[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
982dc791db
A recent refactoring introduced a subtle difference in how encoded frame timing is measured. See https://webrtc-review.googlesource.com/c/src/+/124122/14/video/video_stream_encoder.cc#b1278 After that change, the encode was considered done after encoding was done. The old behavior included the time needed to call stats and the video sink, which might include video quality tests related tasks and RTP packetization. In order to preserve the old behavior I'm moving timestamping to after packetization again. Note that the timing frame info still has a separate timestamp that does explicitly measure encode time. This is used by the experimental new overuse detector, so the effect of this change will be transient anyhow. Bug: chromium:941457, webrtc:10164 Change-Id: Ia990a1ceaeaf2c45d5df2a32d4f017cdb08e3c55 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127569 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27108} |
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| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| crypto | ||
| data | ||
| examples | ||
| logging | ||
| media | ||
| modules | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| style-guide | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .vpython | ||
| abseil-in-webrtc.md | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| common_types.h | ||
| DEPS | ||
| ENG_REVIEW_OWNERS | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| PATENTS | ||
| presubmit_test_mocks.py | ||
| presubmit_test.py | ||
| PRESUBMIT.py | ||
| pylintrc | ||
| README.chromium | ||
| README.md | ||
| style-guide.md | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| whitespace.txt | ||
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct