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

webrtc/modules/audio_processing/audio_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 44 matching lines @@
     num_proc_channels_(num_process_channels),
     output_num_frames_(output_num_frames),
     num_channels_(num_process_channels),
     num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
     num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
     mixed_low_pass_valid_(false),
     reference_copied_(false),
     activity_(AudioFrame::kVadUnknown),
     keyboard_data_(NULL),
     data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)) {
-  RTC_DCHECK_GT(input_num_frames_, 0u);
-  RTC_DCHECK_GT(proc_num_frames_, 0u);
-  RTC_DCHECK_GT(output_num_frames_, 0u);
-  RTC_DCHECK_GT(num_input_channels_, 0u);
-  RTC_DCHECK_GT(num_proc_channels_, 0u);
+  RTC_DCHECK_GT(input_num_frames_, 0);
+  RTC_DCHECK_GT(proc_num_frames_, 0);
+  RTC_DCHECK_GT(output_num_frames_, 0);
+  RTC_DCHECK_GT(num_input_channels_, 0);
+  RTC_DCHECK_GT(num_proc_channels_, 0);
   RTC_DCHECK_LE(num_proc_channels_, num_input_channels_);
 
   if (input_num_frames_ != proc_num_frames_ ||
       output_num_frames_ != proc_num_frames_) {
     // Create an intermediate buffer for resampling.
     process_buffer_.reset(new ChannelBuffer<float>(proc_num_frames_,
                                                    num_proc_channels_));
 
     if (input_num_frames_ != proc_num_frames_) {
       for (size_t i = 0; i < num_proc_channels_; ++i) {
@@ skipping 387 matching lines @@
 
 void AudioBuffer::SplitIntoFrequencyBands() {
   splitting_filter_->Analysis(data_.get(), split_data_.get());
 }
 
 void AudioBuffer::MergeFrequencyBands() {
   splitting_filter_->Synthesis(split_data_.get(), data_.get());
 }
 
 }  // namespace webrtc