RTC_[D]CHECK_op: Remove superfluous casts

webrtc/modules/audio_processing/level_controller/down_sampler.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 51 matching lines @@
   } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
     low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
   } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) {
     low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
   }
 }
 
 void DownSampler::DownSample(rtc::ArrayView<const float> in,
                              rtc::ArrayView<float> out) {
   data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
-  RTC_DCHECK_EQ(static_cast<size_t>(sample_rate_hz_ *
-                                    AudioProcessing::kChunkSizeMs / 1000),
+  RTC_DCHECK_EQ(sample_rate_hz_ * AudioProcessing::kChunkSizeMs / 1000,
                 in.size());
-  RTC_DCHECK_EQ(static_cast<size_t>(AudioProcessing::kSampleRate8kHz *
-                                    AudioProcessing::kChunkSizeMs / 1000),
-                out.size());
+  RTC_DCHECK_EQ(
+      AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000,
+      out.size());
   const size_t kMaxNumFrames =
       AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000;
   float x[kMaxNumFrames];
 
   // Band-limit the signal to 4 kHz.
   if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) {
     low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
 
     // Downsample the signal.
     size_t k = 0;
     for (size_t j = 0; j < out.size(); ++j) {
       RTC_DCHECK_GT(kMaxNumFrames, k);
       out[j] = x[k];
       k += down_sampling_factor_;
     }
   } else {
     std::copy(in.data(), in.data() + in.size(), out.data());
   }
 
   data_dumper_->DumpWav("lc_down_sampler_output", out,
                         AudioProcessing::kSampleRate8kHz, 1);
 }
 
 }  // namespace webrtc