I'll be doing some changes to code it tests (rtp_receiver_audio, specifically) and want to make sure there are tests in place before I touch anything. Fixed test_api_audio not properly checking payload data. Required a fix to LoopBackTransport in test_api to as to act like the regular audio and video parts of WebRTC and separate payload from header data. Also added a test for CNG and cleaned up constants. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2378403004 Cr-Commit-Position: refs/heads/master@{#14529}
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.