bcdfc8975ea28f256f777d841e25f2e749090d96

Correspondingly, change the parser so that it provides the frames grouped by SSRC. Also fix a small bug that made the audio playout test terminate too early before verifying correct logging of all events. Bug: webrtc:8802 Change-Id: I363ef120cf88fe99290998cbc14ab5dbf32e9607 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181066 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31962}
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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 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
Description
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