| commit | 9c47b00e24da2941eb095df5a4459c6d98a8a88d | [log] [tgz] |
|---|---|---|
| author | nisse <nisse@webrtc.org> | Tue Mar 28 11:59:41 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Mar 28 11:59:41 2017 |
| tree | a87c2afe2fef8f9a0303ca9640ee728f815bd83f | |
| parent | 0238ba83272cff3820b90890eb32f5be7ec6d1e7 [diff] |
Don't hardcode MediaType::ANY in FakeNetworkPipe. Instead let each test set the appropriate media type. This simplifies demuxing in Call and later in RtpTransportController. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2774463003 Cr-Commit-Position: refs/heads/master@{#17418}
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.