bb095aa99b
Primarily, this is intended to reduce flakyness of RampUpTest.AudioTransportSequenceNumber. We shouldn't expect audio send rate >= 300 kbps at all time in these tests. And in general, if it's at all relevant to test that bitrate doesn't drop below the start bitrate, a perf test isn't the right place for that. A run of ./third_party/gtest-parallel/gtest-parallel -r 1000 -w 1000 \ --gtest_filter=RampUpTest.AudioTransportSequenceNumber \ out/Release/webrtc_perf_tests passes when I ran it locally after this change, but fails around 4 out of 1000 times before the change. Bug: webrtc:8878 Change-Id: I08614ce5683c9ba6fe4b72bfde83e6a81445a59b Reviewed-on: https://webrtc-review.googlesource.com/96900 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24523}
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