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

webrtc/audio/audio_transport_proxy.cc

 /*
  *  Copyright (c) 2016 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 50 matching lines @@
 
 int32_t AudioTransportProxy::NeedMorePlayData(const size_t nSamples,
                                               const size_t nBytesPerSample,
                                               const size_t nChannels,
                                               const uint32_t samplesPerSec,
                                               void* audioSamples,
                                               size_t& nSamplesOut,
                                               int64_t* elapsed_time_ms,
                                               int64_t* ntp_time_ms) {
   RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
-  RTC_DCHECK_GE(nChannels, 1u);
-  RTC_DCHECK_LE(nChannels, 2u);
+  RTC_DCHECK_GE(nChannels, 1);
+  RTC_DCHECK_LE(nChannels, 2);
   RTC_DCHECK_GE(
       samplesPerSec,
       static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
 
   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
   RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
                 sizeof(AudioFrame::data_));
 
   mixer_->Mix(nChannels, &mixed_frame_);
@@ skipping 21 matching lines @@
 }
 
 void AudioTransportProxy::PullRenderData(int bits_per_sample,
                                          int sample_rate,
                                          size_t number_of_channels,
                                          size_t number_of_frames,
                                          void* audio_data,
                                          int64_t* elapsed_time_ms,
                                          int64_t* ntp_time_ms) {
   RTC_DCHECK_EQ(static_cast<size_t>(bits_per_sample), 16);
-  RTC_DCHECK_GE(number_of_channels, 1u);
-  RTC_DCHECK_LE(number_of_channels, 2u);
+  RTC_DCHECK_GE(number_of_channels, 1);
+  RTC_DCHECK_LE(number_of_channels, 2);
   RTC_DCHECK_GE(static_cast<int>(sample_rate),
                 AudioProcessing::NativeRate::kSampleRate8kHz);
 
   // 100 = 1 second / data duration (10 ms).
   RTC_DCHECK_EQ(static_cast<int>(number_of_frames * 100), sample_rate);
 
   // 8 = bits per byte.
   RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
                 sizeof(AudioFrame::data_));
   mixer_->Mix(number_of_channels, &mixed_frame_);
   *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
   *ntp_time_ms = mixed_frame_.ntp_time_ms_;
 
   const auto output_samples = Resample(mixed_frame_, sample_rate, &resampler_,
                                        static_cast<int16_t*>(audio_data));
   RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
 }
 
 }  // namespace webrtc