ba09f79ba31c543d0dcf3a6d824992685087005a
The warnings were (all signed integer overflow): webrtc/common_audio/signal_processing/levinson_durbin.c:46:25 12 * 268435456 cannot be represented in type 'int' webrtc/modules/audio_processing/aecm/aecm_core.cc:930:69 522240 * 6115 cannot be represented in type 'int' webrtc/modules/audio_processing/aecm/aecm_core_c.cc:455:36 72293096 * 50 cannot be represented in type 'int' webrtc/modules/pacing/alr_detector.cc:70:48 1000000000 * 65 cannot be represented in type 'int' webrtc/modules/rtp_rtcp/source/rtp_sender.cc:947:20 1929277286 + 321546521 cannot be represented in type 'int' BUG=webrtc:8195 Review-Url: https://codereview.webrtc.org/3005003002 Cr-Commit-Position: refs/heads/master@{#19670}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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://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
- Coding style guide
- Code of conduct
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