d40b0f39e0
These tests got flaky under the new jitter buffer. Enhancements: - Use send-side BWE. - Let BWE ramp up before applying packet loss. - Improve packet loss simulation for ULPFEC. - Add delay to fake network pipe for FlexFEC. (Not added for ULPFEC, since this makes those flaky...?) - Add FlexFEC+NACK test, using RTX instead of "raw retransmits". - Tighter checks of received packets' payload types and SSRCs. TESTED= $ ninja -C out/Debug video_engine_tests && third_party/gtest-parallel/gtest-parallel -r 1000 out/Debug/video_engine_tests --gtest_filter="*EndToEnd*Ulpfec*:*EndToEnd*Flexfec*" ninja: Entering directory `out/Debug' ninja: no work to do. [12000/12000] TestWithNewVideoJitterBuffer/EndToEndTest.RecoversWithFlexfecAndNack/1 (14935 ms) BUG=webrtc:7047 Review-Url: https://codereview.webrtc.org/2675573004 Cr-Commit-Position: refs/heads/master@{#16449}
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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