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

webrtc/modules/audio_processing/echo_control_mobile_impl.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 136 matching lines @@
 
     buffer_index += num_frames_per_band;
   }
 }
 
 void EchoControlMobileImpl::PackRenderAudioBuffer(
     const AudioBuffer* audio,
     size_t num_output_channels,
     size_t num_channels,
     std::vector<int16_t>* packed_buffer) {
-  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_GE(160, audio->num_frames_per_band());
   RTC_DCHECK_EQ(num_channels, audio->num_channels());
 
   // The ordering convention must be followed to pass to the correct AECM.
   packed_buffer->clear();
   int render_channel = 0;
   for (size_t i = 0; i < num_output_channels; i++) {
     for (size_t j = 0; j < audio->num_channels(); j++) {
       // Buffer the samples in the render queue.
       packed_buffer->insert(
           packed_buffer->end(),
@@ skipping 12 matching lines @@
 }
 
 int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
                                                int stream_delay_ms) {
   rtc::CritScope cs_capture(crit_capture_);
   if (!enabled_) {
     return AudioProcessing::kNoError;
   }
 
   RTC_DCHECK(stream_properties_);
-  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+  RTC_DCHECK_GE(160, audio->num_frames_per_band());
   RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_output_channels);
   RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_reverse_channels *
                                         audio->num_channels());
 
   int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AECM.
   size_t handle_index = 0;
   for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
     // TODO(ajm): improve how this works, possibly inside AECM.
@@ skipping 181 matching lines @@
   for (auto& canceller : cancellers_) {
     int handle_error = WebRtcAecm_set_config(canceller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = handle_error;
     }
   }
   return error;
 }
 
 }  // namespace webrtc