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diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h
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+++ b/modules/audio_coding/neteq/tools/rtp_generator.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_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for generating RTP headers.
+class RtpGenerator {
+ public:
+  RtpGenerator(int samples_per_ms,
+               uint16_t start_seq_number = 0,
+               uint32_t start_timestamp = 0,
+               uint32_t start_send_time_ms = 0,
+               uint32_t ssrc = 0x12345678)
+      : seq_number_(start_seq_number),
+        timestamp_(start_timestamp),
+        next_send_time_ms_(start_send_time_ms),
+        ssrc_(ssrc),
+        samples_per_ms_(samples_per_ms),
+        drift_factor_(0.0) {
+  }
+
+  // Writes the next RTP header to |rtp_header|, which will be of type
+  // |payload_type|. Returns the send time for this packet (in ms). The value of
+  // |payload_length_samples| determines the send time for the next packet.
+  uint32_t GetRtpHeader(uint8_t payload_type, size_t payload_length_samples,
+                        WebRtcRTPHeader* rtp_header);
+
+  void set_drift_factor(double factor);
+
+ private:
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  uint32_t next_send_time_ms_;
+  const uint32_t ssrc_;
+  const int samples_per_ms_;
+  double drift_factor_;
+  DISALLOW_COPY_AND_ASSIGN(RtpGenerator);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_