This is a reland of 49470c2ac460ed8cce250942e8525c5f14e32778 Tentative reland to rule-out bot flakiness. Original change's description: > Add AV1 RtpDepacketizer class > > Implement Parse function that extracts is_first_packet_in_frame, > is_last_packet_in_frame, and frame_type fields. > > Bug: webrtc:11042 > Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29814} TBR=saza@webrtc.org,philipel@webrtc.org Bug: webrtc:11042 Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29819}
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.
Authoritative list of directories that contain the native API header files.
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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