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diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
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+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+
+#include <stdio.h>
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+class RtpHeaderParser;
+
+namespace test {
+
+class RtpFileSource : public PacketSource {
+ public:
+  // Creates an RtpFileSource reading from |file_name|. If the file cannot be
+  // opened, or has the wrong format, NULL will be returned.
+  static RtpFileSource* Create(const std::string& file_name);
+
+  virtual ~RtpFileSource();
+
+  // Registers an RTP header extension and binds it to |id|.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
+
+  // Returns a pointer to the next packet.
+  virtual Packet* NextPacket();
+
+  // Returns true if the end of file has been reached.
+  virtual bool EndOfFile() const;
+
+ private:
+  static const int kFirstLineLength = 40;
+  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
+  static const size_t kPacketHeaderSize = 8;
+
+  RtpFileSource();
+
+  bool OpenFile(const std::string& file_name);
+
+  bool SkipFileHeader();
+
+  FILE* in_file_;
+  int64_t file_end_;
+  scoped_ptr<RtpHeaderParser> parser_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_