bwe_rtp_play feeds packets from an RTP file into the BWE and prints the estimates. bwe_rtp_to_text parses an RTP file and outputs the result to a text file. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7689006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5466 4adac7df-926f-26a2-2b94-8c16560cd09d
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.