This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4. Reason for revert: Breaks downstream project Original change's description: > Propagate media transport to media channel. > > 1. Pass media transport factory to JSEP transport controller. > 2. Pass media transport to voice media channel. > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel. > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71 > Bug: webrtc:9719 > Reviewed-on: https://webrtc-review.googlesource.com/c/105542 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Peter Slatala <psla@webrtc.org> > Commit-Queue: Anton Sukhanov <sukhanov@google.com> > Cr-Commit-Position: refs/heads/master@{#25152} TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:9719 Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1 Reviewed-on: https://webrtc-review.googlesource.com/c/105840 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25154}
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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