Update webrtc version to m89 #7

mininny · 2021-02-24T09:43:47Z

Update WebRTC version to M89(branch cut at 3489)

This reverts commit 8d4cdd1. Reason for revert: Upstream project needs have changed Original change's description: > Ignore frames that are comming to DVQA after Stop is called > > Bug: webrtc:12247 > Change-Id: Ie3e773bdff66c900956019ac3131bbdb9ee874cd > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196084 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32738} TBR=mbonadei@webrtc.org,srte@webrtc.org,landrey@webrtc.org Change-Id: Ie7483435eae9b0344f875673ca9651ff4d591bd3 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12247 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196280 Reviewed-by: Andrey Logvin <landrey@webrtc.org> Commit-Queue: Andrey Logvin <landrey@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32746}

Bug: webrtc:12247 Change-Id: I946338e0ecf58f91c87c8638977a8bc52e648fd2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196083 Commit-Queue: Andrey Logvin <landrey@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32747}

Bug: webrtc:12247 Change-Id: I608cffad7ee5397c306fb03a36d89f31882c112c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196092 Commit-Queue: Andrey Logvin <landrey@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32748}

Using the field trial WebRTC-VP9-PerformanceFlags, this CL allows you to configure the libvpx VP9 encoder with a list of flags to affect the quality vs speed tradeoff. This CL adds support for: * Speed (effort), for the temporal base layer frames * Speed for higher (non-base) layer frames * De-blocking (as part of the loopfilter) enabled for: 0 = all frames 1 = all but frames from the highest temporal layer 2 = no frames Each entry in the list has a threshold in min number of pixels needed for settings in the entry to apply. Example: Two spatial layers (180p, 360p) with three temporal layers are configured. Field trial "WebRTC-VP9-PerformanceFlags" set to: "min_pixel_count:0|129600,base_layer_speed:5|8,high_layer_speed:7|8,deblock_mode:1|2" This translates to: S0: - TL0: Speed 5, deblocked - TL1: Speed 8, deblocked - TL2: Speed 8, not deblocked S1: - TL0: Speed 7, not deblocked - TL1: Speed 8, not deblocked - TL2: Speed 8, not deblocked Bug: webrtc:11551 Change-Id: Ieef6816d3e0831ff53348ecc4a90260e2ef10422 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188461 Reviewed-by: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32749}

BUG=None No-try: True Change-Id: I6d1b9b99dce65d6382cbb61756e71989f5f3808a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195326 Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32752}

…read as argument. Bug: None Change-Id: I8b140c8cb40787473411ae55da3738166340127f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39512 Reviewed-by: Taylor <deadbeef@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32753}

Bug: None Change-Id: If2c42af6038c2ce1dc4289b949a0a3a279bae1b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195337 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32754}

This CL brings a large improvement to the RNN VAD CPU performance by finally using `VectorMath::DotProduct()` for pitch search. The realtime factor improved from about 390x to 570x for SSE2 (+180x, 45% faster) and to 610x for AVX2 (+235x, 60% faster). RNN VAD benchmark results: ``` +-----+-------+------+------+ | run | none* | SSE2 | AVX2 | +-----+-------+------+------+ | 1 | 393x | 572x | 618x | | 2 | 388x | 568x | 607x | | 3 | 393x | 564x | 599x | +-----+-------+------+------+ ``` *: baseline, no SIMD used for pitch search, but SSE2 used for the RNN Results obtained as follows: 1. Force SSE2 in `DISABLED_RnnVadPerformance` for the RNN part in order to measure the baseline correctly: ``` RnnBasedVad rnn_vad({/*sse2=*/true, /*avx2=*/true, /*neon=*/false}); ``` 2. Run the test: ``` $ ./out/release/modules_unittests \ --gtest_filter=*RnnVadTest*DISABLED_RnnVadPerformance* \ --gtest_also_run_disabled_tests --logs ``` Bug: webrtc:10480 Change-Id: I89a2bd420265540026944b9c0f1fdd4bfda7f475 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195001 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32755}

The HMM based transparent mode classifier is disabled until an issue with diverging filters is resolved. Bug: chromium:1155071 Change-Id: Iee249869f6ece1e48e834b3a4b9249c69a51286c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196341 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32756}

BUG=None Change-Id: Iabb091a10f780ff79a0ed95cf5f01ce1a0571e4f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196340 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32757}

Bug: webrtc:10480 Change-Id: I4ac8ae044261f94db7a1e9559aa61f532602b408 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195446 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32758}

I was not able to reproduce chromium:1146676 locally, so the change in merge.cc is a speculative fix. Bug: chromium:1146835, chromium:1146676, chromium:1137226 Change-Id: I14472ba5b41e58b2d5f27d9833249c14505af18f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194264 Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32759}

Bug: webrtc:12221 Change-Id: I72bb25503d2894aae596426cb40764246c087767 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196342 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32760}

This reverts commit 6e71674. Reason for revert: Unexpected perf change Original change's description: > Adds experimental libvpx VP9 speed settings. > > Using the field trial WebRTC-VP9-PerformanceFlags, this CL allows you to > configure the libvpx VP9 encoder with a list of flags to affect the > quality vs speed tradeoff. This CL adds support for: > > * Speed (effort), for the temporal base layer frames > * Speed for higher (non-base) layer frames > * De-blocking (as part of the loopfilter) enabled for: > 0 = all frames > 1 = all but frames from the highest temporal layer > 2 = no frames > > Each entry in the list has a threshold in min number of pixels needed > for settings in the entry to apply. > > Example: Two spatial layers (180p, 360p) with three temporal > layers are configured. Field trial "WebRTC-VP9-PerformanceFlags" set to: > "min_pixel_count:0|129600,base_layer_speed:5|8,high_layer_speed:7|8,deblock_mode:1|2" > This translates to: > S0: > - TL0: Speed 5, deblocked > - TL1: Speed 8, deblocked > - TL2: Speed 8, not deblocked > S1: > - TL0: Speed 7, not deblocked > - TL1: Speed 8, not deblocked > - TL2: Speed 8, not deblocked > > Bug: webrtc:11551 > Change-Id: Ieef6816d3e0831ff53348ecc4a90260e2ef10422 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188461 > Reviewed-by: Michael Horowitz <mhoro@webrtc.org> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32749} TBR=sprang@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org Change-Id: If910963441ac1a0e002aac7066791c7cc7764a1a No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11551 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196344 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32762}

Changes: 1) Scoped class This is a special class for GLib based objects which we need to manually delete with different functions. Wrapping these objects into Scoped class will destroy them automatically when they go out of scope. 2) Window sharing support Unlike screen sharing, with window sharing we are required to obtain more information from the PipeWire stream, like video crop metadata, which we use to properly set size of our buffer. 3) Support for DmaBuf and MemFd buffer types As of now, we expected the PipeWire stream will provide only plain data which we just need to copy to our buffer. We now add support for new buffer types, which are often preferred for better effeciency. 4) Minor bugfixes: a) Additionally accept PipeWire streams using alpha channels (BGRA, RGBA) b) Add lock over PipeWire loop to prevent potential issues until we fully intialize everything we need c) When obtaining buffers, make sure we work with the latest one Bug: chromium:682122 Change-Id: I64638d5dcbe18e7280550dca0b01b17c511ac98a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194100 Commit-Queue: Jamie Walch <jamiewalch@chromium.org> Reviewed-by: Jamie Walch <jamiewalch@chromium.org> Cr-Commit-Position: refs/heads/master@{#32763}

Plus a few minor code readability improvements. Bug: webrtc:10480 Change-Id: I590d8e203b1d05959a8c15373841e37abe83237e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195334 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32764}

This reverts commit 0496a41. Reason for revert: Causes unexpected changes in perf tests. Original change's description: > Add class InterArrivalDelta to goog_cc > > This cl copies modules/remote_bitrate_estimator/inter_arrival.x to inter_arrival.h and interrival_delta.cc in goog_cc in the first patchset. > In the following- this class is modified to use webrtc::Timestamp and webrtc::Timedelta in order to avoid having to use 24 bit time repressentation. > > Bug: none > Change-Id: I9befe6e3e283cf7e21efa974ae33e8a83e26cbe6 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194004 > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32733} TBR=perkj@webrtc.org,crodbro@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: none Change-Id: I725b246f6ec0c293cb3ada39b1a65a14ef9a001e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196343 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32765}

Refactoring done to more easily and cleanly add SIMD optimizations and to remove `FullyConnectedLayer` from the RNN VAD api. Minor improvements (readability, API): - `FullyConnectedLayer` gets the ActivationFunction enum and not a function view anymore - SSE2 optimization moved into `FullyConnectedLayer::ComputeOutputSse2` - layer name added for improved logs Bug: webrtc:10480 Change-Id: Ida4903a67655e19ef0464f378c433c1f6e96dca7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195444 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32766}

Adds a new "rtc_pipewire_version" build option to specify version of PipeWire we want to build against. We use version "0.2" by default which is version of PipeWire we currently have in sysroot and which is supported even on older systems like RHEL7 and Debian. Bug: chromium:1146942 Change-Id: Ib74b52fa87623a3f960e419916b01586aaeba47f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195441 Commit-Queue: Jamie Walch <jamiewalch@chromium.org> Reviewed-by: Jamie Walch <jamiewalch@chromium.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32768}

Refactoring done to more easily and cleanly add SIMD optimizations and to remove `GatedRecurrentLayer` from the RNN VAD api. Bug: webrtc:10480 Change-Id: Ie1dffdd9b19c57c03a0b634f6818c0780456a66c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195445 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32770}

This CL implements a Resource that aggressively reports overuse or underuse until the encoded stream has the max pixels specified. The pixel limit is controlled with a field trial, e.g: --force-fieldtrials="WebRTC-PixelLimitResource/Enabled-307200/" This caps the resolution to 307200 (=640x480). This can be used by the TestBed to simulate being CPU limited. Note that the resource doesn't care about degradation preference at the moment, so if the degradation preference would be set to "maintain-resolution" the PixelLimitResource would never stop reporting overuse and we would quickly get a low-FPS stream. PixelLimitResource runs a repeating task and reports overuse, underuse or neither every 5 seconds. This ensures we quickly reach the desired resolution. Unit tests are added. I did not add any integration tests (I think that's overkill for a testing-only resource) but I have manually verified that this works as intended. This CL also moves the FakeVideoStreamInputStateProvider into a test/ folder and exposes video_stream_adapter.cc's GetLowerResolutionThan(). Bug: webrtc:12261 Change-Id: Ifbf7c4c05e9dd2843543589bebef3f49b18c38c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195600 Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32771}

Bug: None Change-Id: Iba1d66d9985f449c5d3710d7d615d4b25acf763b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196360 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32772}

Bug: chromium:1155459 Change-Id: I51cb8162a989ba934e3292c86c3ecf749f26f601 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196500 Commit-Queue: Jonas Oreland <jonaso@google.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32773}

Bug: none Change-Id: I7f399449026de58dee28abcede2630269c6b95b5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196505 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32774}

guarded by a new field trial flag WebRTC-Debugging-RtpDump. Packets have a RTP_DUMP postfix for easy grep-ing. BUG=webrtc:10675 Change-Id: I73c0e0db47dca1079cd303c41a8b80fd7ae4a902 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196087 Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32775}

Bug: None Change-Id: I4294cdda93ebca44fa43de63e37d5d1fdeb2fa62 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198120 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32975}

This adds a new way to poll decoder metadata. A default implementation still delegates to the old methods. Root call site is updates to not use the olds methods. Follow-ups will dismantle usage of the olds methods in wrappers. Bug: webrtc:12271 Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32976}

SetNegotiatedHeaderExtensions_w queued a task to update the list of negotiated header extensions on the signal thread from the worker thread, in belief that a later call to GetNegotiatedHeaderExtensions() would happen on the WebRTC proxies, leading to the update happening before the readout. In downstream project, this is not always the case. Fix this by synchronously updating the list of negotiated header extensions. Bug: chromium:1051821 Change-Id: I3266292e7508bb7a22a3f7d871e82c12f60cfc83 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201728 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32977}

Bug: webrtc:12336 Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32978}

…++ code" This reverts commit 8c2250e. Reason for revert: breaks downstream project Original change's description: > Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code > > Bug: webrtc:12336 > Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32978} TBR=danilchap@webrtc.org,mbonadei@webrtc.org,crodbro@webrtc.org Change-Id: I5c9d419785254878a825865808b56841cd30b9b5 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12336 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201731 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32979}

Bug: None Change-Id: Ib8b7c57fc4dea11bd2cb2e1620916c20f6ff9985 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201623 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32980}

Bug: webrtc:11339 Change-Id: I719ff37c11d2383b83dcf06c4b882543d07678d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201725 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32981}

Change log: https://chromium.googlesource.com/chromium/src/+log/189823ba75..42ab9dc8c8 Full diff: https://chromium.googlesource.com/chromium/src/+/189823ba75..42ab9dc8c8 Changed dependencies * src/base: https://chromium.googlesource.com/chromium/src/base/+log/95b11715cf..a361323fd5 * src/build: https://chromium.googlesource.com/chromium/src/build/+log/d68ac97401..d64e5999e3 * src/ios: https://chromium.googlesource.com/chromium/src/ios/+log/ee5a91c153..e13cd29163 * src/testing: https://chromium.googlesource.com/chromium/src/testing/+log/5b9fef9564..049cd24b1f * src/third_party: https://chromium.googlesource.com/chromium/src/third_party/+log/4cb57cb45d..e35e2377db * src/third_party/depot_tools: https://chromium.googlesource.com/chromium/tools/depot_tools.git/+log/fdd89469d6..c603339365 * src/third_party/perfetto: https://android.googlesource.com/platform/external/perfetto.git/+log/8fccd82e4e..94ca9a9578 * src/tools: https://chromium.googlesource.com/chromium/src/tools/+log/c79030c622..03a8864bc6 * src/tools/luci-go: git_revision:16e6d735358b0166f06fd2e4daa0da4cff9918e9..git_revision:77944aa535e42e29faadf6cfa81aee252807d468 * src/tools/luci-go: git_revision:16e6d735358b0166f06fd2e4daa0da4cff9918e9..git_revision:77944aa535e42e29faadf6cfa81aee252807d468 * src/tools/luci-go: git_revision:16e6d735358b0166f06fd2e4daa0da4cff9918e9..git_revision:77944aa535e42e29faadf6cfa81aee252807d468 DEPS diff: https://chromium.googlesource.com/chromium/src/+/189823ba75..42ab9dc8c8/DEPS No update to Clang. TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com, BUG=None Change-Id: Ic0ff1053c1634f815e542b4bd5de7a20aa0e6067 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201782 Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com> Cr-Commit-Position: refs/heads/master@{#32982}

For the case where an unknown header extension URI is attempted to be modified by SetOfferedRtpHeaderExtensions, WebRTC emitted INVALID_PARAMETER. Fix this by emitting UNSUPPORTED_PARAMETER. Bug: chromium:1051821 Change-Id: I98b68e1e3a3f90f9cfa0d45833f46a307c246ad0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201733 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32983}

Creating a Direct3D11Device from a DXGIDevice is necessary for the new WGC based window capturer. However, the CreateDirect3DDeviceFromDXGIDevice API is not available on all versions of Windows, which means we have to load this function from d3d11.dll at runtime. You can see how this function will be used in this CL: 196624: Finish implementing WGC Window Capturer and add unit tests. | https://webrtc-review.googlesource.com/c/src/+/196624 I also ensure we don't leak HSTRINGs in GetActivationFactory and fix up some includes in ScopedComInitializer. Bug: webrtc:9273 Change-Id: Ida240b2c0f637d70ec405de7d8f59f0c12aee1f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196622 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Austin Orion <auorion@microsoft.com> Cr-Commit-Position: refs/heads/master@{#32984}

Bug: None Change-Id: Ia475afed123abaf32df6f1f1a546f5704e2d464f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201421 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32985}

This reverts commit 69241a9. Reason for revert: Breaks WebRTC roll into Chromium. Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} TBR=mbonadei@webrtc.org,hta@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. No-Try: True Bug: webrtc:9987 Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32986}

by updating flag that T1 frame can be referenced when it is encoded rather than when it is sent for encoding. Otherwise when encoder drops T1 frame, configuration for following T2 frame would still try to reference that absent T1 frame leading to invalid references. (cherry picked from commit ba91dbc) Bug: chromium:1168068 Change-Id: I6398275971596b0618bcf9c926f0282f74120976 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202030 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#33002} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202763 Cr-Commit-Position: refs/branch-heads/4389@{#1} Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}

DependencyDescriptor and vp9 wrapper understand key frame differently when it comes to the first layer frame with spatial_id>0 This CL adds and use DD's interpretation of the key frame when deciding if DD should be supported going forward. (cherry picked from commit 0be1846) Bug: webrtc:11999, chromium:1169060 Change-Id: I11a809a315e18bd856bb391576c6ea1f427e33be Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202760 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#33046} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203263 Cr-Commit-Position: refs/branch-heads/4389@{#2} Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}

Using WebRTC-VP9-PerformanceFlags and settings a multi-layer config, and then configuring the codec in non-svc mode would cause us to not set the cpu speed in libvpx. For some reason, that could trigger a crash in the encoder. This CL fixes that, and adds new test coverage for the code affected byt the trial. (cherry picked from commit 03eed7c) Bug: chromium:1167353, webrtc:11551 Change-Id: Iddb92fe03fc12bac37717908a8b5df4f3d411bf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202761 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#33051} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203521 Cr-Commit-Position: refs/branch-heads/4389@{#3} Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}

TEST: tested on chromium. (cherry picked from commit 49dbad0) No-Try: True Bug: webrtc:12397, chromium:1171206 Change-Id: I1e15605f90e253a6ef61ab7ead8c576a80e8f01b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203886 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Minyue Li <minyue@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#33080} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204302 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/branch-heads/4389@{#4} Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}

(cherry picked from commit 5c3ff6b) Bug: chromium:1155071,webrtc:12265,chromium:1155477 Change-Id: I9d3119e9cbfdd5d7b41de2ed0f9dec92f7bf753d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202258 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#33037} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205280 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/branch-heads/4389@{#5} Cr-Branched-From: 7acc2d9-refs/heads/master@{#32986}

# Conflicts: # sdk/BUILD.gn # sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h # sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm # sdk/objc/native/api/audio_device_module.h # sdk/objc/native/src/audio/audio_device_module_ios.h # tools_webrtc/ios/build_ios_libs.py

sf-nathan-park

🚀

x-0o0

LGTM

Make shader public

This reverts commit 76d3e7a. Reason for revert: Causes multiple Chromium WPT tests to crash, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/win10_chromium_x64_rel_ng/685757? Sample stack trace: #0 0x7ff8623fbde9 base::debug::CollectStackTrace() STDERR: #1 0x7ff862311ca3 [2665012:17:1009/162250.249660:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370324, interval (us) = 834 STDERR: base::debug::StackTrace::StackTrace() STDERR: #2 0x7ff8623fb93b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7ff857a70140 [2665012:17:1009/162250.249947:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652370634, interval (us) = 742 STDERR: (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7ff85778edb1 gsignal STDERR: #5 0x7ff857778537 abort STDERR: #6 0x7ff855d5eee2 [2665012:17:1009/162250.250342:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371030, interval (us) = 706 STDERR: [2665012:17:1009/162250.250514:WARNING:timestamp_aligner.cc(131)] too short translated timestamp interval: system time (us) = 3042652371204, interval (us) = 963 STDERR: rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7ff855f14e62 webrtc::LibvpxVp8Encoder::PrepareRawImagesForEncoding() STDERR: #8 0x7ff855f14412 webrtc::LibvpxVp8Encoder::Encode() STDERR: #9 0x7ff855bae765 webrtc::SimulcastEncoderAdapter::Encode() STDERR: #10 0x7ff85607d598 webrtc::VideoStreamEncoder::EncodeVideoFrame() STDERR: #11 0x7ff85607c60d webrtc::VideoStreamEncoder::MaybeEncodeVideoFrame() STDERR: #12 0x7ff8560816f5 webrtc::webrtc_new_closure_impl::ClosureTask<>::Run() STDERR: #13 0x7ff855b352b5 (anonymous namespace)::WebrtcTaskQueue::RunTask() STDERR: #14 0x7ff855b3531e base::internal::Invoker<>::RunOnce() STDERR: #15 0x7ff86239785b base::TaskAnnotator::RunTask() STDERR: #16 0x7ff8623c8557 base::internal::TaskTracker::RunSkipOnShutdown() STDERR: #17 0x7ff8623c7d92 base::internal::TaskTracker::RunTask() STDERR: #18 0x7ff862415a06 base::internal::TaskTrackerPosix::RunTask() STDERR: #19 0x7ff8623c75e6 base::internal::TaskTracker::RunAndPopNextTask() STDERR: #20 0x7ff8623d3a8d base::internal::WorkerThread::RunWorker() STDERR: #21 0x7ff8623d368d base::internal::WorkerThread::RunPooledWorker() STDERR: #22 0x7ff862416509 base::(anonymous namespace)::ThreadFunc() STDERR: #23 0x7ff857a64ea7 start_thread Original change's description: > NV12 support for VP8 simulcast > > Tested using video_loopback with generated NV12 frames. > > Bug: webrtc:11635, webrtc:11975 > Change-Id: I14b2d663c55a83d80e48e226fcf706cb18903193 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186722 > Commit-Queue: Evan Shrubsole <eshr@google.com> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32325} TBR=ilnik@webrtc.org,eshr@google.com # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:11635 Bug: webrtc:11975 Change-Id: I61c8aed1270bc9c2f7f0577fa5ca717a325f548a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187484 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32369}

This reverts commit c5f7108. Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls. Sample failed run: https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995? Sample logs: STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575 STDERR: # last system error: 0 STDERR: # Check failed: (signaling_thread())->IsCurrent() STDERR: # Received signal 6 STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace() STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace() STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler() STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f) STDERR: #4 0x7f81c8d72db1 gsignal STDERR: #5 0x7f81c8d5c537 abort STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog() STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>() STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach() STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL() STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent() STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived() STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket() STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket() STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket() STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket() STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived() STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived() STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept() STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage() STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage() STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage() STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept() STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept() STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage() STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages() STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal() STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState() STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady() STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce() STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask() STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run() STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() STDERR: #40 0x7f81d395ae55 base::RunLoop::Run() STDERR: #41 0x7f81d39c87f2 base::Thread::Run() Original change's description: > Reland "Replace sigslot usages with robocaller library." > > This is a reland of 40261c3 > > Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError > added a new member with a different name and used it in webrtc code. > After this change do two more follow up CLs to completely remove the old code > from google3. > > Original change's description: > > Replace sigslot usages with robocaller library. > > > > - Replace all the top level signals from jsep_transport_controller. > > - There are still sigslot usages in this file so keep the inheritance > > and that is the reason for not having a binary size gain in this CL. > > > > Bug: webrtc:11943 > > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540 > > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#32321} > > Bug: webrtc:11943 > Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946 > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32359} TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,glahiru@webrtc.org Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11943 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32372}

As a first step we only want to enable frame pacing for the case where min playout delay == 0 and max playout delay > 0. (cherry picked from commit b2745ba) Bug: chromium:1237402, chromium:1239469 Change-Id: Icf9641db7566083d0279135efa8618e435d881eb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228640 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#34752} Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229021 Cr-Commit-Position: refs/branch-heads/4577@{#7} Cr-Branched-From: 5196931-refs/heads/master@{#34463}

Andrey Logvin and others added 30 commits December 2, 2020 18:42

Niels Möller and others added 18 commits January 14, 2021 11:45

mininny requested review from famous90, sf-nathan-park and x-0o0 February 24, 2021 09:43

sf-nathan-park approved these changes Feb 24, 2021

View reviewed changes

mininny added 2 commits April 15, 2021 16:48

Make RTCDefaultShader public

8f62743

Modify build configuration

b6a9851

x-0o0 approved these changes Apr 19, 2021

View reviewed changes

Merge pull request #8 from sendbird/feature/mininny/make-shader-public

4a1439e

Make shader public

mininny merged commit 4019d07 into develop Apr 21, 2021

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Update webrtc version to m89 #7

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Update webrtc version to m89 #7

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