e.g all files in the api/test folder not including subdirectories
Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42773}
* Simplified ctor. Get settings (max_qp, content_type, etc) from encoder_config passed to CreateEncoderStreams().
* Some tests assigned VideoEncoderConfig::video_stream_factory to EncoderStreamFactory they created. That's not really needed. VideoStreamEncoder creates the factory if video_stream_factory is not provided [1]. Removed video_stream_factory initialization in tests.
[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/video_stream_encoder.cc;l=1002;drc=1d7d0e6e2c5002815853be251ce43fe88779ac85
Bug: b/347150850, webrtc:42233936
Change-Id: Ie0322abb6c48e1a9bd10e9ed3879e3ed484fea5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355321
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42608}
This reverts commit 86ff48adaea08fd4e7044595e1c25a22fcceac34.
Reason for revert: Speculative revert due to failing downstream tests
Original change's description:
> Rewrite simulcast config to equivalent SVC for vp9 simulcast
>
> This allows to utilize libvpx optimizations considerably improving performance.
> The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
>
> This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
>
> Bug: webrtc:347737882
> Change-Id: Ic48316ad597700ed07e594d592413cf84b6b20d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355003
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42554}
Bug: webrtc:347737882
Change-Id: Ib84c9c0e20763348abfae838f2fb1aff31581a55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355943
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42564}
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
Bug: webrtc:347737882
Change-Id: Ic48316ad597700ed07e594d592413cf84b6b20d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355003
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42554}
OpenH264 cannot be usually used everywhere as it's proprietary and for
that reason it's usually disabled or apps using it are not allowed to be
available in default installations. Using system OpenH264 option allows
us to use e.g. noopenH264, that can be present in default installations
and later replaced by OpenH264 installed from 3rd party repository.
Bug: webrtc:14717
Change-Id: I015aacdb48c0636935f611459f0c9a6aa74a8f94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349301
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#42509}
So that this class can use propagated field trials instead of the global
Bug: webrtc:42220378
Change-Id: Ic1dba0c4967735606904329f7e9e6c09f186b809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42326}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
This relands the
https://webrtc-review.googlesource.com/c/src/+/349920
where patchset#1 is identical to the original change,
patchset#2 undoes (postpones) the expectation downstream propagates the Environment too.
Bug: webrtc:42220378
Change-Id: I4a9a32bb0926a875d37f3ba19dd5309e97546553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350364
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42298}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
Bug: webrtc:42220378
Change-Id: I52816628169a54b18a4405d84fee69b101f92f72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42288}
Environment provides non-null interface for FieldTrialsView and thus VideoCodingModule no longer need to rely on FieldTrialBasedConfig class to provide field_trials when not passed at construction.
Bug: webrtc:10335
Change-Id: Iedfb29e8b29056618a85f2e7a1528da29e3be5c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347701
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42163}
That forces external field trials, thus VideoCoding will be able to remove dependency on the global field trials string through FieldTrialBasedConfig class.
Bug: webrtc:10335
Change-Id: I6d22a7d20a4433801a0086b0863cda78e91f4f60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347646
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42092}
Instead of relying on the global field trial string
Bug: webrtc:10335
Change-Id: I491be089ffc725fd28483edf10eae4ae5d17d651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346263
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42021}
This CL updates H26xPacketBuffer to store and prepend SPS and PPS for
H.264 bitstreams when IDR only keyframe is allowed.
Bug: webrtc:13485
Change-Id: Ic1edc623dff568d54d3ce29b42dd8eab3312f5cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342225
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41986}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
This is a reland of commit a2655449ee310704ee2053fd6d43a5ab7002b755
This CL guards H265 header behind RTC_ENABLE_H265.
Original change's description:
> Add HEVC support for h264_packet_buffer.
>
> Renamed to h26x_packet_buffer as it also supports HEVC now. For HEVC,
> start code is added by depacktizer, and remote endpoint must send
> sequence and picture information in-band.
>
> Co-authored-by: Qiujiao Wu <qiujiao.wu@intel.com>
>
> Bug: webrtc:13485
> Change-Id: I321cb223357d8d15da95cec80ec0c3a43c26734e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333863
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41739}
Bug: webrtc:13485
Change-Id: I478e0ab88adcef34100670a90b12251ab3c9b623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339822
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41748}
Renamed to h26x_packet_buffer as it also supports HEVC now. For HEVC,
start code is added by depacktizer, and remote endpoint must send
sequence and picture information in-band.
Co-authored-by: Qiujiao Wu <qiujiao.wu@intel.com>
Bug: webrtc:13485
Change-Id: I321cb223357d8d15da95cec80ec0c3a43c26734e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333863
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41739}
This is a reland of commit 050ffefd854f8a57071992238723259e9ae0d85a
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
This reverts commit 050ffefd854f8a57071992238723259e9ae0d85a.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
To reduce number calls to the CreateVideoDecoder
Bug: webrtc:15791
Change-Id: I5d6ecc2e5e68165d4e012b3ad7edb6eaa40e1913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41706}
This is a reland of commit 496893e89e5bc8139e50befcb1a26eadbd829b0d
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Iccb9af8bf6a6c37704bc58b6e57238b55761b079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41194}
This reverts commit 496893e89e5bc8139e50befcb1a26eadbd829b0d.
Reason for revert: Breaks https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/16103/overview
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Ifdce738058c3e3fc7c76886939add2813405cae7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327722
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41183}
This enables testing different settings without updating code and rebuilding the test binary. Example of command:
video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
Bug: webrtc:14852
Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41179}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
Convert most field trials used in PCLF tests.
Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d
Bug: webrtc:10335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40909}
