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[ 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>
[ Upstream commit eaa1b01fe709d6a236a9cec74813e0400601fd23 ]
For devices with multiple clock sources connected to a selector, we need
to check what a clock selector control request has returned. This is
needed to ensure that a requested clock source is indeed selected and for
autoclock feature to work.
For devices with single clock source connected, if we get an error there
is nothing else we can do about it. We can't skip clock selector setup as
it is required by some devices. So lets just ignore error in this case.
This should fix various buggy Mackie devices:
[ 649.109785] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.111946] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
[ 649.113822] usb 1-1.3: parse_audio_format_rates_v2v3(): unable to find clock source (clock -32)
There is also interesting info from the Windows documentation [1] (this
is probably why manufacturers dont't even test this feature):
"The USB Audio 2.0 driver doesn't support clock selection. The driver
uses the Clock Source Entity, which is selected by default and never
issues a Clock Selector Control SET CUR request."
Link: https://learn.microsoft.com/en-us/windows-hardware/drivers/audio/usb-2-0-audio-drivers [1]
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217314
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218175
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218342
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240201115308.17838-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0adf963b8463faa44653e22e56ce55f747e68868 ]
The SPDIF hardware block found in the H616 SoC has the same layout as
the one found in the H6 SoC, except that it is missing the receiver
side.
Since the driver currently only supports the transmit function, support
for the H616 is identical to what is currently done for the H6.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Reviewed-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Link: https://msgid.link/r/20240127163247.384439-4-wens@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 346f59d1e8ed0eed41c80e1acb657e484c308e6a ]
Many devices with a single alternate setting do not have a Valid
Alternate Setting Control and validation performed by
validate_sample_rate_table_v2v3() doesn't work on them and is not
really needed. So check the presense of control before sending
altsetting validation requests.
MOTU Microbook IIc is suffering the most without this check. It
takes up to 40 seconds to bootup due to how slow it switches
sampling rates:
[ 2659.164824] usb 3-2: New USB device found, idVendor=07fd, idProduct=0004, bcdDevice= 0.60
[ 2659.164827] usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0
[ 2659.164829] usb 3-2: Product: MicroBook IIc
[ 2659.164830] usb 3-2: Manufacturer: MOTU
[ 2659.166204] usb 3-2: Found last interface = 3
[ 2679.322298] usb 3-2: No valid sample rate available for 1:1, assuming a firmware bug
[ 2679.322306] usb 3-2: 1:1: add audio endpoint 0x3
[ 2679.322321] usb 3-2: Creating new data endpoint #3
[ 2679.322552] usb 3-2: 1:1 Set sample rate 96000, clock 1
[ 2684.362250] usb 3-2: 2:1: cannot get freq (v2/v3): err -110
[ 2694.444700] usb 3-2: No valid sample rate available for 2:1, assuming a firmware bug
[ 2694.444707] usb 3-2: 2:1: add audio endpoint 0x84
[ 2694.444721] usb 3-2: Creating new data endpoint #84
[ 2699.482103] usb 3-2: 2:1 Set sample rate 96000, clock 1
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240129121254.3454481-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 4639c5021029d49fd2f97fa8d74731f167f98919 upstream.
The SWS JS201D need a different pinconfig from windows driver.
Add a quirk to use a specific pinconfig to SWS JS201D.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240205013802.51907-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a2ed0a44d637ef9deca595054c206da7d6cbdcbc upstream.
Customer has reported an issue with specific desktop platform
where two CS42L42 codecs are connected to CS8409 HDA bridge.
If "Master Volume Control" is created then on Ubuntu OS UCM
left/right balance slider in UI audio settings has no effect.
This patch will fix this issue for a target paltform.
Fixes: 20e507724113 ("ALSA: hda/cs8409: Add support for dolphin")
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240122184710.5802-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 086df711d9b886194481b4fbe525eb43e9ae7403 upstream.
WCD938x sound codec driver ignores return status of getting regulators
and returns EINVAL instead of EPROBE_DEFER. If regulator provider
probes after the codec, system is left without probed audio:
wcd938x_codec audio-codec: wcd938x_probe: Fail to obtain platform data
wcd938x_codec: probe of audio-codec failed with error -22
Fixes: 16572522aece ("ASoC: codecs: wcd938x-sdw: add SoundWire driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240117151208.1219755-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c7de2d9bb68a5fc71c25ff96705a80a76c8436eb upstream.
Vaio VJFE-ADL is equipped with ALC269VC, and it needs
ALC298_FIXUP_SPK_VOLUME quirk to make its headset mic work.
Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240201122114.30080-1-edson.drosdeck@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit efb56d84dd9c3de3c99fc396abb57c6d330038b5 upstream.
If you connect an external headset/microphone to the 3.5mm jack on the
Acer Swift 1 SF114-32 it does not recognize the microphone. This fixes
that and gives the user the ability to choose between internal and
headset mic.
Signed-off-by: David Senoner <seda18@rolmail.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240126155626.2304465-1-seda18@rolmail.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 6ef5d5b92f7117b324efaac72b3db27ae8bb3082 ]
There is a path in rt5645_jack_detect_work(), where rt5645->jd_mutex
is left locked forever. That may lead to deadlock
when rt5645_jack_detect_work() is called for the second time.
Found by Linux Verification Center (linuxtesting.org) with SVACE.
Fixes: cdba4301adda ("ASoC: rt5650: add mutex to avoid the jack detection failure")
Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Link: https://lore.kernel.org/r/1707645514-21196-1-git-send-email-khoroshilov@ispras.ru
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit a969210066054ea109d8b7aff29a9b1c98776841 upstream.
The device fails to initialize otherwise, giving the following error:
[ 3676.671641] usb 2-1.1: 1:1: cannot get freq at ep 0x1
Signed-off-by: Julian Sikorski <belegdol+github@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240123084935.2745-1-belegdol+github@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 46188db080bd1df7d2d28031b89e56f2fdbabd67 upstream.
The LPASS WSA macro codec driver is updating the digital gain settings
behind the back of user space on DAPM events if companding has been
enabled.
As compander control is exported to user space, this can result in the
digital gain setting being incremented (or decremented) every time the
sound server is started and the codec suspended depending on what the
UCM configuration looks like.
Soon enough playback will become distorted (or too quiet).
This is specifically a problem on the Lenovo ThinkPad X13s as this
bypasses the limit for the digital gain setting that has been set by the
machine driver.
Fix this by simply dropping the compander gain offset hack. If someone
cares about modelling the impact of the compander setting this can
possibly be done by exporting it as a volume control later.
Note that the volume registers still need to be written after enabling
clocks in order for any prior updates to take effect.
Fixes: 2c4066e5d428 ("ASoC: codecs: lpass-wsa-macro: add dapm widgets and route")
Cc: stable@vger.kernel.org # 5.11
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://msgid.link/r/20240119112420.7446-4-johan+linaro@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 7aeb259086487417f0fecf66e325bee133e8813a ]
When OMTP headset plugin the headset jack of CX8070 and SN6160 sound cards,
the headset type detection circuit will recognize the headset type as CTIA.
At this point, plugout and plugin the headset will get the correct headset
type as OMTP.
The reason for the failure of headset type recognition is that the sound
card creation will enable the VREF voltage of the headset mic, which
interferes with the headset type automatic detection circuit. Plugout and
plugin the headset will restart the headset detection and get the correct
headset type.
The patch is disable the VREF voltage when the headset is not present, and
will enable the VREF voltage when the headset is present.
Signed-off-by: bo liu <bo.liu@senarytech.com>
Link: https://lore.kernel.org/r/20240108110235.3867-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 26257869672fd4a06a60c2da841e15fb2cb47bbe ]
In a couple of loops over the all streams, we check the bitmap against
the loop counter. A more correct reference would be, however, the
index of each stream, instead.
This patch corrects the check of bitmaps to the stream index.
Note that this change doesn't fix anything for now; all existing
drivers set up the stream indices properly, hence the loop count is
always equal with the stream index. That said, this change is only
for consistency.
Link: https://lore.kernel.org/r/20231121154125.4888-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit d915a6850e27efb383cd4400caadfe47792623df upstream.
Audio control requests that sets sampling frequency sometimes fail on
this card. Adding delay between control messages eliminates that problem.
Link: https://bugzilla.kernel.org/show_bug.cgi?id=217601
Cc: <stable@vger.kernel.org>
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20240124130239.358298-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bc7863d18677df66b2c7a0e172c91296ff380f11 upstream.
This HP Laptop uses ALC236 codec with COEF 0x07 idx 1 controlling
the mute LED. This patch enables the already existing quirk for
this device.
Signed-off-by: Çağhan Demir <caghandemir@marun.edu.tr>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20240115172303.4718-1-caghandemir@marun.edu.tr
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a03cfad512ac24a35184d7d87ec0d5489e1cb763 upstream.
There was a typo in oxygen mixer code that didn't update the right
channel value properly for the capture volume. Let's fix it.
This trivial fix was originally reported on Bugzilla.
Fixes: a3601560496d ("[ALSA] oxygen: add front panel controls")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.kernel.org/show_bug.cgi?id=156561
Link: https://lore.kernel.org/r/20240112111023.6208-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 04f8f053252b86c7583895c962d66747ecdc61b7 ]
Ensure the value passed to scarlett2_mixer_ctl_put() is between 0 and
SCARLETT2_MIXER_MAX_VALUE so we don't attempt to access outside
scarlett2_mixer_values[].
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Link: https://lore.kernel.org/r/3b19fb3da641b587749b85fe1daa1b4e696c0c1b.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 50603a67daef161c78c814580d57f7f0be57167e ]
The *_ctl_get() functions which call scarlett2_update_*() were not
checking the return value. Fix to check the return value and pass to
the caller.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Link: https://lore.kernel.org/r/32a5fdc83b05fa74e0fcdd672fbf71d75c5f0a6d.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2190b9aea4eb92ccf3176e35c17c959e40f1a81b ]
Line outputs 3 & 4 on the Gen 3 18i8 are internally the analogue 7 and
8 outputs, and this renumbering is hidden from the user by
line_out_remap(). By allowing higher values (representing non-analogue
outputs) to be passed to line_out_remap(), repeated code from
scarlett2_mux_src_enum_ctl_get() and scarlett2_mux_src_enum_ctl_put()
can be removed.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Link: https://lore.kernel.org/r/3b70267931f5994628ab27306c73cddd17b93c8f.1698342632.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stable-dep-of: 50603a67daef ("ALSA: scarlett2: Add missing error checks to *_ctl_get()")
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ca459dfa7d4ed9098fcf13e410963be6ae9b6bf3 ]
scarlett2_usb_set_config() calls scarlett2_usb_get() but was not
checking the result. Return the error if it fails rather than
continuing with an invalid value.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e15fae6c51a ("ALSA: usb-audio: scarlett2: Allow bit-level access to config")
Link: https://lore.kernel.org/r/def110c5c31dbdf0a7414d258838a0a31c0fab67.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f6ff6931a1c0065a55448108940371e1ac8075f ]
scarlett2_config_save() was ignoring the return value from
scarlett2_usb(). As this function is not called from user-space we
can't return the error, so call usb_audio_err() instead.
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Link: https://lore.kernel.org/r/bf0a15332d852d7825fa6da87d2a0d9c0b702053.1703001053.git.g@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 51add1687f39292af626ac3c2046f49241713273 ]
dmi_platform_data[] first contains a DMI entry matching:
DMI_MATCH(DMI_PRODUCT_NAME, "EF20"),
and then contains an identical entry except for the match being:
DMI_MATCH(DMI_PRODUCT_NAME, "EF20EA"),
Since these are partial (non exact) DMI matches the first match
will also match any board with "EF20EA" in their DMI product-name,
drop the second, redundant, entry.
Fixes: a4dae468cfdd ("ASoC: rt5645: Add ACPI-defined GPIO for ECS EF20 series")
Cc: Chris Chiu <chiu@endlessos.org>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://msgid.link/r/20231126214024.300505-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a6122b0b4211d132934ef99e7b737910e6d54d2f ]
This driver includes the legacy GPIO APIs <linux/gpio.h> and
<linux/of_gpio.h> but does not use any symbols from any of
them.
Drop the includes.
Further the driver is requesting "reset-gpios" rather than
just "reset" from the GPIO framework. This is wrong because
the gpiolib core will add "-gpios" before processing the
request from e.g. device tree. Drop the suffix.
The last problem means that the optional RESET GPIO has
never been properly retrieved and used even if it existed,
but nobody noticed.
Fixes: c1124c09e103 ("ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver.")
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-3-ee9f9d4655eb@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 50678d339d670a92658e5538ebee30447c88ccb3 ]
This driver includes the legacy GPIO APIs <linux/gpio.h> and
<linux/of_gpio.h> but does not use any symbols from any of
them.
Drop the includes.
Further the driver is requesting "reset-gpios" rather than
just "reset" from the GPIO framework. This is wrong because
the gpiolib core will add "-gpios" before processing the
request from e.g. device tree. Drop the suffix.
The last problem means that the optional RESET GPIO has
never been properly retrieved and used even if it existed,
but nobody noticed.
Fixes: 3333cb7187b9 ("ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.")
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-2-ee9f9d4655eb@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
This reverts commit bc7d0133181e5f33ac33ca4f6bb2bce876c8ad88 which is
commit c775cbf62ed4911e4f0f23880f01815753123690 upstream.
It is reported to cause problems, so drop it from the 5.15.y tree for now.
Link: https://lore.kernel.org/r/845b3053-d47b-4717-9665-79b120da133b@sirena.org.uk
Reported-by: Mark Brown <broonie@kernel.org>
Cc: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Cc: Sasha Levin <sashal@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 99c7bb44f5749373bc01b73af02b50b69bcbf43d ]
Add a quirk for the Medion Lifetab S10346, this BYTCR tablet has no CHAN
package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which
is the default for BYTCR devices.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://msgid.link/r/20231217213221.49424-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fb9ad24485087e0f00d84bee7a5914640b2b9024 ]
Volume can have ranges that start with negative values, ex: -84dB to
+40dB. Apply correct range check in snd_soc_limit_volume before setting
the platform_max. Without this patch, for example setting a 0dB limit on
a volume range of -84dB to +40dB would fail.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Johan Hovold <johan+linaro@kernel.org>
Reviewed-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://lore.kernel.org/r/20231204124736.132185-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5f44de697383fcc9a9a1a78f99e09d1838704b90 ]
Change the default MIC detection impedance threshold to 200ohm
to support low mic DC impedance headset.
Signed-off-by: David Rau <David.Rau.opensource@dm.renesas.com>
Link: https://lore.kernel.org/r/20231201042933.26392-1-David.Rau.opensource@dm.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cdba4301adda7c60a2064bf808e48fccd352aaa9 ]
This patch adds the jd_mutex to protect the jack detection control flow.
And only the headset type could check the button status.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Link: https://lore.kernel.org/r/20231122100123.2831753-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f8ba14b780273fd290ddf7ee0d7d7decb44cc365 ]
skl_platform_register() uses krealloc. When krealloc is fail,
then previous memory is not freed. The leak is also when soc
component registration failed.
Signed-off-by: Kamil Duljas <kamil.duljas@gmail.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20231116224112.2209-2-kamil.duljas@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1501f2597dd08601acd42256a4b0a0fc36bf302 ]
This issue is reproduced when W=1 build in compiler gcc-12.
The following are sparse warnings:
sound/soc/codecs/nau8822.c:199:25: sparse: sparse: incorrect type in assignment
sound/soc/codecs/nau8822.c:199:25: sparse: expected unsigned short
sound/soc/codecs/nau8822.c:199:25: sparse: got restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
sound/soc/codecs/nau8822.c:235:25: sparse: sparse: cast to restricted __be16
Reported-by: kernel test robot <lkp@intel.com>
Closes: https://lore.kernel.org/oe-kbuild-all/202311122320.T1opZVkP-lkp@intel.com/
Signed-off-by: David Lin <CTLIN0@nuvoton.com>
Link: https://lore.kernel.org/r/20231117043011.1747594-1-CTLIN0@nuvoton.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
