189f9b1b65
Reason for revert: Breaks downstream project Original issue's description: > Clean up logging in AudioSendStream::SetupSendCodec(). > > As a side effect: > - Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc. > - Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up. > - Which further exposed clang warnings about large inlined default methods in webrtc/config.h. > > BUG=webrtc:4690 > > Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871 > Cr-Commit-Position: refs/heads/master@{#14771} TBR=kwiberg@webrtc.org,solenberg@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2452643002 Cr-Commit-Position: refs/heads/master@{#14774}
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