| commit | a518a39963d34616d8f0e94991c7f5fbb5affb38 | [log] [tgz] |
|---|---|---|
| author | stefan <stefan@webrtc.org> | Tue Feb 21 12:12:23 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Feb 21 12:12:23 2017 |
| tree | 05a6c48a3cad06b3886a5f57afe4de920679cbc3 | |
| parent | 872104ac41d7764f8676c9ea55555210bea4605c [diff] |
Fixes a bug where a video stream can get stuck in the suspended state. This happens if a lot of FEC is allocated when the stream becomes suspended. The required bitrate to unsuspend can then be too high so that the padding bitrate we are allowed to generate is not enough. This CL also switches the tests from using ISAC to OPUS as RampUpTest.UpDownUpAudioVideoTransportSequenceNumberRtx relies on audio BWE to work (which is only compatible with OPUS). I don't know why it didn't fail before. BUG=webrtc:7178 Review-Url: https://codereview.webrtc.org/2705603002 Cr-Commit-Position: refs/heads/master@{#16739}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.