Introduce dchecked_cast, and start using it

webrtc/modules/audio_coding/neteq/statistics_calculator.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 200 matching lines @@
     const DecisionLogic& decision_logic,
     NetEqNetworkStatistics *stats) {
   if (fs_hz <= 0 || !stats) {
     assert(false);
     return;
   }
 
   stats->added_zero_samples = added_zero_samples_;
   stats->current_buffer_size_ms =
       static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
-  const int ms_per_packet = rtc::checked_cast<int>(
+  const int ms_per_packet = rtc::dchecked_cast<int>(
       decision_logic.packet_length_samples() / (fs_hz / 1000));
   stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) *
       ms_per_packet;
   stats->jitter_peaks_found = delay_manager.PeakFound();
   stats->clockdrift_ppm =
       rtc::saturated_cast<int32_t>(delay_manager.EstimatedClockDriftPpm());
 
   stats->packet_loss_rate =
       CalculateQ14Ratio(lost_timestamps_, timestamps_since_last_report_);
 
@@ skipping 56 matching lines @@
     // Ratio must be smaller than 1 in Q14.
     assert((numerator << 14) / denominator < (1 << 14));
     return static_cast<uint16_t>((numerator << 14) / denominator);
   } else {
     // Will not produce a ratio larger than 1, since this is probably an error.
     return 1 << 14;
   }
 }
 
 }  // namespace webrtc