Update includes for webrtc/{base => rtc_base} rename (1/3)

webrtc/modules/audio_coding/neteq/tools/rtpcat.cc

 /*
  *  Copyright (c) 2011 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 <stdio.h>
 
 #include <memory>
 
-#include "webrtc/base/checks.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/test/rtp_file_reader.h"
 #include "webrtc/test/rtp_file_writer.h"
 
 using webrtc::test::RtpFileReader;
 using webrtc::test::RtpFileWriter;
 
 int main(int argc, char* argv[]) {
   if (argc < 3) {
     printf("Concatenates multiple rtpdump files into one.\n");
     printf("Usage: rtpcat in1.rtp int2.rtp [...] out.rtp\n");
@@ skipping 10 matching lines @@
         RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
     RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
     printf("Input RTP file: %s\n", argv[i]);
 
     webrtc::test::RtpPacket packet;
     while (input->NextPacket(&packet))
       RTC_CHECK(output->WritePacket(&packet));
   }
   return 0;
 }