Change default timestamp to 64 bits in all webrtc directories.

webrtc/video/receive_statistics_proxy.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/receive_statistics_proxy.h"
 
 #include <cmath>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/metrics.h"
 
 namespace webrtc {
 
 ReceiveStatisticsProxy::ReceiveStatisticsProxy(
     const VideoReceiveStream::Config& config,
     Clock* clock)
     : clock_(clock),
       config_(config),
       // 1000ms window, scale 1000 for ms to s.
       decode_fps_estimator_(1000, 1000),
       renders_fps_estimator_(1000, 1000),
-      render_fps_tracker_(100u, 10u),
-      render_pixel_tracker_(100u, 10u) {
+      render_fps_tracker_(100, 10u),
+      render_pixel_tracker_(100, 10u) {
   stats_.ssrc = config.rtp.remote_ssrc;
   for (auto it : config.rtp.rtx)
     rtx_stats_[it.second.ssrc] = StreamDataCounters();
 }
 
 ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
   UpdateHistograms();
 }
 
 void ReceiveStatisticsProxy::UpdateHistograms() {
@@ skipping 238 matching lines @@
   ++num_samples;
 }
 
 int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return sum / num_samples;
 }
 
 }  // namespace webrtc