RTC_[D]CHECK_op: Remove superfluous casts

webrtc/test/fake_network_pipe_unittest.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.
  */
 
@@ skipping 52 matching lines @@
   virtual void SetUp() {
     receiver_.reset(new TestReceiver());
     ON_CALL(*receiver_, DeliverPacket(_, _, _, _))
         .WillByDefault(Return(PacketReceiver::DELIVERY_OK));
   }
 
   virtual void TearDown() {
   }
 
   void SendPackets(FakeNetworkPipe* pipe, int number_packets, int packet_size) {
-    RTC_DCHECK_GE(packet_size, static_cast<int>(sizeof(int)));
+    RTC_DCHECK_GE(packet_size, sizeof(int));
     std::unique_ptr<uint8_t[]> packet(new uint8_t[packet_size]);
     for (int i = 0; i < number_packets; ++i) {
       // Set a sequence number for the packets by
       // using the first bytes in the packet.
       memcpy(packet.get(), &i, sizeof(int));
       pipe->SendPacket(packet.get(), packet_size);
     }
   }
 
   int PacketTimeMs(int capacity_kbps, int packet_size) const {
@@ skipping 329 matching lines @@
                receiver->delivered_sequence_numbers_[i];
     if (diff > 1)
       ++num_bursts;
   }
 
   double average_burst_length = static_cast<double>(lost_packets) / num_bursts;
 
   EXPECT_NEAR(kAvgBurstLength, average_burst_length, 0.3);
 }
 }  // namespace webrtc