Introduce dchecked_cast, and start using it

webrtc/modules/audio_coding/acm2/acm_receiver.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 144 matching lines @@
         audio_frame->data_);
     if (samples_per_channel_int < 0) {
       LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
       return -1;
     }
     audio_frame->samples_per_channel_ =
         static_cast<size_t>(samples_per_channel_int);
     audio_frame->sample_rate_hz_ = desired_freq_hz;
     RTC_DCHECK_EQ(
         audio_frame->sample_rate_hz_,
-        rtc::checked_cast<int>(audio_frame->samples_per_channel_ * 100));
+        rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
     resampled_last_output_frame_ = true;
   } else {
     resampled_last_output_frame_ = false;
     // We might end up here ONLY if codec is changed.
   }
 
   // Store current audio in |last_audio_buffer_| for next time.
   memcpy(last_audio_buffer_.get(), audio_frame->data_,
          sizeof(int16_t) * audio_frame->samples_per_channel_ *
              audio_frame->num_channels_);
@@ skipping 211 matching lines @@
 
 void AcmReceiver::GetDecodingCallStatistics(
     AudioDecodingCallStats* stats) const {
   rtc::CritScope lock(&crit_sect_);
   *stats = call_stats_.GetDecodingStatistics();
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc