ab9535c098
Update RtpPacketizerH264::PacketizeStapA to use single_packet_reduction_len when all fragments end up in one H.264 packet. Previous code was using first_packet_reduction_len + last_packet_reduction_len for this case, which can cause an occasional RTC_CHECK crash in RtpPacketizerH264::NextAggregatePacket due to exceeding the available payload capacity of an RTP packet. Bug: webrtc:15477 Change-Id: Iba1564a6a29618bef22f19d82aba938420994b23 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319645 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40737}
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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation