Save the frame transformer set on unsignaled receivers, and set the transformer when the ssrc becomes known. Pass the receiver's ssrc on registering the transformed frame callback, to associate separate frame transformer sinks for each receiver. Bug: chromium:1065838 Bug: chromium:1065838 Change-Id: I2a214bdb6cb9a8012928a03f046f311c344370f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173201 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31051}
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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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