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

webrtc/common_audio/audio_converter.cc

 /*
  *  Copyright (c) 2014 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 91 matching lines @@
  private:
   std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
 };
 
 // Apply a vector of converters in serial, in the order given. At least two
 // converters must be provided.
 class CompositionConverter : public AudioConverter {
  public:
   CompositionConverter(std::vector<std::unique_ptr<AudioConverter>> converters)
       : converters_(std::move(converters)) {
-    RTC_CHECK_GE(converters_.size(), 2u);
+    RTC_CHECK_GE(converters_.size(), 2);
     // We need an intermediate buffer after every converter.
     for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
       buffers_.push_back(
           std::unique_ptr<ChannelBuffer<float>>(new ChannelBuffer<float>(
               (*it)->dst_frames(), (*it)->dst_channels())));
   }
   ~CompositionConverter() override {};
 
   void Convert(const float* const* src, size_t src_size, float* const* dst,
                size_t dst_capacity) override {
@@ skipping 74 matching lines @@
   RTC_CHECK(dst_channels == src_channels || dst_channels == 1 ||
             src_channels == 1);
 }
 
 void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const {
   RTC_CHECK_EQ(src_size, src_channels() * src_frames());
   RTC_CHECK_GE(dst_capacity, dst_channels() * dst_frames());
 }
 
 }  // namespace webrtc