Reason for revert: Breaking webrtc builder. Original issue's description: > Adding IceConfig option to assume TURN/TURN candidate pairs will work. > > This will allow media to be sent on these pairs before a binding > response is received, shortening call setup time. However, this is only > possible if the TURN servers don't require CreatePermission when > communicating with each other. > > R=honghaiz@webrtc.org, pthatcher@webrtc.org > > Committed: https://crrev.com/8e6134eae4117a239de67c9a9dae8f5e3235d803 > Cr-Commit-Position: refs/heads/master@{#13263} NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=pthatcher@webrtc.org,deadbeef@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. Review-Url: https://codereview.webrtc.org/2090823002 Cr-Commit-Position: refs/heads/master@{#13264}
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
Revert of Adding IceConfig option to assume TURN/TURN candidate pairs will work. (patchset #9 id:160001 of https://codereview.webrtc.org/2063823008/ )
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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