RTC_[D]CHECK_op: Remove superfluous casts

webrtc/modules/audio_processing/level_controller/signal_classifier.cc

 /*
  *  Copyright (c) 2016 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 111 matching lines @@
   noise_spectrum_estimator_.Initialize();
   frame_extender_.reset(new FrameExtender(80, 128));
   sample_rate_hz_ = sample_rate_hz;
   initialization_frames_left_ = 2;
   consistent_classification_counter_ = 3;
   last_signal_type_ = SignalClassifier::SignalType::kNonStationary;
 }
 
 void SignalClassifier::Analyze(const AudioBuffer& audio,
                                SignalType* signal_type) {
-  RTC_DCHECK_EQ(audio.num_frames(), static_cast<size_t>(sample_rate_hz_ / 100));
+  RTC_DCHECK_EQ(audio.num_frames(), sample_rate_hz_ / 100);
 
   // Compute the signal power spectrum.
   float downsampled_frame[80];
   down_sampler_.DownSample(rtc::ArrayView<const float>(
                                audio.channels_const_f()[0], audio.num_frames()),
                            downsampled_frame);
   float extended_frame[128];
   frame_extender_->ExtendFrame(downsampled_frame, extended_frame);
   RemoveDcLevel(extended_frame);
   float signal_spectrum[65];
@@ skipping 19 matching lines @@
     last_signal_type_ = *signal_type;
     consistent_classification_counter_ = 3;
   }
 
   if (consistent_classification_counter_ > 0) {
     *signal_type = SignalClassifier::SignalType::kNonStationary;
   }
 }
 
 }  // namespace webrtc