Update includes for webrtc/{base => rtc_base} rename (2/3)

webrtc/video/call_stats.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/video/call_stats.h"
 
 #include <algorithm>
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace {
 // Time interval for updating the observers.
 const int64_t kUpdateIntervalMs = 1000;
 // Weight factor to apply to the average rtt.
 const float kWeightFactor = 0.3f;
 
 void RemoveOldReports(int64_t now, std::list<CallStats::RttTime>* reports) {
@@ skipping 152 matching lines @@
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - time_of_first_rtt_ms_) / 1000;
   if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
     int64_t avg_rtt_ms = (sum_avg_rtt_ms_ + num_avg_rtt_ / 2) / num_avg_rtt_;
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
   }
 }
 
 }  // namespace webrtc