| commit | b686396ec6b6d9e90acc9f6c65eb2cf6a52fe1b8 | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Wed Oct 10 10:23:13 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Oct 10 09:47:46 2018 +0000 |
| tree | b5d30503704f260a0597f5e1ec8cd54a11985a51 | |
| parent | 99a70a2d785c0353c3e0eddb463cf75941b6ffd7 [diff] |
Makes AudioSendStream signal that it's part of allocation. This adds calls to the underlying RtpRtcp module to indicate when audio is part of bitrate allocation. This information is propagated and set in the packet info for each packet. This is part of a series of CLs that allows GoogCC to track sent bitrate that is included in bitrate allocation but without transport feedback. Bug: webrtc:9796 Change-Id: I79b024cb7f2eb8c86421cfa34d38ef68467776c3 Reviewed-on: https://webrtc-review.googlesource.com/c/104882 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25086}
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.