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diff --git a/webrtc/modules/audio_coding/test/TestStereo.h b/webrtc/modules/audio_coding/test/TestStereo.h
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+++ b/webrtc/modules/audio_coding/test/TestStereo.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_TEST_TESTSTEREO_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+
+#include <math.h>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/test/ACMTest.h"
+#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "webrtc/modules/audio_coding/test/PCMFile.h"
+
+#define PCMA_AND_PCMU
+
+namespace webrtc {
+
+enum StereoMonoMode {
+  kNotSet,
+  kMono,
+  kStereo
+};
+
+class TestPackStereo : public AudioPacketizationCallback {
+ public:
+  TestPackStereo();
+  ~TestPackStereo();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(const FrameType frame_type,
+                   const uint8_t payload_type,
+                   const uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   const size_t payload_size,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+  uint16_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+  void set_codec_mode(StereoMonoMode mode);
+  void set_lost_packet(bool lost);
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  int16_t seq_no_;
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  int payload_size_;
+  StereoMonoMode codec_mode_;
+  // Simulate packet losses
+  bool lost_packet_;
+};
+
+class TestStereo : public ACMTest {
+ public:
+  explicit TestStereo(int test_mode);
+  ~TestStereo();
+
+  void Perform() override;
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name and a sampling frequncy matching is not required. This is useful
+  // for codecs which support several sampling frequency.
+  void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
+                         int rate, int pack_size, int channels,
+                         int payload_type);
+
+  void Run(TestPackStereo* channel, int in_channels, int out_channels,
+           int percent_loss = 0);
+  void OpenOutFile(int16_t test_number);
+  void DisplaySendReceiveCodec();
+
+  int test_mode_;
+
+  rtc::scoped_ptr<AudioCodingModule> acm_a_;
+  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+
+  TestPackStereo* channel_a2b_;
+
+  PCMFile* in_file_stereo_;
+  PCMFile* in_file_mono_;
+  PCMFile out_file_;
+  int16_t test_cntr_;
+  uint16_t pack_size_samp_;
+  uint16_t pack_size_bytes_;
+  int counter_;
+  char* send_codec_name_;
+
+  // Payload types for stereo codecs and CNG
+#ifdef WEBRTC_CODEC_G722
+  int g722_pltype_;
+#endif
+  int l16_8khz_pltype_;
+  int l16_16khz_pltype_;
+  int l16_32khz_pltype_;
+#ifdef PCMA_AND_PCMU
+  int pcma_pltype_;
+  int pcmu_pltype_;
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  int opus_pltype_;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_