RTC_[D]CHECK_op: Remove superfluous casts

webrtc/modules/audio_processing/vad/voice_activity_detector.cc

 /*
  *  Copyright (c) 2015 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 21 matching lines @@
 }
 
 VoiceActivityDetector::~VoiceActivityDetector() = default;
 
 // Because ISAC has a different chunk length, it updates
 // |chunkwise_voice_probabilities_| and |chunkwise_rms_| when there is new data.
 // Otherwise it clears them.
 void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
                                          size_t length,
                                          int sample_rate_hz) {
-  RTC_DCHECK_EQ(static_cast<int>(length), sample_rate_hz / 100);
+  RTC_DCHECK_EQ(length, sample_rate_hz / 100);
   RTC_DCHECK_LE(length, kMaxLength);
   // Resample to the required rate.
   const int16_t* resampled_ptr = audio;
   if (sample_rate_hz != kSampleRateHz) {
     RTC_CHECK_EQ(
         resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
         0);
     resampler_.Push(audio, length, resampled_, kLength10Ms, length);
     resampled_ptr = resampled_;
   }
@@ skipping 25 matching lines @@
           0);
       RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
                        features_, &chunkwise_voice_probabilities_[0]),
                    0);
     }
     last_voice_probability_ = chunkwise_voice_probabilities_.back();
   }
 }
 
 }  // namespace webrtc