Remove use of RtpHeaderExtension and clean up

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  Copyright (c) 2012 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 11 matching lines @@
 #include "webrtc/system_wrappers/include/trace.h"
 
 #ifdef _WIN32
 // Disable warning C4355: 'this' : used in base member initializer list.
 #pragma warning(disable : 4355)
 #endif
 
 namespace webrtc {
 
 RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
-  if (extension == RtpExtension::kTOffset)
+  if (extension == RtpExtension::kTimestampOffsetUri)
     return kRtpExtensionTransmissionTimeOffset;
-  if (extension == RtpExtension::kAudioLevel)
+  if (extension == RtpExtension::kAudioLevelUri)
     return kRtpExtensionAudioLevel;
-  if (extension == RtpExtension::kAbsSendTime)
+  if (extension == RtpExtension::kAbsSendTimeUri)
     return kRtpExtensionAbsoluteSendTime;
-  if (extension == RtpExtension::kVideoRotation)
+  if (extension == RtpExtension::kVideoRotationUri)
     return kRtpExtensionVideoRotation;
-  if (extension == RtpExtension::kTransportSequenceNumber)
+  if (extension == RtpExtension::kTransportSequenceNumberUri)
     return kRtpExtensionTransportSequenceNumber;
   RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
   return kRtpExtensionNone;
 }
 
 RtpRtcp::Configuration::Configuration()
     : audio(false),
       receiver_only(false),
       clock(nullptr),
       receive_statistics(NullObjectReceiveStatistics()),
@@ skipping 936 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc