| commit | fc968a283c91cd4395b0ee8f232d29d566cc1316 | [log] [tgz] |
|---|---|---|
| author | Peter Boström <pbos@webrtc.org> | Fri Feb 19 15:14:37 2016 |
| committer | Peter Boström <pbos@webrtc.org> | Fri Feb 19 15:14:44 2016 |
| tree | 63b7221e55ce87e9f5b5e59b9f0329302559cc95 | |
| parent | 88788adcfd9e84e3e8882806e68d759898be0907 [diff] |
Fix sequence-number replay race for padding. Prevents allocating sequence numbers for packets that go out on the network even though sending media is disabled. This race caused a replay of sequence numbers when GetRtpState() on a stopped stream would not return the last sequence number sent, since the pacer thread could request and send padding on a later sequence number before the modules are disconnected from the pacer. BUG=webrtc:5543 R=stefan@webrtc.org TEST=Repeating EndToEndTest.RestartingSendStreamPreservesRtpState 1000 times under TSan. Review URL: https://codereview.webrtc.org/1715703002 . Cr-Commit-Position: refs/heads/master@{#11685}
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.