eb9c3f237b
* Adds a OnPacketSent callback to MediaChannel, which matches with MediaChannel::NetworkInterface::SendPacket. * Moves the OnPacketSent handling to the media channel implementations (video/voice) and removes the PeerConnection/SdpOfferAnswerHandler layer from the call path. * Call::OnSentPacket is called directly from the channels on the network thread. This eliminates a PostTask to the worker thread for every audio/video network packet. * Remove sigslot dependency from MediaChannel (and derived). Bug: webrtc:11993 Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33777}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs