Introduce dchecked_cast, and start using it

webrtc/modules/audio_coding/neteq/expand.cc

Index: webrtc/modules/audio_coding/neteq/expand.cc
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index 2154bfde637179f522f3f2d44c30b97ac244ad82..0c527fe041de64a7a58541582ea562f915bbaf61 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -222,7 +222,7 @@ int Expand::Process(AudioMultiVector* output) {
     //   >= 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;
@@ -315,8 +315,8 @@ int Expand::Process(AudioMultiVector* output) {
       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;
 }
 
@@ -325,7 +325,7 @@ void Expand::SetParametersForNormalAfterExpand() {
   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() {