a34d7766c5
Update RtpSenderAudioTest to call methods on RTPSenderAudio rather than RTPSender, when possible. In particular, avoid RTPSender::SendOutgoingData. Drop parameterization on the WebRTC-SendSideBwe-WithOverhead field trial, since that appears unrelated to these tests. Also delete some unused parts of the RtpSender test. Bug: webrtc:7135 Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0 Reviewed-on: https://webrtc-review.googlesource.com/c/120920 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26516}
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