Fixing neteq_rtpplay

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

 /*
  *  Copyright (c) 2013 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 350 matching lines @@
         std::cerr << "Payload type " <<
             static_cast<int>(rtp_header->header.payloadType) <<
             " not supported or unknown." << std::endl;
         Trace::ReturnTrace();
         exit(1);
     }
   }
   return payload_len;
 }
 
-}  // namespace
-
-int main(int argc, char* argv[]) {
+int RunTest(int argc, char* argv[]) {
   static const int kOutputBlockSizeMs = 10;
 
   std::string program_name = argv[0];
   std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
       "Run " + program_name + " --helpshort for usage.\n"
       "Example usage:\n" + program_name +
       " input.rtp output.{pcm, wav}\n";
   google::SetUsageMessage(usage);
   google::ParseCommandLineFlags(&argc, &argv, true);
 
@@ skipping 241 matching lines @@
   }
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n",
          static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   Trace::ReturnTrace();
   return 0;
 }
 
+}  // namespace
 }  // namespace test
 }  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  webrtc::test::RunTest(argc, argv);
+}