36d4c54500
Reason for revert: Breaks Chrome-FYI bots because of a change in the StartDebugRecording function in audio_processing.h, that is called from Chrome. Original issue's description: > Added option to specify a maximum file size when recording an AEC dump. > > For applications with a strict filesize limit for debug files, > I added an option to specify a maximum filesize for AEC dumps. An > existing unit test is extended to check that the feature works as > advertised. > > BUG=webrtc:4741 > TBR=glaznev@webrtc.org > > Committed: https://crrev.com/ae2c5ad12afc8cc29fe9c59dea432b697b871a87 > Cr-Commit-Position: refs/heads/master@{#11081} TBR=pthatcher@webrtc.org,henrik.lundin@webrtc.org,henrikg@webrtc.org,solenberg@webrtc.org,andrew@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org,glaznev@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1533913004 Cr-Commit-Position: refs/heads/master@{#11087}
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