9a0662ac7e
Since now RTP rate is set correctly for audio, there's no need to use the very last data packet rtp/capture timestamps for generating RTCP SR packets. Using only one (first) packet timestamp eliminates the jitter between rtp and capture timestamps for audio. This jitter comes from the fact that capture timestamp for audio is unknown and we generate bogus timestamp at arbitrary, non-constant offset from the real capture time. Bug: webrtc:9905 Change-Id: I855556184cfe994be39ab7780836a050f5a38c35 Reviewed-on: https://webrtc-review.googlesource.com/c/108580 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25430}
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
- Code of conduct