36500ab634
Keep the logic managing whether audio RTP timestamps have the random start offset added or not inside ChannelSend, so that the ChannelSendFrameTransformerDelegate doesn't need to worry about it. Crucially, this means that frames moved between senders by encoded transforms clients will always use the correct offset for the channel where we actually get sent. Also rename TS variables throughout both classes to be explicit over whether the offset has been added or not. Bug: chromium:1464847 Change-Id: I19955ec4c1cb834161b00dd74622725a070b713a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317900 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40655}
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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation