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

webrtc/modules/audio_processing/agc/agc.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 21 matching lines @@
 
 Agc::Agc()
     : target_level_loudness_(Dbfs2Loudness(kDefaultLevelDbfs)),
       target_level_dbfs_(kDefaultLevelDbfs),
       histogram_(LoudnessHistogram::Create(kNumAnalysisFrames)),
       inactive_histogram_(LoudnessHistogram::Create()) {}
 
 Agc::~Agc() {}
 
 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
-  RTC_DCHECK_GT(length, 0u);
+  RTC_DCHECK_GT(length, 0);
   size_t num_clipped = 0;
   for (size_t i = 0; i < length; ++i) {
     if (audio[i] == 32767 || audio[i] == -32768)
       ++num_clipped;
   }
   return 1.0f * num_clipped / length;
 }
 
 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
   vad_.ProcessChunk(audio, length, sample_rate_hz);
@@ skipping 46 matching lines @@
 
 int Agc::target_level_dbfs() const {
   return target_level_dbfs_;
 }
 
 float Agc::voice_probability() const {
   return vad_.last_voice_probability();
 }
 
 }  // namespace webrtc