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commit b5a236c175b0d984552a5f7c9d35141024c2b261 upstream.
Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d06 ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 98081ca62cbac31fb0f7efaf90b2e7384ce22257 upstream.
Currently we deal with single codec and suspend codec callbacks for
all S3, S4 and runtime PM handling. But it turned out that we want
distinguish the call patterns sometimes, e.g. for applying some init
sequence only at probing and restoring from hibernate.
This patch slightly modifies the common PM callbacks for HD-audio
codec and stores the currently processed PM event in power_state of
the codec's device.power field, which is currently unused. The codec
callback can take a look at this event value and judges which purpose
it's being called.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 31d2350d602511efc9ef626b848fe521233b0387 upstream.
ac97_of_get_child_device() take the refcount of the node explicitly
via of_node_get(), but this leads to an unbalance. The
for_each_child_of_node() loop itself takes the refcount for each
iteration node, hence you don't need to take the extra refcount
again.
Fixes: 2225a3e6af78 ("ALSA: ac97: add codecs devicetree binding")
Reviewed-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 744c67ffeb06f2d2493f4049ba0bd19698ce0adf upstream.
The commit 3baffc4a84d7 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84d7 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2d012c65a9ca26a0ef87ea0a42f1653dd37155f5 upstream.
Current ALSA firewire-motu driver uses the value of 'model' field
of unit directory in configuration ROM for modalias for MOTU
FireWire models. However, as long as I checked, Pre8 and
828mk3(Hybrid) have the same value for the field (=0x100800).
unit | version | model
--------------- | --------- | ----------
828mkII | 0x000003 | 0x101800
Traveler | 0x000009 | 0x107800
Pre8 | 0x00000f | 0x100800 <-
828mk3(FW) | 0x000015 | 0x106800
AudioExpress | 0x000033 | 0x104800
828mk3(Hybrid) | 0x000035 | 0x100800 <-
When updating firmware for MOTU 8pre FireWire from v1.0.0 to v1.0.3,
I got change of the value from 0x100800 to 0x103800. On the other
hand, the value of 'version' field is fixed to 0x00000f. As a quick
glance, the higher 12 bits of the value of 'version' field represent
firmware version, while the lower 12 bits is unknown.
By induction, the value of 'version' field represents actual model.
This commit changes modalias to match the value of 'version' field,
instead of 'model' field. For degug, long name of added sound card
includes hexadecimal value of 'model' field.
Fixes: 6c5e1ac0e144 ("ALSA: firewire-motu: add support for Motu Traveler")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v4.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 721f1e6c1fd137e7e2053d8e103b666faaa2d50c upstream.
Another machine which does not like the power saving (noise):
https://bugzilla.redhat.com/show_bug.cgi?id=1689623
Also, reorder the Lenovo C50 entry to keep the table sorted.
Reported-by: hs.guimaraes@outlook.com
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 05bd7fcdd06b19a10f069af1bea3ad9abac038d7 upstream.
The ADCs are sleeping when the SLEEP bit is set and running when it's
cleared, so the bit should be inverted.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit fcf4daabf08079e6d09958a2992e7446ef8d0438 upstream.
According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cc29ea007347f39f4c5a4d27b0b555955a0277f9 upstream.
The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to
wrong register, correct it.
Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Ackedy-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cbc05fd6708c1744ee6a61cb4c461ff956d30524 upstream.
The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC
until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the
internal DMIC uses another module - snd_soc_skl as the driver. We still
need the NID 0x1a in the quirk to enable the headset MIC.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c0ca5eced22215c1e03e3ad479f8fab0bbb30772 upstream.
Dell Precision 5820 with ALC3234 codec (which is equivalent with
ALC255) shows click noises at (runtime) PM resume on the headphone.
The biggest source of the noise comes from the cleared headphone pin
control at resume, which is done via the standard shutup procedure.
Although we have an override of the standard shutup callback to
replace with NOP, this would skip other needed stuff (e.g. the pull
down of headset power). So, instead, this "fixes" the behavior of
alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag.
When this flag is set, Realtek codec won't call the standard
snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this
flag instead of overriding spec->shutup callback itself. This allows
us to apply the similar fix for other entries easily if needed in
future.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 8bb37a2a4d7c02affef554f5dc05f6d2e39c31f9 upstream.
The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone. This issue can be fixed
by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio
jacks of ASUS UX533FD with ALC294.
Besides, ASUS UX362FA and UX533FD have the same audio initial pin config
values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new
SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD.
Fixes: 4e051106730d ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Ming Shuo Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 167897f4b32c2bc18b3b6183029a33fb420a114e upstream.
Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and
HP Z240 models.
Reported-by: Jeff Burrell <jeff.burrell@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cfc35f9c128cea8fce6a5513b1de50d36f3b209f upstream.
I set 10 seconds for the timeout of the i915 audio component binding
with a hope that recent machines are fast enough to handle all probe
tasks in that period, but I was too optimistic. The binding may take
longer than that, and this caused a problem on the machine with both
audio and graphics driver modules loaded in parallel, as Paul Menzel
experienced. This problem haven't hit so often just because the KMS
driver is loaded in initrd on most machines.
As a simple workaround, extend the timeout to 60 seconds.
Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Reported-by: Paul Menzel <pmenzel+alsa-devel@molgen.mpg.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f97a0944a72b26a2bece72516294e112a890f98a upstream.
In data blocks of common isochronous packet for MOTU devices, PCM
frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described
in IEC 61883-6. The frames are not aligned to quadlet.
For capture PCM substream, ALSA firewire-motu driver constructs PCM
frames by reading data blocks byte-by-byte. However this operation
includes bug for lower byte of the PCM sample. This brings invalid
content of the PCM samples.
This commit fixes the bug.
Reported-by: Peter Sjöberg <autopeter@gmail.com>
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 4641c9394010 ("ALSA: firewire-motu: add MOTU specific protocol layer")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Liquid Saffire 56
commit 7dc661bd8d3261053b69e4e2d0050cd1ee540fc1 upstream.
ALSA bebob driver has an entry for Focusrite Saffire Pro 10 I/O. The
entry matches vendor_id in root directory and model_id in unit
directory of configuration ROM for IEEE 1394 bus.
On the other hand, configuration ROM of Focusrite Liquid Saffire 56
has the same vendor_id and model_id. This device is an application of
TCAT Dice (TCD2220 a.k.a Dice Jr.) however ALSA bebob driver can be
bound to it randomly instead of ALSA dice driver. At present, drivers
in ALSA firewire stack can not handle this situation appropriately.
This commit uses more identical mod_alias for Focusrite Saffire Pro 10
I/O in ALSA bebob driver.
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 042a829d bus_info_length 4, crc_length 42, crc 33437
404 31333934 bus_name "1394"
408 f0649222 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100,
max_rec 9 (1024), max_rom 2, gen 2, spd 2 (S400)
40c 00130e01 company_id 00130e |
410 000606e0 device_id 01000606e0 | EUI-64 00130e01000606e0
root directory
-----------------------------------------------------------------
414 0009d31c directory_length 9, crc 54044
418 04000014 hardware version
41c 0c0083c0 node capabilities per IEEE 1394
420 0300130e vendor
424 81000012 --> descriptor leaf at 46c
428 17000006 model
42c 81000016 --> descriptor leaf at 484
430 130120c2 version
434 d1000002 --> unit directory at 43c
438 d4000006 --> dependent info directory at 450
unit directory at 43c
-----------------------------------------------------------------
43c 0004707c directory_length 4, crc 28796
440 1200a02d specifier id: 1394 TA
444 13010001 version: AV/C
448 17000006 model
44c 81000013 --> descriptor leaf at 498
dependent info directory at 450
-----------------------------------------------------------------
450 000637c7 directory_length 6, crc 14279
454 120007f5 specifier id
458 13000001 version
45c 3affffc7 (immediate value)
460 3b100000 (immediate value)
464 3cffffc7 (immediate value)
468 3d600000 (immediate value)
descriptor leaf at 46c
-----------------------------------------------------------------
46c 00056f3b leaf_length 5, crc 28475
470 00000000 textual descriptor
474 00000000 minimal ASCII
478 466f6375 "Focu"
47c 73726974 "srit"
480 65000000 "e"
descriptor leaf at 484
-----------------------------------------------------------------
484 0004a165 leaf_length 4, crc 41317
488 00000000 textual descriptor
48c 00000000 minimal ASCII
490 50726f31 "Pro1"
494 30494f00 "0IO"
descriptor leaf at 498
-----------------------------------------------------------------
498 0004a165 leaf_length 4, crc 41317
49c 00000000 textual descriptor
4a0 00000000 minimal ASCII
4a4 50726f31 "Pro1"
4a8 30494f00 "0IO"
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 040442e4 bus_info_length 4, crc_length 4, crc 17124
404 31333934 bus_name "1394"
408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255,
max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400)
40c 00130e04 company_id 00130e |
410 018001e9 device_id 04018001e9 | EUI-64 00130e04018001e9
root directory
-----------------------------------------------------------------
414 00065612 directory_length 6, crc 22034
418 0300130e vendor
41c 8100000a --> descriptor leaf at 444
420 17000006 model
424 8100000e --> descriptor leaf at 45c
428 0c0087c0 node capabilities per IEEE 1394
42c d1000001 --> unit directory at 430
unit directory at 430
-----------------------------------------------------------------
430 000418a0 directory_length 4, crc 6304
434 1200130e specifier id
438 13000001 version
43c 17000006 model
440 8100000f --> descriptor leaf at 47c
descriptor leaf at 444
-----------------------------------------------------------------
444 00056f3b leaf_length 5, crc 28475
448 00000000 textual descriptor
44c 00000000 minimal ASCII
450 466f6375 "Focu"
454 73726974 "srit"
458 65000000 "e"
descriptor leaf at 45c
-----------------------------------------------------------------
45c 000762c6 leaf_length 7, crc 25286
460 00000000 textual descriptor
464 00000000 minimal ASCII
468 4c495155 "LIQU"
46c 49445f53 "ID_S"
470 41464649 "AFFI"
474 52455f35 "RE_5"
478 36000000 "6"
descriptor leaf at 47c
-----------------------------------------------------------------
47c 000762c6 leaf_length 7, crc 25286
480 00000000 textual descriptor
484 00000000 minimal ASCII
488 4c495155 "LIQU"
48c 49445f53 "ID_S"
490 41464649 "AFFI"
494 52455f35 "RE_5"
498 36000000 "6"
Cc: <stable@vger.kernel.org> # v3.16+
Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few last-minute fixes for 5.0.
The most significant one is the OF-node refcount fix for ASoC
simple-card, which could be triggered on many boards. Another fix for
ASoC core is for the error handling in topology, while others are
device-specific fixes for Samsung and HD-audio"
* tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: simple-card: fixup refcount_t underflow
ASoC: topology: free created components in tplg load error
ALSA: hda/realtek: Disable PC beep in passthrough on alc285
ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5
ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A few small fixes, a driver fix for Samsung, a fix for refcounting of
of_nodes in the simple-card driver that triggered on a lot of systems
and a fix for topology error handling.
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commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card")
merged simple-card and simple-scu-card. Then it had refcount
underflow bug. This patch fixup it.
We will get below error without this patch.
OF: ERROR: Bad of_node_put() on /sound
CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514
Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT)
Workqueue: events deferred_probe_work_func
Call trace:
dump_backtrace+0x0/0x150
show_stack+0x24/0x30
dump_stack+0xb0/0xec
of_node_release+0xd0/0xd8
kobject_put+0x74/0xe8
of_node_put+0x24/0x30
__of_get_next_child+0x50/0x70
of_get_next_child+0x40/0x68
asoc_simple_card_probe+0x604/0x730
platform_drv_probe+0x58/0xa8
...
Reported-by: Vicente Bergas <vicencb@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Topology resources are no longer needed if any element failed to load.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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It is reported that there's a constant background "hum/whitenoise"
in the headset on the Lenovo X1 machines with the codec alc285, and it
is confirmed that if we run the command below, the noise will stop.
sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0
Then I consulted this issue with Kailang, he told me the pin 0x1d on
this codec is used for PC beep in, the noise probably comes from this
pin and we can also disable the PC beep in passthrough, then the PC
beep in will not affect other sound playback.
Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1660581
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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System76 oryp5
On the System76 Oryx Pro (oryp5), there is a headset microphone input
attached to 0x19 that does not have a jack detect. In order to get it
working, the pin configuration needs to be set correctly, and the
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is
similar to the MIC_NO_PRESENCE fixups for some Dell laptops, except we
have a separate microphone jack that is already configured correctly.
Since the ALC1220 does not have a fixup similar to
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, I have exposed the fixup from the
ALC269 in a way that it can be accessed from the
alc1220_fixup_system76_oryp5 function. In addition, the
alc1220_fixup_clevo_p950 needs to be applied to gain speaker output.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"It's a bit of surprising that we've got more changes than hoped at
this late stage, but they all don't look too scary but small fixes.
One change in ALSA core side is again the PCM regression fix that was
partially addressed for OSS, but now the all relevant change is
reverted instead. Also, a few ASoC core fixes for UAF and OOB are
included, while the rest are usual random device-specific fixes"
* tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: pcm: Revert capture stream behavior change in blocking mode
ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
ALSA: hda - Add quirk for HP EliteBook 840 G5
ASoC: samsung: Prevent clk_get_rate() calls in atomic context
ASoC: rsnd: ssiu: correct shift bit for ssiu9
ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
ASoC: topology: fix oops/use-after-free case with dai driver
ASoC: rsnd: fixup MIX kctrl registration
ASoC: core: Allow soc_find_component lookups to match parent of_node
ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter
ASoC: MAINTAINERS: fsl: Change Fabio's email address
ASoC: hdmi-codec: fix oops on re-probe
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Make sure i2s->rclk_srcrate is properly initialized also during
playback through the secondary DAI.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In the commit 62ba568f7aef ("ALSA: pcm: Return 0 when size <
start_threshold in capture"), we changed the behavior of
__snd_pcm_lib_xfer() to return immediately with 0 when a capture
stream has a high start_threshold. This was intended to be a
correction of the behavior consistency and looked harmless, but this
was the culprit of the recent breakage reported by syzkaller, which
was fixed by the commit e190161f96b8 ("ALSA: pcm: Fix tight loop of
OSS capture stream").
At the time for the OSS fix, I didn't touch the behavior for ALSA
native API, as assuming that this behavior actually is good. But this
turned out to be also broken actually for a similar deployment,
e.g. one thread goes to a write loop in blocking mode while another
thread controls the start/stop of the stream manually.
Overall, the original commit is harmful, and it brings less merit to
keep that behavior. Let's revert it.
Fixes: 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture")
Fixes: e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A selection of driver specific fixes here, along with a few core fixes:
- A fixup for some MFD devices that were broken by the previous fixes
for deferred probe.
- A fix for potential out of bounds array accesses when ordering DAPM
power/up down sequences.
- Avoid use after free issue when unloading and reloading drivers using
topologies.
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The commit a60945fd08e4 ("ALSA: usb-audio: move implicit fb quirks to
separate function") introduced an error in the handling of quirks for
implicit feedback endpoints. This commit fixes this.
If a quirk successfully sets up an implicit feedback endpoint, usb-audio
no longer tries to find the implicit fb endpoint itself.
Fixes: a60945fd08e4 ("ALSA: usb-audio: move implicit fb quirks to separate function")
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This enables mute LED support and fixes switching jacks when the laptop
is docked.
Signed-off-by: Jurica Vukadin <jurica.vukadin@rt-rk.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch moves clk_get_rate() call from trigger() to hw_params()
callback to avoid calling sleeping clk API from atomic context
and prevent deadlock as indicated below.
Before this change clk_get_rate() was being called with same
spinlock held as the one passed to the clk API when registering
clocks exposed by the I2S driver.
[ 82.109780] BUG: sleeping function called from invalid context at kernel/locking/mutex.c:908
[ 82.117009] in_atomic(): 1, irqs_disabled(): 128, pid: 1554, name: speaker-test
[ 82.124235] 3 locks held by speaker-test/1554:
[ 82.128653] #0: cc8c5328 (snd_pcm_link_rwlock){...-}, at: snd_pcm_stream_lock_irq+0x20/0x38
[ 82.137058] #1: ec9eda17 (&(&substream->self_group.lock)->rlock){..-.}, at: snd_pcm_ioctl+0x900/0x1268
[ 82.146417] #2: 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4
[ 82.154650] irq event stamp: 8144
[ 82.157949] hardirqs last enabled at (8143): [<c0a0f574>] _raw_read_unlock_irq+0x24/0x5c
[ 82.166089] hardirqs last disabled at (8144): [<c0a0f6a8>] _raw_read_lock_irq+0x18/0x58
[ 82.174063] softirqs last enabled at (8004): [<c01024e4>] __do_softirq+0x3a4/0x66c
[ 82.181688] softirqs last disabled at (7997): [<c012d730>] irq_exit+0x140/0x168
[ 82.188964] Preemption disabled at:
[ 82.188967] [<00000000>] (null)
[ 82.195728] CPU: 6 PID: 1554 Comm: speaker-test Not tainted 5.0.0-rc5-00192-ga6e6caca8f03 #191
[ 82.204302] Hardware name: SAMSUNG EXYNOS (Flattened Device Tree)
[ 82.210376] [<c0111a54>] (unwind_backtrace) from [<c010d8f4>] (show_stack+0x10/0x14)
[ 82.218084] [<c010d8f4>] (show_stack) from [<c09ef004>] (dump_stack+0x90/0xc8)
[ 82.225278] [<c09ef004>] (dump_stack) from [<c0152980>] (___might_sleep+0x22c/0x2c8)
[ 82.232990] [<c0152980>] (___might_sleep) from [<c0a0a2e4>] (__mutex_lock+0x28/0xa3c)
[ 82.240788] [<c0a0a2e4>] (__mutex_lock) from [<c0a0ad80>] (mutex_lock_nested+0x1c/0x24)
[ 82.248763] [<c0a0ad80>] (mutex_lock_nested) from [<c04923dc>] (clk_prepare_lock+0x78/0xec)
[ 82.257079] [<c04923dc>] (clk_prepare_lock) from [<c049538c>] (clk_core_get_rate+0xc/0x5c)
[ 82.265309] [<c049538c>] (clk_core_get_rate) from [<c0766b18>] (i2s_trigger+0x490/0x6d4)
[ 82.273369] [<c0766b18>] (i2s_trigger) from [<c074fec4>] (soc_pcm_trigger+0x100/0x140)
[ 82.281254] [<c074fec4>] (soc_pcm_trigger) from [<c07378a0>] (snd_pcm_do_start+0x2c/0x30)
[ 82.289400] [<c07378a0>] (snd_pcm_do_start) from [<c07376cc>] (snd_pcm_action_single+0x38/0x78)
[ 82.298065] [<c07376cc>] (snd_pcm_action_single) from [<c073a450>] (snd_pcm_ioctl+0x910/0x1268)
[ 82.306734] [<c073a450>] (snd_pcm_ioctl) from [<c0292344>] (do_vfs_ioctl+0x90/0x9ec)
[ 82.314443] [<c0292344>] (do_vfs_ioctl) from [<c0292cd4>] (ksys_ioctl+0x34/0x60)
[ 82.321808] [<c0292cd4>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28)
[ 82.329431] Exception stack(0xeb875fa8 to 0xeb875ff0)
[ 82.334459] 5fa0: 00033c18 b6e31000 00000004 00004142 00033d80 00033d80
[ 82.342605] 5fc0: 00033c18 b6e31000 00008000 00000036 00008000 00000000 beea38a8 00008000
[ 82.350748] 5fe0: b6e3142c beea384c b6da9a30 b6c9212c
[ 82.355789]
[ 82.357245] ======================================================
[ 82.363397] WARNING: possible circular locking dependency detected
[ 82.369551] 5.0.0-rc5-00192-ga6e6caca8f03 #191 Tainted: G W
[ 82.376395] ------------------------------------------------------
[ 82.382548] speaker-test/1554 is trying to acquire lock:
[ 82.387834] 6d2007f4 (prepare_lock){+.+.}, at: clk_prepare_lock+0x78/0xec
[ 82.394593]
[ 82.394593] but task is already holding lock:
[ 82.400398] 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4
[ 82.408197]
[ 82.408197] which lock already depends on the new lock.
[ 82.416343]
[ 82.416343] the existing dependency chain (in reverse order) is:
[ 82.423795]
[ 82.423795] -> #1 (&(&pri_dai->spinlock)->rlock){..-.}:
[ 82.430472] clk_mux_set_parent+0x34/0xb8
[ 82.434975] clk_core_set_parent_nolock+0x1c4/0x52c
[ 82.440347] clk_set_parent+0x38/0x6c
[ 82.444509] of_clk_set_defaults+0xc8/0x308
[ 82.449186] of_clk_add_provider+0x84/0xd0
[ 82.453779] samsung_i2s_probe+0x408/0x5f8
[ 82.458376] platform_drv_probe+0x48/0x98
[ 82.462879] really_probe+0x224/0x3f4
[ 82.467037] driver_probe_device+0x70/0x1c4
[ 82.471716] bus_for_each_drv+0x44/0x8c
[ 82.476049] __device_attach+0xa0/0x138
[ 82.480382] bus_probe_device+0x88/0x90
[ 82.484715] deferred_probe_work_func+0x6c/0xbc
[ 82.489741] process_one_work+0x200/0x740
[ 82.494246] worker_thread+0x2c/0x4c8
[ 82.498408] kthread+0x128/0x164
[ 82.502131] ret_from_fork+0x14/0x20
[ 82.506204] (null)
[ 82.508976]
[ 82.508976] -> #0 (prepare_lock){+.+.}:
[ 82.514264] __mutex_lock+0x60/0xa3c
[ 82.518336] mutex_lock_nested+0x1c/0x24
[ 82.522756] clk_prepare_lock+0x78/0xec
[ 82.527088] clk_core_get_rate+0xc/0x5c
[ 82.531421] i2s_trigger+0x490/0x6d4
[ 82.535494] soc_pcm_trigger+0x100/0x140
[ 82.539913] snd_pcm_do_start+0x2c/0x30
[ 82.544246] snd_pcm_action_single+0x38/0x78
[ 82.549012] snd_pcm_ioctl+0x910/0x1268
[ 82.553345] do_vfs_ioctl+0x90/0x9ec
[ 82.557417] ksys_ioctl+0x34/0x60
[ 82.561229] ret_fast_syscall+0x0/0x28
[ 82.565477] 0xbeea384c
[ 82.568421]
[ 82.568421] other info that might help us debug this:
[ 82.568421]
[ 82.576394] Possible unsafe locking scenario:
[ 82.576394]
[ 82.582285] CPU0 CPU1
[ 82.586792] ---- ----
[ 82.591297] lock(&(&pri_dai->spinlock)->rlock);
[ 82.595977] lock(prepare_lock);
[ 82.601782] lock(&(&pri_dai->spinlock)->rlock);
[ 82.608975] lock(prepare_lock);
[ 82.612268]
[ 82.612268] *** DEADLOCK ***
Fixes: 647d04f8e07a ("ASoC: samsung: i2s: Ensure the RCLK rate is properly determined")
Reported-by: Krzysztof Kozłowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Currently "0xf << 36" is used to
clear SSIU-9 internal buffer state, which overflows 32-bit value
according to user reference manual, it is always bit4 ~ bit7
of SSI_SYS_STATUS[1,3,5,7] registers indicate
SSIU-9's buffer state, so "0xf << 4" should be used.
This patch fix incorrect shifting issue in SSIU-9 case
Fixes: commit b7169ddea2f2 ("ASoC: rsnd: remove RSND_REG_ from rsnd_reg")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of a few small fixes.
The most significant one is the fix for the possible race at loading
HD-audio drivers. This has been present for long time and surfaced
only in a rare occasion, but finally spotted out"
* tag 'sound-5.0-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/ca0132 - Fix build error without CONFIG_PCI
ALSA: compress: Fix stop handling on compressed capture streams
ALSA: usb-audio: Add support for new T+A USB DAC
ALSA: hda - Serialize codec registrations
ALSA: hda/realtek - Use a common helper for hp pin reference
ALSA: hda/realtek - Fix lose hp_pins for disable auto mute
ALSA: hda/realtek - Headset microphone support for System76 darp5
|
|
commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set
under non-atomic") added new rsnd_ssi_prepare() and moved
rsnd_ssi_master_clk_start() to .prepare.
But, ssi user count (= ssi->usrcnt) is incremented at .init
(= rsnd_ssi_init()).
Because of these timing exchange, ssi->usrcnt check at
rsnd_ssi_master_clk_start() should be adjusted.
Otherwise, 2nd master clock setup will be no check.
This patch fixup this issue.
Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic")
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Reported-by: Valentine Barshak <valentine.barshak@cogentembedded.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
KASAN reports and additional traces point to out-of-bounds accesses to
the dapm_up_seq and dapm_down_seq lookup tables. The indices used are
larger than the array definition.
Fix by adding missing entries for the new widget types in these two
lookup tables, and align them with PGA values.
Also the sequences for the following widgets were not defined. Since
their values defaulted to zero, assign them explicitly
snd_soc_dapm_input
snd_soc_dapm_output
snd_soc_dapm_vmid
snd_soc_dapm_siggen
snd_soc_dapm_sink
Fixes: 8a70b4544ef4 ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.').
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
A call of pci_iounmap() call without CONFIG_PCI leads to a build error
on some architectures. We tried to address this and add a check of
IS_ENABLED(CONFIG_PCI), but this still doesn't seem enough for sh.
Ideally we should fix it globally, it's really a corner case, so let's
paper over it with a simpler ifdef.
Fixes: 1e73359a24fa ("ALSA: hda/ca0132 - make pci_iounmap() call conditional")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds the T+A VID to the generic check in order to enable
native DSD support for T+A devices. This works with the new T+A USB
DAC model SD3100HV and will also work with future devices which
support the XMOS/Thesycon style DSD format.
Signed-off-by: Udo Eberhardt <udo.eberhardt@thesycon.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
rmmod/modprobe tests expose a kernel oops when accessing the dai
driver pointer. This comes from the topology design which operates in
multiple passes. Each object removal happens at a specific iteration,
and the code checks for the iteration (order) number after the memory
containing the order was freed.
Fix this be clearing a reference to the dai driver and check its
validity to avoid dereferences.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Renesas sound device has many IPs and many situations.
If platform/board uses MIXer, situation will be more complex.
To avoid duplicate DVC kctrl registration when MIXer was used,
it had original flags.
But it was issue when sound card was re-binded, because
no one can't cleanup this flags then.
To solve this issue, commit 9c698e8481a15237a ("ASoC: rsnd: tidyup
registering method for rsnd_kctrl_new()") checks registered
card->controls, because if card was re-binded, these were cleanuped
automatically. This patch could solve re-binding issue.
But, it start to avoid MIX kctrl.
To solve these issues, we need below.
To avoid card re-binding issue: check registered card->controls
To avoid duplicate DVC registration: check registered rsnd_kctrl_cfg
To allow multiple MIX registration: check registered rsnd_kctrl_cfg
This patch do it.
Fixes: 9c698e8481a15237a ("ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-By: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
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In the current code, the codec registration may happen both at the
codec bind time and the end of the controller probe time. In a rare
occasion, they race with each other, leading to Oops due to the still
uninitialized card device.
This patch introduces a simple flag to prevent the codec registration
at the codec bind time as long as the controller probe is going on.
The controller probe invokes snd_card_register() that does the whole
registration task, and we don't need to register each piece
beforehand.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Replace the open-codes in many places with a new common helper for
performing the same thing: referring to the primary headphone pin.
This eventually fixes the potentially missing headphone pin on some
weird devices, too.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
When auto_mute = no or spec->suppress_auto_mute = 1, cfg->hp_pins will
lose value.
Add this patch to find hp_pins value.
I add fixed for ALC282 ALC225 ALC256 ALC294 and alc_default_init()
alc_default_shutup().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Only three fixes.
The fix for Realtek HD-audio looks lengthy, but it's just a code
shuffling, and the actual changes are fairly small.
The rest are a PCM core fix for a long-standing bug that was recently
scratched by syzkaller, and a trivial USB-audio quirk for DSD support"
* tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fixed hp_pin no value
ALSA: pcm: Fix tight loop of OSS capture stream
ALSA: usb-audio: Add Opus #3 to quirks for native DSD support
|
|
On the System76 Darter Pro (darp5), there is a headset microphone
input attached to 0x1a that does not have a jack detect. In order to
get it working, the pin configuration needs to be set correctly, and
the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied.
This is similar to the MIC_NO_PRESENCE fixups for some Dell laptops,
except we have a separate microphone jack that is already configured
correctly.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fix hp_pin always no value.
[More notes on the changes:
The hp_pin value that is referred in alc294_hp_init() is always zero
at the moment the function gets called, hence this is actually
useless as in the current code.
And, this kind of init sequence should be called from the codec init
callback, instead of the parser function. So, the first fix in this
patch to move the call call into its own init_hook.
OTOH, this function is needed to be called only once after the boot,
and it'd take too long for invoking at each resume (where the init
callback gets called). So we add a new flag and invoke this only
once as an additional fix.
The one case is still not covered, though: S4 resume. But this
change itself won't lead to any regression in that regard, so we
leave S4 issue as is for now and fix it later. -- tiwai ]
Fixes: bde1a7459623 ("ALSA: hda/realtek - Fixed headphone issue for ALC700")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the trigger=off is passed for a PCM OSS stream, it sets the
start_threshold of the given substream to the boundary size, so that
it won't be automatically started. This can be problematic for a
capture stream, unfortunately, as detected by syzkaller. The scenario
is like the following:
- In __snd_pcm_lib_xfer() that is invoked from snd_pcm_oss_read()
loop, we have a check whether the stream was already started or the
stream can be auto-started.
- The function at this check returns 0 with trigger=off since we
explicitly disable the auto-start.
- The loop continues and repeats calling __snd_pcm_lib_xfer() tightly,
which may lead to an RCU stall.
This patch fixes the bug by simply allowing the wait for non-started
stream in the case of OSS capture. For native usages, it's supposed
to be done by the caller side (which is user-space), hence it returns
zero like before.
(In theory, __snd_pcm_lib_xfer() could wait even for the native API
usage cases, too; but I'd like to stay in a safer side for not
breaking the existing stuff for now.)
Reported-by: syzbot+fbe0496f92a0ce7b786c@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit')
in order to enable Native DSD support.
[ NOTE: this could be handled in the generic way with fp->dvd_raw if
we add 0x10cb to the vendor whitelist, but since 0x10cb shows a
different vendor name (Erantech), put to the individual entry at
this time -- tiwai ]
Signed-off-by: Olek Poplavsky <woodenbits@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
For devices implemented as a MFD it is common to only have a single node
in devicetree representing the whole device. As such when looking up
components in soc_find_components we should match against both the devices
of_node and the devices parent's of_node, as is already done in the rest
of the ASoC core.
This causes regressions for some DAI links at the moment as
soc_find_component was recently added as a check in soc_init_dai_link.
Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A significant amount of fixes at this time, mostly for covering the
recent ASoC issues.
- Fixes for the missing ASoC driver initialization with non-deferred
probes; these triggered other problems in chain, which resulted in
yet more fix commits
- DaVinci runtime PM fix; the diff looks large but it's just a code
shuffling
- Various fixes for ASoC Intel drivers: a regression in HD-A HDMI,
Kconfig dependency, machine driver adjustments, PLL fix.
- Other ASoC driver-specific stuff including the trivial fixes caught
by static analysis
- Usual HD-audio quirks"
* tag 'sound-5.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
ALSA: hda - Add mute LED support for HP ProBook 470 G5
ASoC: amd: Fix potential NULL pointer dereference
ASoC: imx-audmux: change snprintf to scnprintf for possible overflow
ASoC: rt5514-spi: Fix potential NULL pointer dereference
ASoC: dapm: change snprintf to scnprintf for possible overflow
ASoC: rt5682: Fix PLL source register definitions
ASoC: core: Don't defer probe on optional, NULL components
ASoC: core: Make snd_soc_find_component() more robust
ASoC: soc-core: fix init platform memory handling
ASoC: intel: skl: Fix display power regression
ALSA: hda/realtek - Fix typo for ALC225 model
ASoC: soc-core: Hold client_mutex around soc_init_dai_link()
ASoC: Intel: Boards: move the codec PLL configuration to _init
ASoC: soc-core: defer card probe until all component is added to list
ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages
ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode
ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks
ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized
ASoC: rt5682: Fix recording no sound issue
ASoC: Intel: atom: Make PCI dependency explicit
...
|
|
AD/DA ASRC function control two ASRC clock sources separately.
Whether AD/DA filter select which clock source, we enable AD/DA ASRC
function for all cases.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Support speaker and mic mute LEDs on HP ProBook 470 G5.
BugLink: https://bugs.launchpad.net/bugs/1811254
Signed-off-by: Anthony Wong <anthony.wong@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
hdmi-codec oopses the kernel when it is unbound from a successfully
bound audio subsystem, and is then rebound:
Unable to handle kernel NULL pointer dereference at virtual address 0000001c
pgd = ee3f0000
[0000001c] *pgd=3cc59831
Internal error: Oops: 817 [#1] PREEMPT ARM
Modules linked in: ext2 snd_soc_spdif_tx vmeta dove_thermal snd_soc_kirkwood ofpart marvell_cesa m25p80 orion_wdt mtd spi_nor des_generic gpio_ir_recv snd_soc_kirkwood_spdif bmm_dmabuf auth_rpcgss nfsd autofs4 etnaviv thermal_sys hwmon gpu_sched tda9950
CPU: 0 PID: 1005 Comm: bash Not tainted 4.20.0+ #1762
Hardware name: Marvell Dove (Cubox)
PC is at hdmi_dai_probe+0x68/0x80
LR is at find_held_lock+0x20/0x94
pc : [<c04c7de0>] lr : [<c0063bf4>] psr: 600f0013
sp : ee15bd28 ip : eebd8b1c fp : c093b488
r10: ee048000 r9 : eebdab18 r8 : ee048600
r7 : 00000001 r6 : 00000000 r5 : 00000000 r4 : ee82c100
r3 : 00000006 r2 : 00000001 r1 : c067e38c r0 : ee82c100
Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none[ 297.318599] Control: 10c5387d Table: 2e3f0019 DAC: 00000051
Process bash (pid: 1005, stack limit = 0xee15a248)
...
[<c04c7de0>] (hdmi_dai_probe) from [<c04b7060>] (soc_probe_dai.part.9+0x34/0x70)
[<c04b7060>] (soc_probe_dai.part.9) from [<c04b81a8>] (snd_soc_instantiate_card+0x734/0xc9c)
[<c04b81a8>] (snd_soc_instantiate_card) from [<c04b8b6c>] (snd_soc_add_component+0x29c/0x378)
[<c04b8b6c>] (snd_soc_add_component) from [<c04b8c8c>] (snd_soc_register_component+0x44/0x54)
[<c04b8c8c>] (snd_soc_register_component) from [<c04c64b4>] (devm_snd_soc_register_component+0x48/0x84)
[<c04c64b4>] (devm_snd_soc_register_component) from [<c04c7be8>] (hdmi_codec_probe+0x150/0x260)
[<c04c7be8>] (hdmi_codec_probe) from [<c0373124>] (platform_drv_probe+0x48/0x98)
This happens because hdmi_dai_probe() attempts to access the HDMI
codec private data, but this has not been assigned by hdmi_dai_probe()
before it calls devm_snd_soc_register_component(). Move the call to
dev_set_drvdata() before devm_snd_soc_register_component() to avoid
this oops.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
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Hui Wang
Takashi Iwai
Takashi Sakamoto
Jaroslav Kysela
Codrin Ciubotariu
S.j. Wang
Jian-Hong Pan
Linus Torvalds
Kuninori Morimoto
Bard liao
Jeremy Soller
Sylwester Nawrocki
Manuel Reinhardt
Jurica Vukadin
Jiada Wang
Pierre-Louis Bossart
Udo Eberhardt
Guennadi Liakhovetski
Kailang Yang
Olek Poplavsky
Charles Keepax
Shuming Fan
Anthony Wong
Russell King