| commit | 493a650b1ef04f64f398bd16b78df3f9be0af8bd | [log] [tgz] |
|---|---|---|
| author | Ruslan Burakov <kuddai@google.com> | Wed Feb 27 15:32:48 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 27 15:08:34 2019 +0000 |
| tree | 5ebc10665af7715bfab897d8ff75bb5548e28eb3 | |
| parent | 48e7065ac6a34badd04d8f9d5dab64305e5b7b31 [diff] |
Propagate base minimum delay from video jitter buffer to webrtc/api. On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface, GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes the same concept. Bug: webrtc:10287 Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957 Reviewed-on: https://webrtc-review.googlesource.com/c/123482 Commit-Queue: Ruslan Burakov <kuddai@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26877}
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.
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.