Add support for transport wide sequence numbers

webrtc/video/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 43 matching lines @@
   for (const auto& ext : config.rtp.extensions) {
     // One-byte-extension local identifiers are in the range 1-14 inclusive.
     DCHECK_GE(ext.id, 1);
     DCHECK_LE(ext.id, 14);
     if (ext.name == RtpExtension::kAudioLevel) {
       CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, ext.id));
     } else if (ext.name == RtpExtension::kAbsSendTime) {
       CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, ext.id));
+    } else if (ext.name == RtpExtension::kTransportSequenceNumber) {
+      CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionTransportSequenceNumber, ext.id));
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return webrtc::AudioReceiveStream::Stats();
 }
 
@@ skipping 23 matching lines @@
       header.extension.hasAbsoluteSendTime) {
     int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc