| commit | 942bc2e4b9133e6346600e43fdf11c9b37c19bbf | [log] [tgz] |
|---|---|---|
| author | Zhi Huang <zhihuang@webrtc.org> | Mon Nov 13 21:26:07 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Nov 13 22:50:11 2017 |
| tree | 13656e028aa3fa3d81139fb14302c6812840ed72 | |
| parent | 620d9c919504d9d7af94366fa22730f139a991db [diff] |
Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. |packet_overhead| field is added to rtc::NetworkRoute structure. In PackTransportInternal: 1. network_route() is added which returns the current network route. 2. debug_name() is removed. 3. transport_name() is moved from DtlsTransportInternal and IceTransportInternal to PacketTransportInternal. When the selected candidate pair is changed, the P2PTransportChannel will fire the SignalNetworkRouteChanged instead of SignalSelectedCandidatePairChanged to upper layers. The Rtp/SrtpTransport takes the responsibility of calculating the transport overhead from the BaseChannel so that the BaseChannel doesn't need to depend on P2P layer transports. TBR=pthatcher@webrtc.org Bug: webrtc:7013 Change-Id: If9928b25a7259544c2d9c42048b53ab24292fc67 Reviewed-on: https://webrtc-review.googlesource.com/22767 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20664}
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.