Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) Reason for revert: Broke video sending for iOS AppRTCDemo. To repro, run iOS AppRTCDemo in Release in loopback mode. The revision prior to this change worked. Original issue's description: > Reland of: 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://chromium.googlesource.com/external/webrtc/+/2d5491783a264c6e57614f87f1a69dc61bf44609 TBR=pthatcher@webrtc.org,deadbeef@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/2092273003 Cr-Commit-Position: refs/heads/master@{#13289}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.