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diff --git a/webrtc/modules/video_coding/test/rtp_player.h b/webrtc/modules/video_coding/test/rtp_player.h
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+/*
+ *  Copyright (c) 2012 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_VIDEO_CODING_TEST_RTP_PLAYER_H_
+#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/video_coding/include/video_coding_defines.h"
+
+namespace webrtc {
+class Clock;
+
+namespace rtpplayer {
+
+class PayloadCodecTuple {
+ public:
+  PayloadCodecTuple(uint8_t payload_type,
+                    const std::string& codec_name,
+                    VideoCodecType codec_type)
+      : name_(codec_name),
+        payload_type_(payload_type),
+        codec_type_(codec_type) {}
+
+  const std::string& name() const { return name_; }
+  uint8_t payload_type() const { return payload_type_; }
+  VideoCodecType codec_type() const { return codec_type_; }
+
+ private:
+  std::string name_;
+  uint8_t payload_type_;
+  VideoCodecType codec_type_;
+};
+
+typedef std::vector<PayloadCodecTuple> PayloadTypes;
+typedef std::vector<PayloadCodecTuple>::const_iterator PayloadTypesIterator;
+
+// Implemented by RtpPlayer and given to client as a means to retrieve
+// information about a specific RTP stream.
+class RtpStreamInterface {
+ public:
+  virtual ~RtpStreamInterface() {}
+
+  // Ask for missing packets to be resent.
+  virtual void ResendPackets(const uint16_t* sequence_numbers,
+                             uint16_t length) = 0;
+
+  virtual uint32_t ssrc() const = 0;
+  virtual const PayloadTypes& payload_types() const = 0;
+};
+
+// Implemented by a sink. Wraps RtpData because its d-tor is protected.
+class PayloadSinkInterface : public RtpData {
+ public:
+  virtual ~PayloadSinkInterface() {}
+};
+
+// Implemented to provide a sink for RTP data, such as hooking up a VCM to
+// the incoming RTP stream.
+class PayloadSinkFactoryInterface {
+ public:
+  virtual ~PayloadSinkFactoryInterface() {}
+
+  // Return NULL if failed to create sink. 'stream' is guaranteed to be
+  // around for as long as the RtpData. The returned object is owned by
+  // the caller (RtpPlayer).
+  virtual PayloadSinkInterface* Create(RtpStreamInterface* stream) = 0;
+};
+
+// The client's view of an RtpPlayer.
+class RtpPlayerInterface {
+ public:
+  virtual ~RtpPlayerInterface() {}
+
+  virtual int NextPacket(int64_t timeNow) = 0;
+  virtual uint32_t TimeUntilNextPacket() const = 0;
+  virtual void Print() const = 0;
+};
+
+RtpPlayerInterface* Create(const std::string& inputFilename,
+                           PayloadSinkFactoryInterface* payloadSinkFactory,
+                           Clock* clock,
+                           const PayloadTypes& payload_types,
+                           float lossRate,
+                           int64_t rttMs,
+                           bool reordering);
+
+}  // namespace rtpplayer
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_