Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

webrtc/media/base/rtputils.cc

 /*
  *  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 261 matching lines @@
   if (len < kMinRtpPacketLen)
     return false;
 
   return (static_cast<const uint8_t*>(data)[0] >> 6) == kRtpVersion;
 }
 
 bool IsValidRtpPayloadType(int payload_type) {
   return payload_type >= 0 && payload_type <= 127;
 }
 
+bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size) {
+  return (rtcp ? size >= kMinRtcpPacketLen : size >= kMinRtpPacketLen) &&
+         size <= kMaxRtpPacketLen;
+}
+
+const char* RtpRtcpStringLiteral(bool rtcp) {
+  return rtcp ? "RTCP" : "RTP";
+}
+
 bool ValidateRtpHeader(const uint8_t* rtp,
                        size_t length,
                        size_t* header_length) {
   if (header_length) {
     *header_length = 0;
   }
 
   if (length < kMinRtpPacketLen) {
     return false;
   }
@@ skipping 163 matching lines @@
     UpdateRtpAbsSendTimeExtension(start, rtp_length,
                                   packet_time_params.rtp_sendtime_extension_id,
                                   time_us);
   }
 
   UpdateRtpAuthTag(start, rtp_length, packet_time_params);
   return true;
 }
 
 }  // namespace cricket