This change removes RTCP Feedback loop if we are using datagram transport by removing transport sequence numbers from RTP packets and recreating RTCP Feedback from Datagram ACKs and Timestamps. - For outgoing RTP packets, remove transport sequence number and store it with datagram_id. Note that removing transport sequence numbers does not only save 4-8 bytes per packet, but also prevents generation of feedback packets on the receiver side. - When datagram ACKs, we re-created RTCP feedback with timestamp. - Replacing previous assumption that datagram_id was the same as packet_id by storing incremental counter of datagram ids (I noticed some packets come without packet_id, which is a bit strange, but easy to support and it's also good not to rely on packet_ids being unique across multiple ssrcs). Bug: webrtc:9719 Change-Id: Iecfe938ecea1a74e7c9e1484f0e985d72643d4a1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145269 Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28542}
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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
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Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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