A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl. The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces. The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions. The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high. The new mode is default disabled and can be enabled using a field trial. Bug: webrtc:13322,webrtc:13966 Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36899}
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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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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