: stable@vger.kernel.org
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://patch.msgid.link/aIdDO6ld50WQwNim@m.b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
commit 8cbe564974248ee980562be02f2b1912769562c7 upstream.
In __hdmi_lpe_audio_probe(), strscpy() is incorrectly called with the
length of the source string (excluding the NUL terminator) rather than
the size of the destination buffer. This results in one character less
being copied from 'card->shortname' to 'pcm->name'.
Use the destination buffer size instead to ensure the card name is
copied correctly.
Cc: stable@vger.kernel.org
Fixes: 75b1a8f9d62e ("ALSA: Convert strlcpy to strscpy when return value is unused")
Signed-off-by: Thorsten Blum <thorsten.blum@linux.dev>
Link: https://patch.msgid.link/20250805234156.60294-1-thorsten.blum@linux.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
|
|
[ Upstream commit 9f320dfb0ffc555aa2eac8331dee0c2c16f67633 ]
There are a couple of cases where the error is ignored or the error
code isn't propagated in ca0132_alt_select_out(). Fix those.
Fixes: def3f0a5c700 ("ALSA: hda/ca0132 - Add quirk output selection structures.")
Link: https://patch.msgid.link/20250806094423.8843-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit ca592e20659e0304ebd8f4dabb273da4f9385848 ]
There is no PHY for the XCVR module on i.MX93, the channel status needs
to be obtained from FSL_XCVR_RX_CS_DATA_* registers. And channel status
acknowledge (CSA) bit should be set once channel status is processed.
Fixes: e240b9329a30 ("ASoC: fsl_xcvr: Add support for i.MX93 platform")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://patch.msgid.link/20250710030405.3370671-2-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit ec4a10ca4a68ec97f12f4d17d7abb74db34987db ]
In commit 32c9c06adb5b ("ASoC: mediatek: disable buffer pre-allocation")
buffer pre-allocation was disabled to accommodate newer platforms that
have a limited reserved memory region for the audio frontend.
Turns out disabling pre-allocation across the board impacts platforms
that don't have this reserved memory region. Buffer allocation failures
have been observed on MT8173 and MT8183 based Chromebooks under low
memory conditions, which results in no audio playback for the user.
Since some MediaTek platforms already have dedicated reserved memory
pools for the audio frontend, the plan is to enable this for all of
them. This requires device tree changes. As a fallback, reinstate the
original policy of pre-allocating audio buffers at probe time of the
reserved memory pool cannot be found or used.
This patch covers the MT8173, MT8183, MT8186 and MT8192 platforms for
now, the reason being that existing MediaTek platform drivers that
supported reserved memory were all platforms that mainly supported
ChromeOS, and is also the set of devices that I can verify.
Fixes: 32c9c06adb5b ("ASoC: mediatek: disable buffer pre-allocation")
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Signed-off-by: Chen-Yu Tsai <wenst@chromium.org>
Link: https://patch.msgid.link/20250612074901.4023253-7-wenst@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 7e10d7242ea8a5947878880b912ffa5806520705 ]
This structure is really too larget to be allocated on the stack:
sound/soc/soc-ops.c:435:5: error: stack frame size (1296) exceeds limit (1280) in 'snd_soc_limit_volume' [-Werror,-Wframe-larger-than]
Change the function to dynamically allocate it instead.
There is probably a better way to do it since only two integer fields
inside of that structure are actually used, but this is the simplest
rework for the moment.
Fixes: 783db6851c18 ("ASoC: ops: Enforce platform maximum on initial value")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://patch.msgid.link/20250610093057.2643233-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit f4c77d5af0a9cd0ee22617baa8b49d0e151fbda7 ]
commit 7f1186a8d738661 ("ASoC: soc-dai: check return value at
snd_soc_dai_set_tdm_slot()") checks return value of
xlate_tdm_slot_mask() (A1)(A2).
/*
* ...
(Y) * TDM mode can be disabled by passing 0 for @slots. In this case @tx_mask,
* @rx_mask and @slot_width will be ignored.
* ...
*/
int snd_soc_dai_set_tdm_slot(...)
{
...
if (...)
(A1) ret = dai->driver->ops->xlate_tdm_slot_mask(...);
else
(A2) ret = snd_soc_xlate_tdm_slot_mask(...);
if (ret)
goto err;
...
}
snd_soc_xlate_tdm_slot_mask() (A2) will return -EINVAL if slots was 0 (X),
but snd_soc_dai_set_tdm_slot() allow to use it (Y).
(A) static int snd_soc_xlate_tdm_slot_mask(...)
{
...
if (!slots)
(X) return -EINVAL;
...
}
Call xlate_tdm_slot_mask() only if slots was non zero.
Reported-by: Giedrius Trainavičius <giedrius@blokas.io>
Closes: https://lore.kernel.org/r/CAMONXLtSL7iKyvH6w=CzPTxQdBECf++hn8RKL6Y4=M_ou2YHow@mail.gmail.com
Fixes: 7f1186a8d738661 ("ASoC: soc-dai: check return value at snd_soc_dai_set_tdm_slot()")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://patch.msgid.link/8734cdfx59.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|
|
[ Upstream commit 6f80be548588429100eb1f5e25dc2a714d583ffe ]
add DMI entry for ASUS Vivobook PRO 15X (M6501RM)
to make the internal microphone function
Signed-off-by: Alexandru Andries <alex.andries.aa@gmail.com>
Link: https://patch.msgid.link/20250707220730.361290-1-alex.andries.aa@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
|