Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

webrtc/media/base/rtputils.h

Index: webrtc/media/base/rtputils.h
diff --git a/webrtc/media/base/rtputils.h b/webrtc/media/base/rtputils.h
index 1a710576117e72debbdfb925ec3a225d8afcd9ed..f7aa85c828da0e8d8f772a04c34c9a16e54d7a42 100644
--- a/webrtc/media/base/rtputils.h
+++ b/webrtc/media/base/rtputils.h
@@ -58,6 +58,14 @@ bool IsRtpPacket(const void* data, size_t len);
 // True if |payload type| is 0-127.
 bool IsValidRtpPayloadType(int payload_type);
 
+// True if |size| is appropriate for the indicated packet type.
+bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
+
+// TODO(zstein): Consider using an enum instead of a bool to differentiate
+// between RTP and RTCP.
+// Returns "RTCP" or "RTP" according to |rtcp|.
+const char* RtpRtcpStringLiteral(bool rtcp);
+
 // Verifies that a packet has a valid RTP header.
 bool ValidateRtpHeader(const uint8_t* rtp,
                        size_t length,