Re-enable PCAP reading in 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 381 matching lines @@
       // We have already printed the codec map. Just end the program.
       return 0;
     }
     // Print usage information.
     std::cout << google::ProgramUsage();
     return 0;
   }
 
   printf("Input file: %s\n", argv[1]);
 
-  // TODO(ivoc): Modify the RtpFileSource::Create and RtcEventLogSource::Create
-  //             functions to return a nullptr on failure instead of crashing
-  //             the program.
-
-  // This temporary solution uses a RtpFileReader directly to check if the file
-  // is a valid RtpDump file.
   bool is_rtp_dump = false;
-  {
-    rtc::scoped_ptr<webrtc::test::RtpFileReader> rtp_reader(
-        webrtc::test::RtpFileReader::Create(
-            webrtc::test::RtpFileReader::kRtpDump, argv[1]));
-    if (rtp_reader)
-      is_rtp_dump = true;
-  }
   rtc::scoped_ptr<webrtc::test::PacketSource> file_source;
   webrtc::test::RtcEventLogSource* event_log_source = nullptr;
-  if (is_rtp_dump) {
+  if (webrtc::test::RtpFileSource::ValidRtpDump(argv[1]) ||
+      webrtc::test::RtpFileSource::ValidPcap(argv[1])) {
+    is_rtp_dump = true;
     file_source.reset(webrtc::test::RtpFileSource::Create(argv[1]));
   } else {
     event_log_source = webrtc::test::RtcEventLogSource::Create(argv[1]);
     file_source.reset(event_log_source);
   }
 
   assert(file_source.get());
 
   // Check if an SSRC value was provided.
   if (!FLAGS_ssrc.empty()) {
@@ skipping 213 matching lines @@
     }
   }
   printf("Simulation done\n");
   printf("Produced %i ms of audio\n",
          static_cast<int>(time_now_ms - start_time_ms));
 
   delete neteq;
   webrtc::Trace::ReturnTrace();
   return 0;
 }