This commit prepares WebRtcSession so that it can be cleanly copy & pasted into PeerConnection in the next commit. To accomplish this, the following was done: 1. Added a pointer to the owning PeerConnection to WebRtcSession. 2. Replace WebRtcSession state enum with signaling state. 3. All signals/observers only observed by PeerConnection were replaced with direct calls to PeerConnection methods. 4. All duplicated fields were moved to PeerConnection. 5. The remaining tests that still use WebRtcSession for mocks were updated to minimize dependence on WebRtcSession construction. Bug: webrtc:8323 Change-Id: Ifc1a4ee819dcc9beca5363291012f7d5563ff7b1 Reviewed-on: https://webrtc-review.googlesource.com/9020 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20573}
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.
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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