Add support for transport wide sequence numbers

webrtc/video_engine/vie_receiver.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 40 matching lines @@
       rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       fec_receiver_(FecReceiver::Create(this)),
       rtp_rtcp_(NULL),
       vcm_(module_vcm),
       remote_bitrate_estimator_(remote_bitrate_estimator),
       ntp_estimator_(new RemoteNtpTimeEstimator(clock_)),
       receiving_(false),
       restored_packet_in_use_(false),
       receiving_ast_enabled_(false),
       receiving_cvo_enabled_(false),
+      receiving_tsn_enabled_(false),
       last_packet_log_ms_(-1) {
   assert(remote_bitrate_estimator);
 }
 
 ViEReceiver::~ViEReceiver() {
   UpdateHistograms();
 }
 
 void ViEReceiver::UpdateHistograms() {
   FecPacketCounter counter = fec_receiver_->GetPacketCounter();
@@ skipping 121 matching lines @@
     } else {
       return false;
     }
   } else {
     receiving_cvo_enabled_ = false;
     return rtp_header_parser_->DeregisterRtpHeaderExtension(
         kRtpExtensionVideoRotation);
   }
 }
 
+bool ViEReceiver::SetReceiveTransportSequenceNumber(bool enable, int id) {
+  if (enable) {
+    if (rtp_header_parser_->RegisterRtpHeaderExtension(
+            kRtpExtensionTransportSequenceNumber, id)) {
+      receiving_tsn_enabled_ = true;
+      return true;
+    } else {
+      return false;
+    }
+  } else {
+    receiving_tsn_enabled_ = false;
+    return rtp_header_parser_->DeregisterRtpHeaderExtension(
+        kRtpExtensionTransportSequenceNumber);
+  }
+}
+
 int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
                                    size_t rtp_packet_length,
                                    const PacketTime& packet_time) {
   return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet),
                          rtp_packet_length, packet_time);
 }
 
 int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet,
                                     size_t rtcp_packet_length) {
   return InsertRTCPPacket(static_cast<const uint8_t*>(rtcp_packet),
@@ skipping 243 matching lines @@
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc