Danil Chapovalov 70ffead256 Reimplemented fix for bogus RTP timestamp in RTCP packet created before RTP packet.
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}
2016-07-20 13:27:09 +00:00
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