There exist a bug in the video_coding::PacketBuffer which triggers when a frame is the same size as the buffer. A trivial workaround is to increase the start size to something big so that this never happens in practice. The bug has been fixed but we still want to test the workaround in ToT, which is why this CL exist. BUG=webrtc:8028, chromium:752886 R=stefan@webrtc.org Review-Url: https://codereview.webrtc.org/2994093002 . Cr-Commit-Position: refs/heads/master@{#19308}
Reland of move linux_internal from the autoroller CQ. (patchset #1 id:1 of https://codereview.webrtc.org/2990233002/ )
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
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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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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