67ac9e8ecb
The post-pacing part of the RTP sender has been moved from RTPSender into the new RtpSenderEgress class. However, that class is not directly used and instead a subset of method calls are passed through RTPSender. This CL prepares for removing dependencies between RTPSender and RtpSenderEgress. All current behavior is preserved, and unit tests are unchanged to verify this. For more context, see patch set 2. Change-Id: If795f2603aeb6302ac1565d9efaea514af240dc7 Bug: webrtc:11036 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158020 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29616}
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