Make rtc::TimestampWrapAroundHandler handle backwards wrapping

webrtc/video/video_quality_test.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.
  */
 #include <stdio.h>
@@ skipping 143 matching lines @@
                const PacketOptions& options) override {
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
 
     int64_t current_time =
         Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     bool result = transport_->SendRtp(packet, length, options);
     {
       rtc::CritScope lock(&crit_);
-      int64_t timestamp = wrap_handler_.Unwrap(header.timestamp);
 
       if (rtp_timestamp_delta_ == 0) {
-        rtp_timestamp_delta_ = timestamp - first_send_frame_.timestamp();
+        rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
         first_send_frame_.Reset();
       }
-      timestamp -= rtp_timestamp_delta_;
+      int64_t timestamp =
+          wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
       send_times_[timestamp] = current_time;
       if (!transport_->DiscardedLastPacket() &&
           header.ssrc == ssrc_to_analyze_) {
         encoded_frame_sizes_[timestamp] +=
             length - (header.headerLength + header.paddingLength);
       }
     }
     return result;
   }
 
@@ skipping 914 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc