Mark Brown [Thu, 1 Dec 2022 17:07:43 +0000 (17:07 +0000)]
kselftest/alsa: Don't any configuration in the sample config
The values in the one example configuration file we currently have are the
default values for the two tests we have so there's no need to actually set
them. Comment them out as examples, with a rename for the tests so that we
can update the tests in the code more easily.
Mark Brown [Thu, 1 Dec 2022 17:07:42 +0000 (17:07 +0000)]
kselftest/alsa: Report failures to set the requested channels as skips
If constraint selection gives us a number of channels other than the one
that we asked for that isn't a failure, that is the device implementing
constraints and advertising that it can't support whatever we asked
for. Report such cases as a test skip rather than failure so we don't have
false positives.
Mark Brown [Thu, 1 Dec 2022 17:07:41 +0000 (17:07 +0000)]
kselftest/alsa: Report failures to set the requested sample rate as skips
If constraint selection gives us a sample rate other than the one that we
asked for that isn't a failure, that is the device implementing sample
rate constraints and advertising that it can't support whatever we asked
for. Report such cases as a test skip rather than failure so we don't have
false positives.
Mark Brown [Thu, 1 Dec 2022 17:07:40 +0000 (17:07 +0000)]
kselftest/alsa: Refactor pcm-test to list the tests to run in a struct
In order to help make the list of tests a bit easier to maintain refactor
things so we pass the tests around as a struct with the parameters in,
enabling us to add new tests by adding to a table with comments saying
what each of the number are. We could also use named initializers if we get
more parameters.
Takashi Sakamoto [Wed, 30 Nov 2022 14:33:13 +0000 (23:33 +0900)]
ALSA: dice: add support for Focusrite Saffire Pro 40 with TCD3070 ASIC
TC Applied Technologies (TCAT) produces TCD3070 as final DICE ASIC for
communication in IEEE 1394 bus for IEC 61883-1/6 protocol. As long as I
know, latter model of Focusrite Saffire Pro 40 is an application of the
ASIC and only in the market for consumers.
This patchset adds support for the device. The device has several
remarkable points.
1. No support for extended synchronization information section in TCAT
general protocol. The value of GLOBAL_EXTENDED_STATUS register is always
zero. Additionally, NOTIFY_EXT_STATUS message is never emitted.
2. No support for TCAT protocol extension. Hard coding is required for
format of CIP payload.
3. During several seconds after changing sampling rate, the block to
process PCM frames is under disfunction. When starting packet streaming
during the state, the block is never function till configuring different
sampling rate and several seconds.
This commit adds support for the device. The item 1 and 2 can be
adaptable, while item 3 is not. It's not preferable that user process
is forced to sleep during the disfunction in the call of ioctl(2) with
SNDRV_PCM_IOCTL_HW_PARAMS or SNDRV_PCM_IOCTL_PREPARE request. It's
inconvenient but let user configure preferable sampling rate in advance
of starting PCM substream.
The content of configuration ROM in the device I used is available at:
* https://github.com/takaswie/am-config-roms/
I note that any mixer control operation is implemented by unique
transaction. The frame of request consists of 16 bytes header followed
by payload.
header (4 quadlets):
1st: the type of request, prefixed with 0x8000
2nd: counter at 2 bytes in MSB side, the length of data at 2 bytes in LSB
side
3rd: parameter 0
4th: parameter 1
payload (variable length if need):
5th-: data according to parameters
The request frame is sent by block write request to 0x'ffff'e040'01c0.
The frame of response is similar to the frame of request, but it is
header only, thus fixed to 16 bytes. The response frame is sent to the
address which is registered by lock transaction to 0x'ffff'e040'0008.
If the operation results in batch of data, the 2nd quadlet of header
includes the length of data like request. The data is itself readable
by read block request to 0x'ffff'e040'0030, which includes both
header and payload for data, thus the length to read should be the
length of data plus 16 bytes for header
The actual value of request, parameter 0, parameter 1, and data is
unclear yet.
Jaroslav Kysela [Tue, 29 Nov 2022 08:53:06 +0000 (09:53 +0100)]
selftests: alsa - move shared library configuration code to conf.c
The minimal alsa-lib configuration code is similar in both mixer
and pcm tests. Move this code to the shared conf.c source file.
Also, fix the build rules inspired by rseq tests. Build libatest.so
which is linked to the both test utilities dynamically.
Also, set the TEST_FILES variable for lib.mk.
Cc: linux-kselftest@vger.kernel.org Cc: Shuah Khan <shuah@kernel.org> Reported-by: Mark Brown <broonie@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Tested-by: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20221129085306.2345763-1-perex@perex.cz Signed-off-by: Takashi Iwai <tiwai@suse.de>
John Keeping [Tue, 29 Nov 2022 13:00:59 +0000 (13:00 +0000)]
ALSA: usb-audio: Add quirk for Tascam Model 12
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.
Baisong Zhong [Mon, 21 Nov 2022 11:16:30 +0000 (19:16 +0800)]
ALSA: seq: fix undefined behavior in bit shift for SNDRV_SEQ_FILTER_USE_EVENT
Shifting signed 32-bit value by 31 bits is undefined, so changing
significant bit to unsigned. The UBSAN warning calltrace like below:
UBSAN: shift-out-of-bounds in sound/core/seq/seq_clientmgr.c:509:22
left shift of 1 by 31 places cannot be represented in type 'int'
...
Call Trace:
<TASK>
dump_stack_lvl+0x8d/0xcf
ubsan_epilogue+0xa/0x44
__ubsan_handle_shift_out_of_bounds+0x1e7/0x208
snd_seq_deliver_single_event.constprop.21+0x191/0x2f0
snd_seq_deliver_event+0x1a2/0x350
snd_seq_kernel_client_dispatch+0x8b/0xb0
snd_seq_client_notify_subscription+0x72/0xa0
snd_seq_ioctl_subscribe_port+0x128/0x160
snd_seq_kernel_client_ctl+0xce/0xf0
snd_seq_oss_create_client+0x109/0x15b
alsa_seq_oss_init+0x11c/0x1aa
do_one_initcall+0x80/0x440
kernel_init_freeable+0x370/0x3c3
kernel_init+0x1b/0x190
ret_from_fork+0x1f/0x30
</TASK>
Baisong Zhong [Mon, 21 Nov 2022 11:00:44 +0000 (19:00 +0800)]
ALSA: pcm: fix undefined behavior in bit shift for SNDRV_PCM_RATE_KNOT
Shifting signed 32-bit value by 31 bits is undefined, so changing
significant bit to unsigned. The UBSAN warning calltrace like below:
UBSAN: shift-out-of-bounds in sound/core/pcm_native.c:2676:21
left shift of 1 by 31 places cannot be represented in type 'int'
...
Call Trace:
<TASK>
dump_stack_lvl+0x8d/0xcf
ubsan_epilogue+0xa/0x44
__ubsan_handle_shift_out_of_bounds+0x1e7/0x208
snd_pcm_open_substream+0x9f0/0xa90
snd_pcm_oss_open.part.26+0x313/0x670
snd_pcm_oss_open+0x30/0x40
soundcore_open+0x18b/0x2e0
chrdev_open+0xe2/0x270
do_dentry_open+0x2f7/0x620
path_openat+0xd66/0xe70
do_filp_open+0xe3/0x170
do_sys_openat2+0x357/0x4a0
do_sys_open+0x87/0xd0
do_syscall_64+0x34/0x80
Jaroslav Kysela [Tue, 8 Nov 2022 11:59:14 +0000 (12:59 +0100)]
selftests: alsa - add PCM test
This initial code does a simple sample transfer tests. By default,
all PCM devices are detected and tested with short and long
buffering parameters for 4 seconds. If the sample transfer timing
is not in a +-100ms boundary, the test fails. Only the interleaved
buffering scheme is supported in this version.
The configuration may be modified with the configuration files.
A specific hardware configuration is detected and activated
using the sysfs regex matching. This allows to use the DMI string
(/sys/class/dmi/id/* tree) or any other system parameters
exposed in sysfs for the matching for the CI automation.
The configuration file may also specify the PCM device list to detect
the missing PCM devices.
v1..v2:
- added missing alsa-local.h header file
Cc: Mark Brown <broonie@kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart@intel.com> Cc: Liam Girdwood <liam.r.girdwood@intel.com> Cc: Jesse Barnes <jsbarnes@google.com> Cc: Jimmy Cheng-Yi Chiang <cychiang@google.com> Cc: Curtis Malainey <cujomalainey@google.com> Cc: Brian Norris <briannorris@chromium.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz> Reviewed-by: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20221108115914.3751090-1-perex@perex.cz Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kuninori Morimoto [Wed, 16 Nov 2022 00:12:39 +0000 (00:12 +0000)]
ALSA: pcm: avoid nused-but-set-variable warning
It will indicate below warning if W=1 was added and CONFIG_SND_DEBUG
was not set. This patch adds __maybe_unused and avoid it.
${LINUX}/sound/core/pcm_native.c: In function 'constrain_mask_params':
${LINUX}/sound/core/pcm_native.c:291:25: error: variable 'old_mask' set but not used [-Werror=unused-but-set-variable]
291 | struct snd_mask old_mask;
| ^~~~~~~~
${LINUX}/sound/core/pcm_native.c: In function 'constrain_interval_params':
${LINUX}/sound/core/pcm_native.c:327:29: error: variable 'old_interval' set but not used [-Werror=unused-but-set-variable]
327 | struct snd_interval old_interval;
| ^~~~~~~~~~~~
${LINUX}/sound/core/pcm_native.c: In function 'constrain_params_by_rules':
${LINUX}/sound/core/pcm_native.c:368:29: error: variable 'old_interval' set but not used [-Werror=unused-but-set-variable]
368 | struct snd_interval old_interval;
| ^~~~~~~~~~~~
${LINUX}/sound/core/pcm_native.c:367:25: error: variable 'old_mask' set but not used [-Werror=unused-but-set-variable]
367 | struct snd_mask old_mask;
| ^~~~~~~~
${LINUX}/sound/core/pcm_native.c: In function 'snd_pcm_hw_params_choose':
${LINUX}/sound/core/pcm_native.c:652:29: error: variable 'old_interval' set but not used [-Werror=unused-but-set-variable]
652 | struct snd_interval old_interval;
| ^~~~~~~~~~~~
${LINUX}/sound/core/pcm_native.c:651:25: error: variable 'old_mask' set but not used [-Werror=unused-but-set-variable]
651 | struct snd_mask old_mask;
| ^~~~~~~~
cc1: all warnings being treated as errors
make[3]: *** [${LINUX}/scripts/Makefile.build:250: sound/core/pcm_native.o] error 1
Takashi Iwai [Mon, 14 Nov 2022 14:16:58 +0000 (15:16 +0100)]
ALSA: memalloc: Allocate more contiguous pages for fallback case
Currently the fallback SG allocation tries to allocate each single
page, and this tends to result in the reverse order of memory
addresses when large space is available at boot, as the kernel takes a
free page from the top to the bottom in the zone. The end result
looks as if non-contiguous (although it actually is). What's worse is
that it leads to an overflow of BDL entries for HD-audio.
For avoiding such a problem, this patch modifies the allocation code
slightly; now it tries to allocate the larger contiguous chunks as
much as possible, then reduces to the smaller chunks only if the
allocation failed -- a similar strategy as the existing
snd_dma_alloc_pages_fallback() function.
Along with the trick, drop the unused address array from
snd_dma_sg_fallback object. It was needed in the past when
dma_alloc_coherent() was used, but with the standard page allocator,
it became superfluous and never referred.
Takashi Iwai [Tue, 15 Nov 2022 17:02:35 +0000 (18:02 +0100)]
ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book Pro 360
Samsung Galaxy Book Pro 360 (13" 2021 NP930QBD-ke1US) with codec SSID
144d:c1a6 requires the same workaround for enabling the speaker amp
like other Samsung models with ALC298 codec.
Emil Flink [Tue, 15 Nov 2022 14:45:01 +0000 (15:45 +0100)]
ALSA: hda/realtek: fix speakers for Samsung Galaxy Book Pro
The Samsung Galaxy Book Pro seems to have the same issue as a few
other Samsung laptops, detailed in kernel bug report 207423. Sound from
headphone jack works, but not the built-in speakers.
Takashi Iwai [Sat, 12 Nov 2022 14:12:23 +0000 (15:12 +0100)]
ALSA: usb-audio: Drop snd_BUG_ON() from snd_usbmidi_output_open()
snd_usbmidi_output_open() has a check of the NULL port with
snd_BUG_ON(). snd_BUG_ON() was used as this shouldn't have happened,
but in reality, the NULL port may be seen when the device gives an
invalid endpoint setup at the descriptor, hence the driver skips the
allocation. That is, the check itself is valid and snd_BUG_ON()
should be dropped from there. Otherwise it's confusing as if it were
a real bug, as recently syzbot stumbled on it.
Takashi Iwai [Sat, 12 Nov 2022 08:52:24 +0000 (09:52 +0100)]
Merge tag 'asoc-fix-v6.2-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.1
A relatively large collection of fixes and new platform quirks here,
they're all fairly minor though - the widest possible impact is the fix
to the use of prefixes on regulator names which would have broken any
device that integrates regulators with DAPM and was used in a system
where it had a name prefix assigning to it.
Takashi Iwai [Sat, 12 Nov 2022 08:47:18 +0000 (09:47 +0100)]
ALSA: memalloc: Try dma_alloc_noncontiguous() at first
The latest fix for the non-contiguous memalloc helper changed the
allocation method for a non-IOMMU system to use only the fallback
allocator. This should have worked, but it caused a problem sometimes
when too many non-contiguous pages are allocated that can't be treated
by HD-audio controller.
As a quirk workaround, go back to the original strategy: use
dma_alloc_noncontiguous() at first, and apply the fallback only when
it fails, but only for non-IOMMU case.
We'll need a better fix in the fallback code as well, but this
workaround should paper over most cases.
Ye Bin [Thu, 10 Nov 2022 14:45:39 +0000 (22:45 +0800)]
ALSA: hda: fix potential memleak in 'add_widget_node'
As 'kobject_add' may allocated memory for 'kobject->name' when return error.
And in this function, if call 'kobject_add' failed didn't free kobject.
So call 'kobject_put' to recycling resources.
Takashi Iwai [Thu, 10 Nov 2022 13:22:16 +0000 (14:22 +0100)]
ALSA: memalloc: Don't fall back for SG-buffer with IOMMU
When the non-contiguous page allocation for SG buffer allocation
fails, the memalloc helper tries to fall back to the old page
allocation methods. This would, however, result in the bogus page
addresses when IOMMU is enabled. Usually in such a case, the fallback
allocation should fail as well, but occasionally it succeeds and
hitting a bad access.
The fallback was thought for non-IOMMU case, and as the error from
dma_alloc_noncontiguous() with IOMMU essentially implies a fatal
memory allocation error, we should return the error straightforwardly
without fallback. This avoids the corner case like the above.
The patch also renames the local variable "dma_ops" with snd_ prefix
for avoiding the name conflict.
Ai Chao [Thu, 10 Nov 2022 06:34:52 +0000 (14:34 +0800)]
ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue
For Hamedal C20, the current rate is different from the runtime rate,
snd_usb_endpoint stop and close endpoint to resetting rate.
if snd_usb_endpoint close the endpoint, sometimes usb will
disconnect the device.
Jussi Laako [Tue, 8 Nov 2022 22:12:41 +0000 (00:12 +0200)]
ALSA: usb-audio: Add DSD support for Accuphase DAC-60
Accuphase DAC-60 option card supports native DSD up to DSD256,
but doesn't have support for auto-detection. Explicitly enable
DSD support for the correct altsetting.
Takashi Iwai [Tue, 8 Nov 2022 14:07:21 +0000 (15:07 +0100)]
ALSA: usb-audio: Add quirk entry for M-Audio Micro
M-Audio Micro (0762:201a) defines the descriptor as vendor-specific,
while the content seems class-compliant. Just overriding the probe
makes the device working.
Takashi Iwai [Tue, 8 Nov 2022 06:58:24 +0000 (07:58 +0100)]
ALSA: usb-audio: Remove redundant workaround for Roland quirk
The recent fix for the delayed card registration made the current
workaround for QUIRK_AUTODETECT superfluous, since the card
registration itself is delayed until the last interface probe.
This patch drops the redundant workaround in
create_autodetect_quirks() for simplification.
Takashi Iwai [Tue, 8 Nov 2022 06:58:23 +0000 (07:58 +0100)]
ALSA: usb-audio: Yet more regression for for the delayed card registration
Although we tried to fix the regression for the recent changes with
the delayed card registration, it doesn't seem covering the all
cases; e.g. on Roland EDIROL M-100FX, where the generic quirk for
Roland devices is applied, it misses the card registration because the
detection of the last interface (apparently for MIDI) fails.
This patch is an attempt to recover from those failures by calling the
card register also at the error path for the secondary interfaces.
The card register condition is also extended to match with the old
check in the previous patch, too (i.e. the simple check of the
interface number) for catching the probe with errors.
Zhu Ning [Fri, 28 Oct 2022 02:04:56 +0000 (10:04 +0800)]
ASoC: sof_es8336: reduce pop noise on speaker
The Speaker GPIO needs to be turned on slightly behind the codec turned on.
It also need to be turned off slightly before the codec turned down.
Current code uses delay in DAPM_EVENT to do it but the mdelay delays the
DAPM itself and thus has no effect. A delayed_work is added to turn on the
speaker.
The Speaker is turned off in .trigger since trigger is called slightly
before the DAPM events.
Signed-off-by: Zhu Ning <zhuning@everest-semi.com>
------------
Peter Ujfalusi [Mon, 7 Nov 2022 09:04:33 +0000 (11:04 +0200)]
ASoC: SOF: topology: No need to assign core ID if token parsing failed
Move the return value check before attempting to assign the core ID to the
swidget since we are going to fail the sof_widget_ready() and free up
swidget anyways.
Fixes: 909dadf21aae ("ASoC: SOF: topology: Make DAI widget parsing IPC agnostic") Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20221107090433.5146-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Xian Wang [Fri, 4 Nov 2022 20:29:13 +0000 (13:29 -0700)]
ALSA: hda/ca0132: add quirk for EVGA Z390 DARK
The Z390 DARK mainboard uses a CA0132 audio controller. The quirk is
needed to enable surround sound and 3.5mm headphone jack handling in
the front audio connector as well as in the rear of the board when in
stereo mode.
Page 97 of the linked manual contains instructions to setup the
controller.
Jason Montleon [Thu, 3 Nov 2022 14:46:12 +0000 (10:46 -0400)]
ASoC: rt5677: fix legacy dai naming
Starting with 6.0-rc1 the CPU DAI is not registered and the sound
card is unavailable. Adding legacy_dai_naming causes it to function
properly again.
Fixes: fc34ece41f71 ("ASoC: Refactor non_legacy_dai_naming flag") Signed-off-by: Jason Montleon <jmontleo@redhat.com> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20221103144612.4431-2-jmontleo@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
Jason Montleon [Thu, 3 Nov 2022 14:46:11 +0000 (10:46 -0400)]
ASoC: rt5514: fix legacy dai naming
Starting with 6.0-rc1 these messages are logged and the sound card
is unavailable. Adding legacy_dai_naming to the rt5514-spi causes
it to function properly again.
[ 16.928454] kbl_r5514_5663_max kbl_r5514_5663_max: ASoC: CPU DAI
spi-PRP0001:00 not registered
[ 16.928561] platform kbl_r5514_5663_max: deferred probe pending
Kai Vehmanen [Tue, 1 Nov 2022 11:49:13 +0000 (13:49 +0200)]
ASoC: SOF: ipc3-topology: use old pipeline teardown flow with SOF2.1 and older
Originally in commit b2ebcf42a48f ("ASoC: SOF: free widgets in
sof_tear_down_pipelines() for static pipelines"), freeing of pipeline
components at suspend was only done with recent FW as there were known
limitations in older firmware versions.
Tests show that if static pipelines are used, i.e. all pipelines are
setup whenever firmware is powered up, the reverse action of freeing all
components at power down, leads to firmware failures with also SOF2.0
and SOF2.1 based firmware.
The problems can be specific to certain topologies with e.g. components
not prepared to be freed at suspend (as this did not happen with older
SOF kernels).
To avoid hitting these problems when kernel is upgraded and used with an
older firmware, bump the firmware requirement to SOF2.2 or newer. If an
older firmware is used, and pipeline is a static one, do not free the
components at suspend. This ensures the suspend flow remains backwards
compatible with older firmware versions. This limitation does not apply
if the product configuration is updated to dynamic pipelines.
The limitation is not linked to firmware ABI, as the interface to free
pipeline components has been available already before ABI3.19. The
problem is in the implementation, so firmware version should be used to
decide whether it is safe to use the newer flow or not. This patch adds
a new SOF_FW_VER() macro to compare SOF firmware release versions.
Pierre-Louis Bossart [Mon, 31 Oct 2022 19:55:05 +0000 (15:55 -0400)]
ALSA: hda: clarify comments on SCF changes
The commit 1f9d3d98694b1 ("ALSA: hda - set intel audio clock to a
proper value") added a number of misleading comments.
There is no ability to detect if an SCF value was set or not, what the
code does is prevent the use of the 6MHz audio clock represented by
the value 0 in LCTL.SCF. Changing the SCF settings does require the
link to be power-cycled, but in all other cases the link is powered
automatically when exiting reset. In other words, the power-cycle is
an exception to the rule that the HDaudio legacy driver does not need
to program SPA/CPA bits.
In addition, the SCF related changes are only relevant for the first
link.
No functionality change, only comment clarifications.
Martin Povišer [Thu, 27 Oct 2022 09:58:00 +0000 (11:58 +0200)]
ASoC: tas2780: Fix set_tdm_slot in case of single slot
There's a special branch in the set_tdm_slot op for the case of nslots
being 1, but:
(1) That branch can never work (there's a check for tx_mask being
non-zero, later there's another check for it *being* zero; one or
the other always throws -EINVAL).
(2) The intention of the branch seems to be what the general other
branch reduces to in case of nslots being 1.
For those reasons remove the 'nslots being 1' special case.
Martin Povišer [Thu, 27 Oct 2022 09:57:59 +0000 (11:57 +0200)]
ASoC: tas2764: Fix set_tdm_slot in case of single slot
There's a special branch in the set_tdm_slot op for the case of nslots
being 1, but:
(1) That branch can never work (there's a check for tx_mask being
non-zero, later there's another check for it *being* zero; one or
the other always throws -EINVAL).
(2) The intention of the branch seems to be what the general other
branch reduces to in case of nslots being 1.
For those reasons remove the 'nslots being 1' special case.
Fixes: 827ed8a0fa50 ("ASoC: tas2764: Add the driver for the TAS2764") Suggested-by: Jos Dehaes <jos.dehaes@gmail.com> Signed-off-by: Martin Povišer <povik+lin@cutebit.org> Link: https://lore.kernel.org/r/20221027095800.16094-2-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie@kernel.org>
Martin Povišer [Thu, 27 Oct 2022 09:57:58 +0000 (11:57 +0200)]
ASoC: tas2770: Fix set_tdm_slot in case of single slot
There's a special branch in the set_tdm_slot op for the case of nslots
being 1, but:
(1) That branch can never work (there's a check for tx_mask being
non-zero, later there's another check for it *being* zero; one or
the other always throws -EINVAL).
(2) The intention of the branch seems to be what the general other
branch reduces to in case of nslots being 1.
For those reasons remove the 'nslots being 1' special case.
Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Suggested-by: Jos Dehaes <jos.dehaes@gmail.com> Signed-off-by: Martin Povišer <povik+lin@cutebit.org> Link: https://lore.kernel.org/r/20221027095800.16094-1-povik+lin@cutebit.org Signed-off-by: Mark Brown <broonie@kernel.org>
Chen Zhongjin [Fri, 28 Oct 2022 03:16:03 +0000 (11:16 +0800)]
ASoC: core: Fix use-after-free in snd_soc_exit()
KASAN reports a use-after-free:
BUG: KASAN: use-after-free in device_del+0xb5b/0xc60
Read of size 8 at addr ffff888008655050 by task rmmod/387
CPU: 2 PID: 387 Comm: rmmod
Hardware name: QEMU Standard PC (i440FX + PIIX, 1996)
Call Trace:
<TASK>
dump_stack_lvl+0x79/0x9a
print_report+0x17f/0x47b
kasan_report+0xbb/0xf0
device_del+0xb5b/0xc60
platform_device_del.part.0+0x24/0x200
platform_device_unregister+0x2e/0x40
snd_soc_exit+0xa/0x22 [snd_soc_core]
__do_sys_delete_module.constprop.0+0x34f/0x5b0
do_syscall_64+0x3a/0x90
entry_SYSCALL_64_after_hwframe+0x63/0xcd
...
</TASK>
It's bacause in snd_soc_init(), snd_soc_util_init() is possble to fail,
but its ret is ignored, which makes soc_dummy_dev unregistered twice.
snd_soc_init()
snd_soc_util_init()
platform_device_register_simple(soc_dummy_dev)
platform_driver_register() # fail
platform_device_unregister(soc_dummy_dev)
platform_driver_register() # success
...
snd_soc_exit()
snd_soc_util_exit()
# soc_dummy_dev will be unregistered for second time
To fix it, handle error and stop snd_soc_init() when util_init() fail.
Also clean debugfs when util_init() or driver_register() fail.
Takashi Iwai [Thu, 27 Oct 2022 06:52:33 +0000 (08:52 +0200)]
ALSA: aoa: Fix I2S device accounting
i2sbus_add_dev() is supposed to return the number of probed devices,
i.e. either 1 or 0. However, i2sbus_add_dev() has one error handling
that returns -ENODEV; this will screw up the accumulation number
counted in the caller, i2sbus_probe().
Fix the return value to 0 and add the comment for better understanding
for readers.
Instead just use del_timer_sync() which will wait for the timer to finish
before continuing. No need to check if the timer is active or not when
doing so.
This doesn't fix the race of a possible re-arming of the timer, but at
least it won't use the data that has just been freed.
Yang Yingliang [Thu, 27 Oct 2022 01:34:38 +0000 (09:34 +0800)]
ALSA: aoa: i2sbus: fix possible memory leak in i2sbus_add_dev()
dev_set_name() in soundbus_add_one() allocates memory for name, it need be
freed when of_device_register() fails, call soundbus_dev_put() to give up
the reference that hold in device_initialize(), so that it can be freed in
kobject_cleanup() when the refcount hit to 0. And other resources are also
freed in i2sbus_release_dev(), so it can return 0 directly.
Takashi Iwai [Thu, 27 Oct 2022 06:26:32 +0000 (08:26 +0200)]
Merge tag 'asoc-fix-v6.1-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.1
Quite a few fixes here, a lot driver specific, plus some new quirks.
There was a bit of a mess with the runtime PM handling due to some
confusion in the API there which resulted in a number of commits and
reverts but that should all be stable now.
This model requires an additional detection quirk to enable the
internal microphone - BIOS doesn't seem to support AcpDmicConnected
(nothing in acpidump output).
Paul Cercueil [Tue, 25 Oct 2022 15:01:49 +0000 (16:01 +0100)]
ASoC: dapm: Don't use prefix for regulator name
When a component has a prefix, and uses a SND_SOC_DAPM_REGULATOR_SUPPLY,
the name of the regulator should not use the prefix, otherwise it won't
be properly matched in the DT/ACPI.
Jason A. Donenfeld [Tue, 25 Oct 2022 00:03:13 +0000 (02:03 +0200)]
ALSA: rme9652: use explicitly signed char
With char becoming unsigned by default, and with `char` alone being
ambiguous and based on architecture, signed chars need to be marked
explicitly as such. This fixes warnings like:
Jason A. Donenfeld [Mon, 24 Oct 2022 16:29:29 +0000 (18:29 +0200)]
ALSA: au88x0: use explicitly signed char
With char becoming unsigned by default, and with `char` alone being
ambiguous and based on architecture, signed chars need to be marked
explicitly as such. This fixes warnings like:
Takashi Iwai [Fri, 21 Oct 2022 12:27:22 +0000 (14:27 +0200)]
ALSA: usb-audio: Add quirks for M-Audio Fast Track C400/600
M-Audio Fast Track C400 and C600 devices (0763:2030 and 0763:2031,
respectively) seem requiring the explicit setup for the implicit
feedback mode. This patch adds the quirk entries for those.
Yang Yingliang [Thu, 20 Oct 2022 11:01:57 +0000 (19:01 +0800)]
ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init()
If snd_hdac_device_register() fails, 'codec' and name allocated in
dev_set_name() called in snd_hdac_device_init() are leaked. Fix this
by calling put_device(), so they can be freed in snd_hda_codec_dev_release()
and kobject_cleanup().
Fixes: 829c67319806 ("ASoC: SOF: Intel: Introduce HDA codec init and exit routines") Fixes: dfe66a18780d ("ALSA: hdac_ext: add extended HDA bus") Signed-off-by: Yang Yingliang <yangyingliang@huawei.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20221020110157.1450191-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Lenovo Thinkbook 14+ 2022 (ThinkBook 14 G4+ ARA) uses Ryzen
6000 processor, and has the same microphone problem as other
ThinkPads with AMD Ryzen 6000 series CPUs, which has been
listed in https://bugzilla.kernel.org/show_bug.cgi?id=216267.
Adding 21D0 to quirks table solves this microphone problem
for ThinkBook 14 G4+ ARA.
Yang Yingliang [Thu, 20 Oct 2022 10:59:37 +0000 (18:59 +0800)]
ASoC: Intel: Skylake: fix possible memory leak in skl_codec_device_init()
If snd_hdac_device_register() fails, 'codec' and name allocated in
dev_set_name() called in snd_hdac_device_init() are leaked. Fix this
by calling put_device(), so they can be freed in snd_hda_codec_dev_release()
and kobject_cleanup().
Fixes: e4746d94d00c ("ASoC: Intel: Skylake: Introduce HDA codec init and exit routines") Signed-off-by: Yang Yingliang <yangyingliang@huawei.com> Suggested-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20221020105937.1448951-1-yangyingliang@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Maciej S. Szmigiero [Thu, 20 Oct 2022 20:46:26 +0000 (22:46 +0200)]
ALSA: ac97: Use snd_ctl_rename() to rename a control
With the recent addition of hashed controls lookup it's not enough to just
update the control name field, the hash entries for the modified control
have to be updated too.
snd_ctl_rename() takes care of that, so use it instead of directly
modifying the control name.
While we are at it, check also that the new control name doesn't
accidentally overwrite the available buffer space.
Maciej S. Szmigiero [Thu, 20 Oct 2022 20:46:25 +0000 (22:46 +0200)]
ALSA: ca0106: Use snd_ctl_rename() to rename a control
With the recent addition of hashed controls lookup it's not enough to just
update the control name field, the hash entries for the modified control
have to be updated too.
snd_ctl_rename() takes care of that, so use it instead of directly
modifying the control name.
Maciej S. Szmigiero [Thu, 20 Oct 2022 20:46:24 +0000 (22:46 +0200)]
ALSA: emu10k1: Use snd_ctl_rename() to rename a control
With the recent addition of hashed controls lookup it's not enough to just
update the control name field, the hash entries for the modified control
have to be updated too.
snd_ctl_rename() takes care of that, so use it instead of directly
modifying the control name.
Maciej S. Szmigiero [Thu, 20 Oct 2022 20:46:23 +0000 (22:46 +0200)]
ALSA: hda/realtek: Use snd_ctl_rename() to rename a control
With the recent addition of hashed controls lookup it's not enough to just
update the control name field, the hash entries for the modified control
have to be updated too.
snd_ctl_rename() takes care of that, so use it instead of directly
modifying the control name.
Maciej S. Szmigiero [Thu, 20 Oct 2022 20:46:22 +0000 (22:46 +0200)]
ALSA: usb-audio: Use snd_ctl_rename() to rename a control
With the recent addition of hashed controls lookup it's not enough to just
update the control name field, the hash entries for the modified control
have to be updated too.
snd_ctl_rename() takes care of that, so use it instead of directly
modifying the control name.
Maciej S. Szmigiero [Thu, 20 Oct 2022 20:46:21 +0000 (22:46 +0200)]
ALSA: control: add snd_ctl_rename()
Add a snd_ctl_rename() function that takes care of updating the control
hash entries for callers that already have the relevant struct snd_kcontrol
at hand and hold the control write lock (or simply haven't registered the
card yet).
Pierre-Louis Bossart [Wed, 19 Oct 2022 16:21:15 +0000 (11:21 -0500)]
ALSA/ASoC: hda: move SPIB/DRMS functionality from ext layer
The SPIB and DRMS capabilities are orthogonal to the DSP enablement
and can be used whether the stream is coupled or not.
The existing code partitioning makes limited sense, the capabilities
are parsed at the sound/hda level but helpers are located in
sound/hda/ext.
This patch moves all the SPIB/DRMS functionality to the sound/hda
layer. This reduces the complexity of the sound/hda/ext layer which is
now limited to handling the multi-link extensions and stream
coupling/decoupling helpers.
Note that this is an iso-functionality code move and rename, the
HDaudio legacy driver would need additional changes to make use of
these capabilities.
commit 0b00a5615dc40 ("ALSA: hdac_ext: add hdac extended controller")
introduced a for() loop on the number of HDaudio codecs that seems
completely useless.
a) the body of the loop does not make use of the loop index, and
b) the LSDIID register is related to the SDI line, so there can only
be one codec per multi-link descriptor.
Pierre-Louis Bossart [Wed, 19 Oct 2022 16:21:13 +0000 (11:21 -0500)]
ALSA: hda: ext: reduce ambiguity between 'multi-link' and 'link' DMA
My esteemed colleagues keep using the same words for different things.
The multi-link structure needs to be handled whether the DSP is
enabled or not.
The host and link DMAs are only relevant when the DSP is enabled.
Things get convoluted when there's an ambiguity between the LOSIDV
settings in the multi-link register space and the selection of the
stream_tag for the link DMA.
Clarify with a rename that the static functions used are related to
the host and link DMAs only.
Pierre-Louis Bossart [Wed, 19 Oct 2022 16:21:12 +0000 (11:21 -0500)]
ALSA/ASoC: hda: ext: add 'bus' prefix for multi-link stream setting
All the helpers dealing with multi-link configurations are located in
the hdac_ext_controller.c, except the two set/clear routines that
modify the LOSIDV registers.
For consistency, move the two helpers and add the 'bus' prefix. One
could argue that the 'ml' prefix might be more relevant but that would
be a larger code change.
Pierre-Louis Bossart [Wed, 19 Oct 2022 16:21:11 +0000 (11:21 -0500)]
ALSA/ASoC: hda: ext: remove 'link' prefix for stream-related operations
We should only use 'link' in the context of multi-link
configurations. Streams are configured from a different register space
and are not dependent on link except for LOSIDV settings.
Yang Yingliang [Wed, 19 Oct 2022 09:30:25 +0000 (17:30 +0800)]
ALSA: ac97: fix possible memory leak in snd_ac97_dev_register()
If device_register() fails in snd_ac97_dev_register(), it should
call put_device() to give up reference, or the name allocated in
dev_set_name() is leaked.
Mark Brown [Wed, 19 Oct 2022 15:37:01 +0000 (16:37 +0100)]
ASoC: jz4752b: Capture fixes
Merge series from Siarhei Volkau <lis8215@gmail.com>:
The patchset fixes:
- Line In path stays powered off during capturing or
bypass to mixer.
- incorrectly represented dB values in alsamixer, et al.
- incorrect represented Capture input selector in alsamixer
in Playback tab.
- wrong control selected as Capture Master
Aidan MacDonald [Wed, 19 Oct 2022 01:23:02 +0000 (02:23 +0100)]
ASoC: simple-card: Fix up checks for HW param fixups
The "convert-xxx" properties only have an effect for DPCM DAI links.
A DAI link is only created as DPCM if the device tree requires it;
part of this involves checking for the use of "convert-xxx" properties.
When the convert-sample-format property was added, the checks got out
of sync. A DAI link that specified only convert-sample-format but did
not pass any of the other DPCM checks would not go into DPCM mode and
the convert-sample-format property would be silently ignored.
Fix this by adding a function to do the "convert-xxx" property checks,
instead of open-coding it in simple-card and audio-graph-card. And add
"convert-sample-format" to the check function so that DAI links using
it will be initialized correctly.
Fixes: 047a05366f4b ("ASoC: simple-card-utils: Fixup DAI sample format") Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com> Acked-by: Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/20221019012302.633830-1-aidanmacdonald.0x0@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Vehmanen [Tue, 18 Oct 2022 12:13:32 +0000 (15:13 +0300)]
ASoC: SOF: ipc4-mtrace: protect per-core nodes against multiple open
Add protection against multiple open of the mtrace/coreN debugfs
nodes. This is not supported in the implementation, and this will
show up as unexpected behaviour of the interface, and potential
use of already freed memory.
Fixes: f4ea22f7aa75 ("ASoC: SOF: ipc4: Add support for mtrace log extraction") Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20221018121332.20802-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
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