commit | bb095aa99bb42ccd3d42ff392ca2c04154982a51 | [log] [tgz] |
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author | Niels Möller <nisse@webrtc.org> | Thu Aug 30 13:46:50 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Sep 03 07:28:39 2018 |
tree | e8958e29f3d91dabb687afc110d0fd93a3c0f92f | |
parent | 689b5874d4fb5ee1767aef0d05e0a929ac2ea46e [diff] |
Allow send bitrate < start bitrate in RampUpTest. 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.
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.