This reverts commit 5fd6e5ec1feed9e937efbe27ba9924cee3be2b81. Reason for revert: breaks downstream project please still keep default arguments for CreateEventLog Original change's description: > Remove deprecated functions from RtcEventLog > > The unified Log() interface replaces the many old LogX() functions. This helps hide dependencies between the modules which log different events. > > TBR=stefan@webrtc.org > > Bug: webrtc:8111 > Change-Id: I5ea9fd50ba6da87d5867513c81c5e3bdb0524a32 > Reviewed-on: https://webrtc-review.googlesource.com/2689 > Commit-Queue: Elad Alon <eladalon@webrtc.org> > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Reviewed-by: Elad Alon <eladalon@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20159} TBR=danilchap@webrtc.org,eladalon@webrtc.org,terelius@webrtc.org,stefan@webrtc.org Change-Id: Iefc195f5804dabc0f76b87f889ff55481f4d285b No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8111 Reviewed-on: https://webrtc-review.googlesource.com/6842 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20164}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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.
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://webrtc.googlesource.com/src
- 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
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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