Breaking out RTP header parsing from the RTP module.

This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/video_coding/main/test/test_util.h

@@ -50,7 +50,7 @@ int MTRxTxTest(CmdArgs& args);
 double NormalDist(double mean, double stdDev);
 
 struct RtpPacket {
-  int8_t data[1650]; // max packet size
+  uint8_t data[1650]; // max packet size
   int32_t length;
   int64_t receiveTime;
 };