commit | ba4c0e45ffcb9c1871389b07b9be5b515d7841d5 | [log] [tgz] |
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author | stefan <stefan@webrtc.org> | Thu Feb 04 12:12:24 2016 |
committer | Commit bot <commit-bot@chromium.org> | Thu Feb 04 12:12:31 2016 |
tree | d3a562f95d5431145775a58771c2f14d3ce5b04e | |
parent | 2ddb8bd3593a2a2797fea16f358cc89c140f6154 [diff] |
Add send-side BWE to WebRtcVoiceEngine under a finch experiment. This adds negotiation of both transport sequence number and transport feedback. Only offers transport seq num if the WebRTC-Audio-SendSideBwe finch experiment is enabled. TBR=mflodman@webrtc.org BUG=webrtc:5263 Review URL: https://codereview.webrtc.org/1604563002 Cr-Commit-Position: refs/heads/master@{#11487}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.