f288f5b2d4
RtpVideoStreamReceiver was forked to RtpVideoStreamReceiver2 recently, so the code that checks for this parameter needs to be present in the forked location, but it wasn't. This also enables RtpVideoStreamReceiver2TestH264.InBandSpsPps test on MSAN, which was another already fixed bug that wasn't ported over to the recently forked RtpVideoStreamReceiver2. See webrtc:11595 for information about the fork. See webrtc:11769 for information about this fmtp parameter. See webrtc:11376 for the original MSAN issue. Bug: webrtc:11957, webrtc:11595, webrtc:11769, webrtc:8423 Change-Id: I3734d077b2883c2f747ad35a0189b83c1915c3ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184524 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32144}
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