commit | fe90ad195f16e654e0b6281641b08d8821236308 | [log] [tgz] |
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author | hbos <hbos@webrtc.org> | Mon Feb 20 10:05:13 2017 |
committer | Commit bot <commit-bot@chromium.org> | Mon Feb 20 10:05:13 2017 |
tree | b40dc82906dc5b284f6a578a035919bfc90369d4 | |
parent | 6aeef74b6e04bfa78a90b38b5d2323b24f0875d2 [diff] |
TrackMediaInfoMap: Allow same SSRC for send and receive side. Running video loopback on https://appr.tc/ revealed that it is possible to use the same SSRC for a local and remote audio or video track. This caused a DCHECK crash. The constructor of TrackMediaInfoMap is updated to support this mapping and the unittest is updated (moved and modified a test from being a death test to being a non-death test). I've verified that this fixes the bug. BUG=chromium:693087 Review-Url: https://codereview.webrtc.org/2703783002 Cr-Commit-Position: refs/heads/master@{#16713}
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.