Add support for transport wide sequence numbers

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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/video/video_send_stream.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video/video_capture_input.h"
 #include "webrtc/video_engine/vie_channel.h"
 #include "webrtc/video_engine/vie_channel_group.h"
 #include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
@@ skipping 99 matching lines @@
     int id = config_.rtp.extensions[i].id;
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     DCHECK_GE(id, 1);
     DCHECK_LE(id, 14);
     if (extension == RtpExtension::kTOffset) {
       CHECK_EQ(0, vie_channel_->SetSendTimestampOffsetStatus(true, id));
     } else if (extension == RtpExtension::kAbsSendTime) {
       CHECK_EQ(0, vie_channel_->SetSendAbsoluteSendTimeStatus(true, id));
     } else if (extension == RtpExtension::kVideoRotation) {
       CHECK_EQ(0, vie_channel_->SetSendVideoRotationStatus(true, id));
+    } else if (extension == RtpExtension::kTransportSequenceNumber) {
+      CHECK_EQ(0, vie_channel_->SetSendTransportSequenceNumber(true, id));
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 
   // TODO(pbos): Consider configuring REMB in Call.
   channel_group_->SetChannelRembStatus(true, false, vie_channel_);
 
   // Enable NACK, FEC or both.
   const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0;
@@ skipping 350 matching lines @@
                                        vie_channel_->IsSendingFecEnabled());
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc