RtpRtcp allows to register header extension by name

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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 
 #include <string.h>
 
 #include <set>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/common_types.h"
+#include "webrtc/config.h"
 #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 {
+namespace {
+RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
+  if (extension == RtpExtension::kTOffset)
+    return kRtpExtensionTransmissionTimeOffset;
+  if (extension == RtpExtension::kAudioLevel)
+    return kRtpExtensionAudioLevel;
+  if (extension == RtpExtension::kAbsSendTime)
+    return kRtpExtensionAbsoluteSendTime;
+  if (extension == RtpExtension::kVideoRotation)
+    return kRtpExtensionVideoRotation;
+  if (extension == RtpExtension::kTransportSequenceNumber)
+    return kRtpExtensionTransportSequenceNumber;
+  RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
+  return kRtpExtensionNone;
+}
+}  // namespace
 
 RtpRtcp::Configuration::Configuration()
     : audio(false),
       receiver_only(false),
       clock(nullptr),
       receive_statistics(NullObjectReceiveStatistics()),
       outgoing_transport(nullptr),
       intra_frame_callback(nullptr),
       bandwidth_callback(nullptr),
       transport_feedback_callback(nullptr),
@@ skipping 614 matching lines @@
                                     const std::vector<uint32_t>& ssrcs) {
   rtcp_sender_.SetREMBData(bitrate, ssrcs);
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
     const RTPExtensionType type,
     const uint8_t id) {
   return rtp_sender_.RegisterRtpHeaderExtension(type, id);
 }
 
+bool ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(const std::string& type,
+                                                   const uint8_t id) {
+  return rtp_sender_.RegisterRtpHeaderExtension(StringToRtpExtensionType(type),
+                                                id) == 0;
+}
+
 int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
     const RTPExtensionType type) {
   return rtp_sender_.DeregisterRtpHeaderExtension(type);
 }
 
 // (TMMBR) Temporary Max Media Bit Rate.
 bool ModuleRtpRtcpImpl::TMMBR() const {
   return rtcp_sender_.TMMBR();
 }
 
@@ skipping 305 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc