RTC_[D]CHECK_op: Remove "u" suffix on integer constants

webrtc/modules/audio_device/audio_device_buffer.cc

 /*
  *  Copyright (c) 2012 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 391 matching lines @@
   // zero in most calls and then have no effect of the stats. It is only updated
   // approximately two times per second and can then change the stats.
   task_queue_.PostTask([this, max_abs, num_samples_out] {
     UpdatePlayStats(max_abs, num_samples_out);
   });
   return static_cast<int32_t>(num_samples_out);
 }
 
 int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
   RTC_DCHECK_RUN_ON(&playout_thread_checker_);
-  RTC_DCHECK_GT(play_buffer_.size(), 0u);
+  RTC_DCHECK_GT(play_buffer_.size(), 0);
   const size_t bytes_per_sample = sizeof(int16_t);
   memcpy(audio_buffer, play_buffer_.data(),
          play_buffer_.size() * bytes_per_sample);
   // Return samples per channel or number of frames.
   return static_cast<int32_t>(play_buffer_.size() / play_channels_);
 }
 
 void AudioDeviceBuffer::StartPeriodicLogging() {
   task_queue_.PostTask(rtc::Bind(&AudioDeviceBuffer::LogStats, this,
                                  AudioDeviceBuffer::LOG_START));
@@ skipping 102 matching lines @@
                                         size_t samples_per_channel) {
   RTC_DCHECK_RUN_ON(&task_queue_);
   ++play_callbacks_;
   play_samples_ += samples_per_channel;
   if (max_abs > max_play_level_) {
     max_play_level_ = max_abs;
   }
 }
 
 }  // namespace webrtc