09408115cd
This CL moves the responsibility for demuxing from FakeNetworkPipe to DirectTransport. This makes the interface for FakeNetworkPipe more consistent. It exposes fewer different interfaces for different usages. It also means that any time degradations applied to the packets due in FakeNetworkPipe in tests will now be propagated to Call in a more realistic manner. Previously the time was set to uninitialized which meant that Call filled in values based on the system clock. Bug: webrtc:9054 Change-Id: Ie534062f5ae9ad992c06b19e43804138a35702f0 Reviewed-on: https://webrtc-review.googlesource.com/64260 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23017}
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