Add support for transport wide sequence numbers

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 12 matching lines @@
 // Disable warning C4355: 'this' : used in base member initializer list.
 #pragma warning(disable : 4355)
 #endif
 
 namespace webrtc {
 
 RtpRtcp::Configuration::Configuration()
     : id(-1),
       audio(false),
       receiver_only(false),
-      clock(NULL),
+      clock(nullptr),
       receive_statistics(NullObjectReceiveStatistics()),
-      outgoing_transport(NULL),
-      intra_frame_callback(NULL),
-      bandwidth_callback(NULL),
-      rtt_stats(NULL),
-      rtcp_packet_type_counter_observer(NULL),
+      outgoing_transport(nullptr),
+      intra_frame_callback(nullptr),
+      bandwidth_callback(nullptr),
+      rtt_stats(nullptr),
+      rtcp_packet_type_counter_observer(nullptr),
       audio_messages(NullObjectRtpAudioFeedback()),
-      remote_bitrate_estimator(NULL),
-      paced_sender(NULL),
-      send_bitrate_observer(NULL),
-      send_frame_count_observer(NULL),
-      send_side_delay_observer(NULL) {
+      remote_bitrate_estimator(nullptr),
+      paced_sender(nullptr),
+      packet_router(nullptr),
+      send_bitrate_observer(nullptr),
+      send_frame_count_observer(nullptr),
+      send_side_delay_observer(nullptr) {
 }
 
 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
   if (configuration.clock) {
     return new ModuleRtpRtcpImpl(configuration);
   } else {
     // No clock implementation provided, use default clock.
     RtpRtcp::Configuration configuration_copy;
     memcpy(&configuration_copy, &configuration,
            sizeof(RtpRtcp::Configuration));
     configuration_copy.clock = Clock::GetRealTimeClock();
     return new ModuleRtpRtcpImpl(configuration_copy);
   }
 }
 
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
     : rtp_sender_(configuration.id,
                   configuration.audio,
                   configuration.clock,
                   configuration.outgoing_transport,
                   configuration.audio_messages,
                   configuration.paced_sender,
+                  configuration.packet_router,
+                  configuration.send_time_callback,
                   configuration.send_bitrate_observer,
                   configuration.send_frame_count_observer,
                   configuration.send_side_delay_observer),
       rtcp_sender_(configuration.id,
                    configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
                    configuration.rtcp_packet_type_counter_observer),
       rtcp_receiver_(configuration.id,
                      configuration.clock,
@@ skipping 916 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc