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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.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20221201170745.1111236-4-broonie@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20221201170745.1111236-3-broonie@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20221201170745.1111236-2-broonie@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Following a commit 1dd0dd0b1f ("ALSA: firewire: Remove some left-over
license text in sound/firewire"), this patch removes it added carelessly.
Fixes: 2133dc91d6 ("ALSA: dice: add support for Focusrite Saffire Pro 40 with TCD3070 ASIC")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20221201030100.31495-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20221130143313.43880-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
There is a HP ProBook which using ALC236 codec and need the
ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF quirk to make mute LED and
micmute LED work.
Signed-off-by: Andy Chi <andy.chi@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221128022849.13759-1-andy.chi@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Strings need to be specially marked in trace events to ensure the
content is captured, othewise the trace just shows the value of the
pointer.
Signed-off-by: John Keeping <john@metanate.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20221125162327.297440-1-john@metanate.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pci_disable_device() need be called while module exiting, switch to use
pcim_enable(), pci_disable_device() will be called in pcim_release().
Fixes: 3285ea10e9 ("ALSA: asihpi - Interrelated HPI tidy up.")
Signed-off-by: Liu Shixin <liushixin2@huawei.com>
Link: https://lore.kernel.org/r/20221126021429.3029562-1-liushixin2@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The newly added PCM test produces a binary which is not ignored by git
when built in tree, fix that.
Fixes: aba51cd094 ("selftests: alsa - add PCM test")
Signed-off-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20221125153654.1037868-1-broonie@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Signed-off-by: Baisong Zhong <zhongbaisong@huawei.com>
Link: https://lore.kernel.org/r/20221121111630.3119259-1-zhongbaisong@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Signed-off-by: Baisong Zhong <zhongbaisong@huawei.com>
Link: https://lore.kernel.org/r/20221121110044.3115686-1-zhongbaisong@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/874juzg3kd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Fixes: a8d302a0b7 ("ALSA: memalloc: Revive x86-specific WC page allocations again")
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20221114141658.29620-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
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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.
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.
Fixes: 9736a32513 ("ALSA: memalloc: Don't fall back for SG-buffer with IOMMU")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Link: https://lore.kernel.org/r/CAHk-=wgSH5ubdvt76gNwa004ooZAEJL_1Q-Fyw5M2FDdqL==dg@mail.gmail.com
Link: https://lore.kernel.org/r/20221112084718.3305-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Ye Bin <yebin10@huawei.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221110144539.2989354-1-yebin@huaweicloud.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Fixes: a8d302a0b7 ("ALSA: memalloc: Revive x86-specific WC page allocations again")
Reported-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.22.394.2211041541090.3532114@eliteleevi.tm.intel.com
Link: https://lore.kernel.org/r/20221110132216.30605-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Ai Chao <aichao@kylinos.cn>
Link: https://lore.kernel.org/r/20221110063452.295110-1-aichao@kylinos.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221108221241.1220878-1-jussi@sonarnerd.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We are able to power down the GPU and audio via the GPU driver
so flag these asics as supporting runtime pm.
Signed-off-by: Evan Quan <evan.quan@amd.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20221108084746.583058-1-evan.quan@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Fixes: 39efc9c8a9 ("ALSA: usb-audio: Fix last interface check for registration")
Link: https://bugzilla.suse.com/show_bug.cgi?id=1205111
Link: https://lore.kernel.org/r/20221108065824.14418-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Fixes: 39efc9c8a9 ("ALSA: usb-audio: Fix last interface check for registration")
Cc: <stable@vger.kernel.org>
Link: https://bugzilla.suse.com/show_bug.cgi?id=1205111
Link: https://lore.kernel.org/r/20221108065824.14418-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
------------
v1: cancel delayed work while disabling speaker.
Link: https://lore.kernel.org/r/20221028020456.90286-1-zhuning0077@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 909dadf21a ("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>
snd_soc_util_exit() is called in __init snd_soc_init() for cleanup.
Remove the __exit annotation for it to fix the build warning:
WARNING: modpost: sound/soc/snd-soc-core.o: section mismatch in reference: init_module (section: .init.text) -> snd_soc_util_exit (section: .exit.text)
Fixes: 6ec27c5388 ("ASoC: core: Fix use-after-free in snd_soc_exit()")
Signed-off-by: Chen Zhongjin <chenzhongjin@huawei.com>
Link: https://lore.kernel.org/r/20221031134031.256511-1-chenzhongjin@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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: fc34ece41f ("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>
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
Fixes: fc34ece41f ("ASoC: Refactor non_legacy_dai_naming flag")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216641
Signed-off-by: Jason Montleon <jmontleo@redhat.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221103144612.4431-1-jmontleo@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Originally in commit b2ebcf42a4 ("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.
Link: https://github.com/thesofproject/sof/issues/6475
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221101114913.1292671-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 1f9d3d9869 ("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.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20221031195505.249929-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: eae9f9ce18 ("ASoC: add tas2780 driver")
Suggested-by: Jos Dehaes <jos.dehaes@gmail.com>
Signed-off-by: Martin Povišer <povik+lin@cutebit.org>
Link: https://lore.kernel.org/r/20221027095800.16094-3-povik+lin@cutebit.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 827ed8a0fa ("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>
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: 1a476abc72 ("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>
