1a7162dbc9
Reason for revert: Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland. Original issue's description: > Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. > > This eliminates the need for the extra layer of indirection provided by > mediastreamprovider.h. It will thus make it easier to implement new > functionality in RtpSender/RtpReceiver. > > It also brings us one step closer to the end goal of combining "senders" > and "send streams". Currently the sender still needs to go through the > BaseChannel and MediaChannel, using an SSRC as a key. > > R=pthatcher@webrtc.org > > Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c > Cr-Commit-Position: refs/heads/master@{#13285} TBR=pthatcher@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2099843003 Cr-Commit-Position: refs/heads/master@{#13286}
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc