Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

webrtc/media/base/rtputils.h

 /*
  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 40 matching lines @@
 
 bool SetRtpSsrc(void* data, size_t len, uint32_t value);
 // Assumes version 2, no padding, no extensions, no csrcs.
 bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
 
 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,
                        size_t* header_length);
 
 // Helper method which updates the absolute send time extension if present.
 bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
                                    size_t length,
                                    int extension_id,
                                    uint64_t time_us);
 
 // Applies specified |options| to the packet. It updates the absolute send time
 // extension header if it is present present then updates HMAC.
 bool ApplyPacketOptions(uint8_t* data,
                         size_t length,
                         const rtc::PacketTimeUpdateParams& packet_time_params,
                         uint64_t time_us);
 
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_RTPUTILS_H_