| commit | 2d81eb33f5fe313ba36277067d2c9b77d1b6b54e | [log] [tgz] |
|---|---|---|
| author | terelius <terelius@webrtc.org> | Tue Oct 25 07:04:37 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Oct 25 14:04:44 2016 +0000 |
| tree | d5184724ca54573902158226b720f45f1bce1774 | |
| parent | 1836fd6257a692959b3b49ba99ef587ad9995871 [diff] |
Fix BWE simulations so that it uses the delay based BWE. Rename kFullSendSideEstimator -> kSendSideEstimator and add new class SendSideBweSender (controlled by kSendSideEstimator) that actually uses the send side BWE. Move the mock to logging/rtc_event_log/mock. Allow congestion_controller, remote_bitrate_estimator and audio to depend on loggging/rtc_event_log BUG=webrtc:6526 NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2431093003 Cr-Commit-Position: refs/heads/master@{#14772}
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.