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

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.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.
  */
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h"
@@ skipping 229 matching lines @@
 }
 
 // Generates a frame of packets belonging to a stream at a given bitrate and
 // with a given ssrc. The stream is pushed through a very simple simulated
 // network, and is then given to the receive-side bandwidth estimator.
 // Returns true if an over-use was seen, false otherwise.
 // The StreamGenerator::updated() should be used to check for any changes in
 // target bitrate after the call to this function.
 bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(uint32_t ssrc,
                                                          uint32_t bitrate_bps) {
-  RTC_DCHECK_GT(bitrate_bps, 0u);
+  RTC_DCHECK_GT(bitrate_bps, 0);
   stream_generator_->SetBitrateBps(bitrate_bps);
   testing::RtpStream::PacketList packets;
   int64_t next_time_us = stream_generator_->GenerateFrame(
       &packets, clock_.TimeInMicroseconds());
   bool overuse = false;
   while (!packets.empty()) {
     testing::RtpStream::RtpPacket* packet = packets.front();
     bitrate_observer_->Reset();
     // The simulated clock should match the time of packet->arrival_time
     // since both are used in IncomingPacket().
@@ skipping 345 matching lines @@
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     absolute_send_time = AddAbsSendTime(absolute_send_time,
                                         kFrameIntervalAbsSendTime);
     bitrate_estimator_->Process();
   }
   uint32_t bitrate_after = 0;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc