| commit | 075c6d7f7ed6586f7ccdf5c3eed77b0b0afdd434 | [log] [tgz] |
|---|---|---|
| author | brandtr <brandtr@webrtc.org> | Mon Jan 09 13:11:09 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Jan 09 13:11:09 2017 |
| tree | a969858c79802e1731a31bbd3ce14ca9f5fe110e | |
| parent | 29fe6f338fdd632708073e03dba21f41599a4a2e [diff] |
Temporarily remove SSRC DCHECK in RTPSender::SendToNetwork. Removing the DCHECK due to (sometimes) failing voe_auto_test. Long-term, this DCHECK should be readded. Before that can happen, the SSRC in the RTPSender should be made immutable. TESTED=No failures when running third_party/gtest-parallel/gtest-parallel --repeat=5000 --gtest_filter="VolumeTest.ManualInputMutingMutesMicrophone" out/Debug/voe_auto_test. BUG=webrtc:6887 Review-Url: https://codereview.webrtc.org/2610873002 Cr-Commit-Position: refs/heads/master@{#15962}
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.