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

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.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.
  */
 
 #include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
 namespace test {
 
 int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
                                        size_t encoded_len,
                                        int sample_rate_hz,
                                        int16_t* decoded,
                                        SpeechType* speech_type) {
   if (encoded_len == 0) {
     // Decoder is asked to produce codec-internal comfort noise.
     RTC_DCHECK(!encoded);  // NetEq always sends nullptr in this case.
     RTC_DCHECK(cng_mode_);
-    RTC_DCHECK_GT(last_decoded_length_, 0u);
+    RTC_DCHECK_GT(last_decoded_length_, 0);
     std::fill_n(decoded, last_decoded_length_, 0);
     *speech_type = kComfortNoise;
     return last_decoded_length_;
   }
 
-  RTC_CHECK_GE(encoded_len, 12u);
+  RTC_CHECK_GE(encoded_len, 12);
   uint32_t timestamp_to_decode =
       ByteReader<uint32_t>::ReadLittleEndian(encoded);
   uint32_t samples_to_decode =
       ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
   if (samples_to_decode == 0) {
     // Number of samples in packet is unknown.
     if (last_decoded_length_ > 0) {
       // Use length of last decoded packet, but since this is the total for all
       // channels, we have to divide by 2 in the stereo case.
       samples_to_decode = rtc::CheckedDivExact(
@@ skipping 13 matching lines @@
     RTC_CHECK(input_->Seek(jump));
   }
 
   next_timestamp_from_input_ =
       rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode);
 
   uint32_t original_payload_size_bytes =
       ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
   if (original_payload_size_bytes == 1) {
     // This is a comfort noise payload.
-    RTC_DCHECK_GT(last_decoded_length_, 0u);
+    RTC_DCHECK_GT(last_decoded_length_, 0);
     std::fill_n(decoded, last_decoded_length_, 0);
     *speech_type = kComfortNoise;
     cng_mode_ = true;
     return last_decoded_length_;
   }
 
   cng_mode_ = false;
   RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
 
   if (stereo_) {
     InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
                                          decoded);
     samples_to_decode *= 2;
   }
 
   *speech_type = kSpeech;
   return last_decoded_length_ = samples_to_decode;
 }
 
 void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
                                         size_t samples,
                                         size_t original_payload_size_bytes,
                                         rtc::ArrayView<uint8_t> encoded) {
-  RTC_CHECK_GE(encoded.size(), 12u);
+  RTC_CHECK_GE(encoded.size(), 12);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
   ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
                                           rtc::checked_cast<uint32_t>(samples));
   ByteWriter<uint32_t>::WriteLittleEndian(
       &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
 }
 
 }  // namespace test
 }  // namespace webrtc