The data structures in RtpPacketHistory were chosen based on assumption of few packets with possible sparse segments due to missing acking. In practice high bitrate usages with full histories seem to be more of a problem. Due to that, change storage from an std::map to an std::deque and live with potential segments of nullptr. Also limit size of padding prio set so that doesn't become a bottleneck. Bug: webrtc:8975 Change-Id: I3b6314fb3255937d25362ff2cd906efb7b1397f7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145901 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29117}
Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video.
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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.
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- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
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Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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