5def7b9fde
Reason for revert: Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection. Original issue's description: > Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) > > Reason for revert: > Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream. > > Original issue's description: > > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) > > > > Reason for revert: > > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail. > > > > Original issue's description: > > > Adding the ability to create an RtpSender without a track. > > > > > > This CL also changes AddStream to immediately create a sender, rather > > > than waiting until the track is seen in SDP. And the PeerConnection now > > > builds the list of "send streams" from the list of senders, rather than > > > the collection of local media streams. > > > > > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5 > > > Cr-Commit-Position: refs/heads/master@{#10414} > > > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > > NOPRESUBMIT=true > > NOTREECHECKS=true > > NOTRY=true > > > > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb > > Cr-Commit-Position: refs/heads/master@{#10417} > > TBR=pthatcher@webrtc.org,pthatcher@chromium.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae > Cr-Commit-Position: refs/heads/master@{#10730} TBR=pthatcher@webrtc.org,pthatcher@chromium.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1460323002 Cr-Commit-Position: refs/heads/master@{#10732}
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