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

webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc

 /*
  *  Copyright (c) 2015 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 58 matching lines @@
     timestamps_per_frame = 160;
   } else if (payload.size() % 50 == 0) {
     // 30 ms frames.
     bytes_per_frame = 50;
     timestamps_per_frame = 240;
   } else {
     LOG(LS_WARNING) << "AudioDecoderIlbc::ParsePayload: Invalid payload";
     return results;
   }
 
-  RTC_DCHECK_EQ(0u, payload.size() % bytes_per_frame);
+  RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
   if (payload.size() == bytes_per_frame) {
     std::unique_ptr<EncodedAudioFrame> frame(
         new LegacyEncodedAudioFrame(this, std::move(payload)));
     results.emplace_back(timestamp, 0, std::move(frame));
   } else {
     size_t byte_offset;
     uint32_t timestamp_offset;
     for (byte_offset = 0, timestamp_offset = 0;
          byte_offset < payload.size();
          byte_offset += bytes_per_frame,
              timestamp_offset += timestamps_per_frame) {
       std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
           this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
       results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
     }
   }
 
   return results;
 }
 
 int AudioDecoderIlbc::SampleRateHz() const {
   return 8000;
 }
 
 size_t AudioDecoderIlbc::Channels() const {
   return 1;
 }
 
 }  // namespace webrtc