Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps."

This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtcp_sender.h

@@ -72,6 +72,8 @@ public:
 
     int32_t SetRemoteSSRC( const uint32_t ssrc);
 
+    int32_t SetCameraDelay(const int32_t delayMS);
+
     int32_t CNAME(char cName[RTCP_CNAME_SIZE]);
     int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]);
 
@@ -195,9 +197,6 @@ private:
                       const uint16_t* nackList,
                           std::string* nackString);
 
-    bool RtpTimestampNow(uint32_t ntp_secs_now, uint32_t ntp_fracs_now,
-                         uint32_t* timestamp_now) const;
-
 private:
     int32_t            _id;
     const bool               _audio;
@@ -230,6 +229,8 @@ private:
     std::map<uint32_t, RTCPReportBlock*> _reportBlocks;
     std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs;
 
+    int32_t         _cameraDelayMS;
+
     // Sent
     uint32_t        _lastSendReport[RTCP_NUMBER_OF_SR];  // allow packet loss and RTT above 1 sec
     uint32_t        _lastRTCPTime[RTCP_NUMBER_OF_SR];