| commit | b03b6c8a945859d489b560983821eb0d0b9ee843 | [log] [tgz] |
|---|---|---|
| author | Per Kjellander <perkj@webrtc.org> | Sun Jan 03 09:26:03 2021 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Jan 07 09:29:05 2021 |
| tree | 67accfdf7c55c193a0850e6545cbf27b73cb11fe | |
| parent | c96601e20cfd3d02e614a4e7f278e4732e31f273 [diff] |
Move setting of encoder bitrate allocation callback type to VideoSendStream It turned out that the negotiated rtp header extensions are not fully known in WebRtcVideoChannel::AddSendStream. The cl also remove the unnecessary factory for creating VideoStreamEncoder. Bug: webrtc:12000 Change-Id: If994c8deb69f3ce4212896d3ad757dac94c6e09f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198840 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32916}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.