diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index 0aaf8c71fd..adce1cfd79 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -399,23 +399,12 @@ int main(int argc, char* argv[]) {
 
   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]);
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 9681ad17ea..e3a829bd24 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -32,6 +32,18 @@ RtpFileSource* RtpFileSource::Create(const std::string& file_name) {
   return source;
 }
 
+bool RtpFileSource::ValidRtpDump(const std::string& file_name) {
+  rtc::scoped_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  return !!temp_file;
+}
+
+bool RtpFileSource::ValidPcap(const std::string& file_name) {
+  rtc::scoped_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  return !!temp_file;
+}
+
 RtpFileSource::~RtpFileSource() {
 }
 
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index d0856a819c..cd7d7e874c 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -34,6 +34,10 @@ class RtpFileSource : public PacketSource {
   // opened, or has the wrong format, NULL will be returned.
   static RtpFileSource* Create(const std::string& file_name);
 
+  // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
+  static bool ValidRtpDump(const std::string& file_name);
+  static bool ValidPcap(const std::string& file_name);
+
   virtual ~RtpFileSource();
 
   // Registers an RTP header extension and binds it to |id|.