WebRTC 1.0 has added the transceiver API to PeerConnection. This
is the first step towards exposing this to WebRTC consumers. For
now, transceivers can be added and fetched but there is not yet
support for creating offers/answers or setting local/remote
descriptions. That support ("Unified Plan") will be added in
follow-up CLs.
The transceiver API is currently only available if the application
opts in by specifying the kUnifiedPlan SDP semantics when creating
the PeerConnection.
Bug: webrtc:7600
Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
Reviewed-on: https://webrtc-review.googlesource.com/23880
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20896}
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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