Change default timestamp to 64 bits in all webrtc directories.

webrtc/video/send_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.
  */
 
@@ skipping 79 matching lines @@
 }
 
 SendStatisticsProxy::UmaSamplesContainer::UmaSamplesContainer(
     const char* prefix,
     const VideoSendStream::Stats& stats,
     Clock* const clock)
     : uma_prefix_(prefix),
       clock_(clock),
       max_sent_width_per_timestamp_(0),
       max_sent_height_per_timestamp_(0),
-      input_frame_rate_tracker_(100u, 10u),
-      sent_frame_rate_tracker_(100u, 10u),
+      input_frame_rate_tracker_(100, 10u),
+      sent_frame_rate_tracker_(100, 10u),
       first_rtcp_stats_time_ms_(-1),
       first_rtp_stats_time_ms_(-1),
       start_stats_(stats) {}
 
 SendStatisticsProxy::UmaSamplesContainer::~UmaSamplesContainer() {}
 
 void AccumulateRtpStats(const VideoSendStream::Stats& stats,
                         const VideoSendStream::Config& config,
                         StreamDataCounters* total_rtp_stats,
                         StreamDataCounters* rtx_stats) {
@@ skipping 512 matching lines @@
   return Fraction(min_required_samples, 1000.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Fraction(
     int min_required_samples, float multiplier) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return static_cast<int>((sum * multiplier / num_samples) + 0.5f);
 }
 }  // namespace webrtc