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Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too. It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet. BUG=webrtc:1600 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1639253007 . Cr-Commit-Position: refs/heads/master@{#13503}
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.