commit | 5565981e1722c9aeeab8bd7440eac121417b64b0 | [log] [tgz] |
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author | Åsa Persson <asapersson@webrtc.org> | Mon Jun 18 17:51:32 2018 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jun 20 07:26:09 2018 +0000 |
tree | 26100e5850cce1e72f63efd0add093a0ff0ccb9b | |
parent | f88a22cf1150c0aa0c27b9963af250ff928f485a [diff] |
Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters. Target bitrate is set to 0.75 of the max bitrate. Bug: webrtc:9341, webrtc:8655 Change-Id: I9a8c8bb95bb1532d45f05578832418464452340e Reviewed-on: https://webrtc-review.googlesource.com/79821 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23676}
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.