RTC_[D]CHECK_op: Remove superfluous casts

webrtc/modules/audio_device/fine_audio_buffer.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 88 matching lines @@
   RTC_CHECK_LE(bytes_left, 0);
   // Put the samples that were written to |buffer| but are not used in the
   // cache.
   size_t cache_location = desired_frame_size_bytes_;
   int8_t* cache_ptr = &buffer[cache_location];
   playout_cached_bytes_ = number_of_requests * bytes_per_10_ms_ -
                           (desired_frame_size_bytes_ - playout_cached_bytes_);
   // If playout_cached_bytes_ is larger than the cache buffer, uninitialized
   // memory will be read.
   RTC_CHECK_LE(playout_cached_bytes_, bytes_per_10_ms_);
-  RTC_CHECK_EQ(static_cast<size_t>(-bytes_left), playout_cached_bytes_);
+  RTC_CHECK_EQ(-bytes_left, playout_cached_bytes_);
   playout_cached_buffer_start_ = 0;
   memcpy(playout_cache_buffer_.get(), cache_ptr, playout_cached_bytes_);
 }
 
 void FineAudioBuffer::DeliverRecordedData(const int8_t* buffer,
                                           size_t size_in_bytes,
                                           int playout_delay_ms,
                                           int record_delay_ms) {
   // Check if the temporary buffer can store the incoming buffer. If not,
   // move the remaining (old) bytes to the beginning of the temporary buffer
@@ skipping 20 matching lines @@
     device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms, 0);
     device_buffer_->DeliverRecordedData();
     // Read next chunk of 10ms data.
     record_read_pos_ += bytes_per_10_ms_;
     // Reduce number of cached bytes with the consumed amount.
     record_cached_bytes_ -= bytes_per_10_ms_;
   }
 }
 
 }  // namespace webrtc