| commit | 979d6f96a81ed73fd02b601a07699c5e2a39e724 | [log] [tgz] |
|---|---|---|
| author | Danil Chapovalov <danilchap@webrtc.org> | Tue Nov 28 14:06:35 2017 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Nov 28 13:52:08 2017 +0000 |
| tree | d6f611f8933c017425b67a1762f05179829c7a3c | |
| parent | 4fa5da54d510814882f27217595d7b0f4343fd2b [diff] |
in RtcpTransceiver tests use dedicate RtcpParserTransport class to pass packet to RtcpPacketParser This helpers make tests setup cleaner and makes explicit expectation on number of packets passed to the transport. Bug: webrtc:8239 Change-Id: I2d5975be59327cee440e87dbd0701b93514c9726 Reviewed-on: https://webrtc-review.googlesource.com/22460 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20911}
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.
Authoritative list of directories that contain the native API header files.