An explicit bool conversion operator will still be used implicitly
when an expression appears in "bool context", e.g., as the condition
in an if statement, or as argument to logical operators. The
`explicit` annotation prevents conversion in other contexts, e.g.,
converting both a and b to bool in an expression like `a == b`.
Bug: None
Change-Id: I79ef35b1ea831e6011ae472900375ae8a3e617ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250664
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35927}
This fixes a crash introduced with recent move of Scoped class for
glib objects into a separated implementation.
Bug: chromium:1291247
Change-Id: I49d56bc0811f52434213516f51ca9e8712692e15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250840
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35922}
Adds missing RTC_NO_SANITIZE("cfi-icall") attributes to a few needed
functions for PipeWire initialization. These are methods that call (or
call methods that end up inlined and call) function pointers as a result
of dlopen'ing a lib. For ShareScreencastStream, the generated
InitializeStubs method appears to trigger this; while the egl_dmabuf
destructor appears to need this due to the EglDestroyContext and
EglTerminate calls that it makes.
Bug: webrtc:13659
Change-Id: Idb4af985293224957a50d17d9042524af2b66138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250702
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35918}
Moves the `Scoped` template (meant for clearing up the references) into
separate utils so as to allow for reuse in future. Other portal instances
e.g. remote desktop portal will benefit from this later.
Bug: chromium:1291247
Change-Id: Ie36415573edcbe4f697cf97b568243f09f26915d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249400
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#35916}
When the minimum mic level is overridden via the field trial named
WebRTC-Audio-AgcMinMicLevelExperiment, AGC1 can still lower the gain
beyond the minimum value (namely, when clipping is observed).
This CL changes the behavior of the field trial. When specified, the
override always applies and therefore the mic level is guaranteed to
never become lower than what the field trial specifies.
Tested: RTC call in Chromium with and without --force-fieldtrials="
WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-255"
Bug: chromium:1275566
Change-Id: I42ff45add54c11084f5ca6a2b95887c627c3c3aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250141
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35914}
- Switch from ptr+size to rtc::ArrayView
- Remove `AgcManagerDirect::sample_rate_hz_` since it's always 16 kHz
- Stop passing nullptr in agc_manager_direct_unittest.cc when
`AgcManagerDirect::Process()` is called
- Allow to correctly run the tests added in the child CL (see [1])
[1] https://webrtc-review.googlesource.com/c/src/+/250141
Bug: webrtc:7494
Change-Id: I0292d7038d6510ca7c58af32b6003a1e4b121b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35910}
The 6-parameter Initialize method is removed. The has_keyboard parameter
in the StreamConfig constructor is removed together with the underlying
member and helper functions.
Bug: chromium:1271981, b/217349489
Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221
Reviewed-by: Sam Zackrisson <saza@google.com>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35908}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
This is a reland of 9d230d54c7eef31ac1100f0aeef1374dd1ac62fa
Original change's description:
> (Un/)Subscribe RtpVideoSender for feedback on the transport queue.
>
> * RtpVideoSender now registers/unregisters for feedback callback
> inside of SetActive(), which runs on the transport queue.
> * Transport feedback is given on the transport queue
> * Registration/unregistration for feedback is done on the same
> * Removed the last mutex from TransportFeedbackDemuxer.
>
> Ultimately, this work is related to moving state from the Call
> class, that's related to network configuration, but due to the code
> is currently written is attached to the worker thread, over to the
> Transport, where it's used (e.g. suspended_video_send_ssrcs_).
>
> Bug: webrtc:13517, webrtc:11993
> Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35777}
Bug: webrtc:13517, webrtc:11993
Change-Id: I766e569abea8bae96d32267a951fcdc195ced8a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249782
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35863}
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
This emulates behaviour from frame buffer 2, but does not handle stats.
In contrast to frame buffer 2, all work happens on the same task queue.
FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind
a field trial WebRTC-FrameBuffer3.
This separates frame scheduling behaviour into a few components,
VideoReceiveStreamTimeoutTracker
* Handles the stream timeouts.
FrameDecodeScheduler
* Manages the scheduling and cancelling of frames being sent to the
decoder.
FrameDecodeTiming
* Handles the timing and ordering of frames to be decoded.
Other changes
* Adds CurrentSize() method to FrameBuffer3
* Move timing to a separate library
* Does a thread check for Receive statistics as this is now
on the worker thread.
* Adds `FlushImmediate` method to RunLoop so that
video_receive_stream2_unittest can pass when scheduling is happening
on the worker thread.
Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721
Bug: webrtc:13343
Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35847}
This reverts commit 3babb8af238a531cbff27951604b09bb78b762cd.
Reason for revert:
- Causes regressions to transceivers, see https://crbug.com/1291956 for more information, including tests to reproduce the issue.
This CL is not a pure revert. While it reverts everything else, it does
keep the new enum value (kProfilePredictiveHigh444). This is as to not
break Chromium which already depend on it. It is not listed in the
kProfilePatterns though so the enum value should never be applicable.
Original change's description:
> Added support for H264 YUV444 (I444) decoding.
>
> Added Nutanix Inc. to the AUTHORS file.
>
> PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/234540
>
> Bug: chromium:1251096
> Change-Id: I99a1b1e4d8b60192ff96f92334a430240875c66c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235340
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35684}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1251096, chromium:1291956
Change-Id: Ib4d8ea4898f9832914d88e7076e6b39da0c804ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249791
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35835}
1) Fixes crash on dlclose when using NVidia driver
2) Closes EGLDisplay and EGLContext on destruction
3) Prints correct errors for EGL calls
Bug: chromium:1290566
Change-Id: Icfb3cad2e7c054030821479be7e48d77a4e0d5e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249795
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35833}
Running bitexactness tests only on Linux makes it significantly easier to
update them, while still giving many of the same benefits.
Bug: webrtc:12518, b/216736217
Change-Id: I7f3c9a27c0fc14b7ee0e83aede2e7702cfa79141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249787
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35829}
This reverts commit 9d230d54c7eef31ac1100f0aeef1374dd1ac62fa.
Reason for revert: Speculative revert to see if it's the cause of a few perf changes (some bad, some not so bad).
Bug: webrtc:13613
Original change's description:
> (Un/)Subscribe RtpVideoSender for feedback on the transport queue.
>
> * RtpVideoSender now registers/unregisters for feedback callback
> inside of SetActive(), which runs on the transport queue.
> * Transport feedback is given on the transport queue
> * Registration/unregistration for feedback is done on the same
> * Removed the last mutex from TransportFeedbackDemuxer.
>
> Ultimately, this work is related to moving state from the Call
> class, that's related to network configuration, but due to the code
> is currently written is attached to the worker thread, over to the
> Transport, where it's used (e.g. suspended_video_send_ssrcs_).
>
> Bug: webrtc:13517, webrtc:11993
> Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35777}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13517, webrtc:11993
Change-Id: I824623b3b1c14f0ca7049a2a0890c6d97b7fb608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249600
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35815}
This is a delegate that is used by video_receive_stream2 to handle frame
buffer tasks like threading, and stats. This will be used in a follow up
to use FrameBuffer3 as a strategy selected by field trial.
Unit-tests will be used in follow-up CLs containing Frame Buffer 3, and
are expected to work with both Frame buffer proxy versions.
Change-Id: I524279343d60a348d044d9085d618f12d7bf3a23
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241605
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35803}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
Just applied a short sed script. See bug description for
the motiviation for this change.
This is the command that was used to generate the changes:
$ find . -type f \( -iname '*.cc' -o -iname '*.h' \) -print0 | \
xargs -0 sed -i -e 's/(const override)/(const, override)/'
Bug: webrtc:13090
Change-Id: Iec7d280f9d55263a972dbb3bd644ebfcd2eb38cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249088
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35801}
This adds a field trial to change the pacing rate to pace
at a rate relative to max(bwe ,lower link capacity)
Bug: none
Change-Id: Ibe9ef3e08eb422e9abff6488780e82188958eeeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248865
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35795}
This unblocks lowering the precision of low precision tasks which are
the default.
Bug: webrtc:13604
Change-Id: Icd663cbbf5b0bf87ac83a4a0abd58699e6e27e8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248862
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35782}
Use cases of TaskQueue or TaskQueueBase that are considered high
precision are updated to make use of PostDelayedHighPrecisionTask
(see go/postdelayedtask-precision-in-webrtc) instead of PostDelayedTask.
The cases here are the ones covered by that document, plus some
testing-only uses. The FrameBuffer2 and DataTracker use cases will
be covered by separate CLs because FrameBuffer2 uses
RepeatingTaskHandle and DataTracker uses dcsctp::Timer.
This protects these use cases against regressions when PostDelayedTask
gets its precision lowered.
This CL also adds TaskQueue::PostDelayedHighPrecisionTask which calls
TaskQueueBase::PostDelayedHighPrecisionTask (same pattern as for
PostDelayedTask).
Bug: webrtc:13604
Change-Id: I7dcab59cbe4d274d27b734ceb4fc06daa12ffd0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248864
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35781}
* RtpVideoSender now registers/unregisters for feedback callback
inside of SetActive(), which runs on the transport queue.
* Transport feedback is given on the transport queue
* Registration/unregistration for feedback is done on the same
* Removed the last mutex from TransportFeedbackDemuxer.
Ultimately, this work is related to moving state from the Call
class, that's related to network configuration, but due to the code
is currently written is attached to the worker thread, over to the
Transport, where it's used (e.g. suspended_video_send_ssrcs_).
Bug: webrtc:13517, webrtc:11993
Change-Id: I057d0e2597e6cb746b335e0308599cd547350e56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248165
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35777}
Before, this call was being made from the SendPacket path of the
pacer. The transport will post a task to the transport queue regardless
so this change moves the lock inside of the demuxer away from the
pacer and over to the transport queue.
Moving forward, the calls to register/unregister with the feedback
demuxer, will occur on the transport queue as well and we can change
the transport OnTransportFeedback() implementation to forward the
calls to the demuxer on the transport queue as well. That will bring
all calls into the same execution context and we won't need a lock.
Bug: webrtc:13517, webrtc:11993
Change-Id: If714ca6d2b164a1a2b6bcb8c99787372064a31a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248164
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35775}
After the refactoring, the test fixture is only used for creating the
object under test and dependencies. This leads to more readable code and
allows more flexibility when creating the object under test.
Bug: none
Change-Id: I643330290da17efe0a02fe5dc6b884136705de0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/248140
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35770}
Make PipeWire stream shared through DesktopCaptureOptions (similar to
X11 implementation sharing XDisplay) so we can implement better cursor
support with our own MouseCursorMonitor implementation.
Bug: webrtc:13429
Change-Id: I781482aa29cee0c105c42e5109f28e95dde9881b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238174
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35765}
When disabled, the test ResolutionAdaptsToAvailableBandwidth fails when
using frame buffer3. It is not clear if that is a problem with the test
or if that behaviour is required, and thus it is safer to have this
enabled by default and experiment with turning it off in the future.
Change-Id: I7a6ae14c37a0cdc3e203f39f6cc0c3ad87038a60
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247700
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35764}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.
They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.
Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.
Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
A number of utility functions to be shared between frame buffer 2
and the new frame scheduling implementation based on frame buffer 3.
Change-Id: Icc932c6c76fddeeedc8aa64ec27c7e0c955abfd0
Bug: webrtc:13343
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241604
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35743}
This CL updates both the static GOF pattern with the correct flags for
temporal_up_switch, as well the flexible mode logic to base the flag
on dependency descriptors instead use reference buffers.
Bug: webrtc:13576
Change-Id: I578f744bec51d1f3531da5f4a89d12f05a16a6c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247187
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35741}
