RTC_[D]CHECK_op: Remove superfluous casts

webrtc/modules/audio_device/android/audio_track_jni.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 237 matching lines @@
   if (!audio_device_buffer_) {
     ALOGE("AttachAudioBuffer has not been called!");
     return;
   }
   // Pull decoded data (in 16-bit PCM format) from jitter buffer.
   int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
   if (samples <= 0) {
     ALOGE("AudioDeviceBuffer::RequestPlayoutData failed!");
     return;
   }
-  RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_);
+  RTC_DCHECK_EQ(samples, frames_per_buffer_);
   // Copy decoded data into common byte buffer to ensure that it can be
   // written to the Java based audio track.
   samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
   RTC_DCHECK_EQ(length, bytes_per_frame * samples);
 }
 
 }  // namespace webrtc