Introduce dchecked_cast, and start using it

webrtc/modules/audio_coding/neteq/expand.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.
  */
 
@@ skipping 204 matching lines @@
 
     // Combine voiced and unvoiced contributions.
 
     // Set a suitable cross-fading slope.
     // For lag =
     //   <= 31 * fs_mult            => go from 1 to 0 in about 8 ms;
     //  (>= 31 .. <= 63) * fs_mult  => go from 1 to 0 in about 16 ms;
     //   >= 64 * fs_mult            => go from 1 to 0 in about 32 ms.
     // temp_shift = getbits(max_lag_) - 5.
     int temp_shift =
-        (31 - WebRtcSpl_NormW32(rtc::checked_cast<int32_t>(max_lag_))) - 5;
+        (31 - WebRtcSpl_NormW32(rtc::dchecked_cast<int32_t>(max_lag_))) - 5;
     int16_t mix_factor_increment = 256 >> temp_shift;
     if (stop_muting_) {
       mix_factor_increment = 0;
     }
 
     // Create combined signal by shifting in more and more of unvoiced part.
     temp_shift = 8 - temp_shift;  // = getbits(mix_factor_increment).
     size_t temp_length = (parameters.current_voice_mix_factor -
         parameters.voice_mix_factor) >> temp_shift;
     temp_length = std::min(temp_length, current_lag);
@@ skipping 72 matching lines @@
       assert(output->Size() == current_lag);
     }
     (*output)[channel_ix].OverwriteAt(temp_data, current_lag, 0);
   }
 
   // Increase call number and cap it.
   consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ?
       kMaxConsecutiveExpands : consecutive_expands_ + 1;
   expand_duration_samples_ += output->Size();
   // Clamp the duration counter at 2 seconds.
-  expand_duration_samples_ =
-      std::min(expand_duration_samples_, rtc::checked_cast<size_t>(fs_hz_ * 2));
+  expand_duration_samples_ = std::min(expand_duration_samples_,
+                                      rtc::dchecked_cast<size_t>(fs_hz_ * 2));
   return 0;
 }
 
 void Expand::SetParametersForNormalAfterExpand() {
   current_lag_index_ = 0;
   lag_index_direction_ = 0;
   stop_muting_ = true;  // Do not mute signal any more.
   statistics_->LogDelayedPacketOutageEvent(
-      rtc::checked_cast<int>(expand_duration_samples_) / (fs_hz_ / 1000));
+      rtc::dchecked_cast<int>(expand_duration_samples_) / (fs_hz_ / 1000));
 }
 
 void Expand::SetParametersForMergeAfterExpand() {
   current_lag_index_ = -1; /* out of the 3 possible ones */
   lag_index_direction_ = 1; /* make sure we get the "optimal" lag */
   stop_muting_ = true;
 }
 
 bool Expand::Muted() const {
   if (first_expand_ || stop_muting_)
@@ skipping 630 matching lines @@
   const size_t kMaxRandSamples = RandomVector::kRandomTableSize;
   while (samples_generated < length) {
     size_t rand_length = std::min(length - samples_generated, kMaxRandSamples);
     random_vector_->IncreaseSeedIncrement(seed_increment);
     random_vector_->Generate(rand_length, &random_vector[samples_generated]);
     samples_generated += rand_length;
   }
 }
 
 }  // namespace webrtc