commit | 8c316c1a89b8ba3c31996d7dc7943fdd68e29c20 | [log] [tgz] |
---|---|---|
author | Zhi Huang <zhihuang@webrtc.org> | Mon Nov 13 21:13:45 2017 +0000 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Nov 13 21:13:55 2017 +0000 |
tree | 18778b84a46efe8447fbaa04b5e4487f17e43c29 | |
parent | 71677452f9cf210aa98162c6f4bd8d339e625337 [diff] |
Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." This reverts commit 71677452f9cf210aa98162c6f4bd8d339e625337. Reason for revert: Broke Chromium. Original change's description: > Replaced the SignalSelectedCandidatePairChanged with a new signal. > > |transport 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. > > Bug: webrtc:7013 > Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c > Reviewed-on: https://webrtc-review.googlesource.com/13520 > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20661} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,pthatcher@webrtc.org Change-Id: Ie0c76786855b65bb8caba7065593c961e4bf9de7 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7013 Reviewed-on: https://webrtc-review.googlesource.com/22764 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20662}
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.