Remove system_wrappers/interface/trace_event.h

webrtc/modules/video_coding/main/source/jitter_buffer.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.
  */
 #include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
 
 #include <assert.h>
 
 #include <algorithm>
 #include <utility>
 
+#include "webrtc/base/trace_event.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
 #include "webrtc/modules/video_coding/main/source/frame_buffer.h"
 #include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
 #include "webrtc/modules/video_coding/main/source/internal_defines.h"
 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
 #include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
 #include "webrtc/modules/video_coding/main/source/packet.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
-#include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 
 // Use this rtt if no value has been reported.
 static const int64_t kDefaultRtt = 200;
 
 typedef std::pair<uint32_t, VCMFrameBuffer*> FrameListPair;
 
 bool IsKeyFrame(FrameListPair pair) {
   return pair.second->FrameType() == kVideoFrameKey;
@@ skipping 1213 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc