Preserve RTP states for restarted VideoSendStreams.

A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.

Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -73,6 +73,10 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Set SequenceNumber, default is a random number.
   virtual int32_t SetSequenceNumber(const uint16_t seq) OVERRIDE;
 
+  virtual void SetRtpStateForSsrc(uint32_t ssrc,
+                                  const RtpState& rtp_state) OVERRIDE;
+  virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) OVERRIDE;
+
   virtual uint32_t SSRC() const OVERRIDE;
 
   // Configure SSRC, default is a random number.