9ab520e24b
This is a reland of b0ac94307e1787f83de2b9a2dc3b58309ea8654b Original change's description: > Avoid encrypting empty audio packet. > > Bug: b/132861665 > Change-Id: I161ba8697ae88857927f27fa6d3914b7201fdeab > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137049 > Commit-Queue: Minyue Li <minyue@webrtc.org> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28006} Bug: b/132861665 Change-Id: Ia9be25116c7d10fee847ee25c484e6422be24b31 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138218 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28086}
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