ea7ef2ad1d
Most of the implementation in rtp_sender.cc is a copy paste for both Audio & Video RTP senders. This change moves all the common behavior into the base RtpSenderInternal class. Template method pattern is used to accomodate for the very slight differences between audio and video senders. Bug: None Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92 Reviewed-on: https://webrtc-review.googlesource.com/c/123411 Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Amit Hilbuch <amithi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26758}
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