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diff --git a/webrtc/modules/audio_coding/test/TestAllCodecs.h b/webrtc/modules/audio_coding/test/TestAllCodecs.h
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+++ b/webrtc/modules/audio_coding/test/TestAllCodecs.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_TESTALLCODECS_H_
+#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_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"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Config;
+
+class TestPack : public AudioPacketizationCallback {
+ public:
+  TestPack();
+  ~TestPack();
+
+  void RegisterReceiverACM(AudioCodingModule* acm);
+
+  int32_t SendData(FrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size,
+                   const RTPFragmentationHeader* fragmentation) override;
+
+  size_t payload_size();
+  uint32_t timestamp_diff();
+  void reset_payload_size();
+
+ private:
+  AudioCodingModule* receiver_acm_;
+  uint16_t sequence_number_;
+  uint8_t payload_data_[60 * 32 * 2 * 2];
+  uint32_t timestamp_diff_;
+  uint32_t last_in_timestamp_;
+  uint64_t total_bytes_;
+  size_t payload_size_;
+};
+
+class TestAllCodecs : public ACMTest {
+ public:
+  explicit TestAllCodecs(int test_mode);
+  ~TestAllCodecs();
+
+  void Perform() override;
+
+ private:
+  // The default value of '-1' indicates that the registration is based only on
+  // codec name, and a sampling frequency matching is not required.
+  // This is useful for codecs which support several sampling frequency.
+  // Note! Only mono mode is tested in this test.
+  void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
+                         int rate, int packet_size, size_t extra_byte);
+
+  void Run(TestPack* channel);
+  void OpenOutFile(int test_number);
+  void DisplaySendReceiveCodec();
+
+  int test_mode_;
+  rtc::scoped_ptr<AudioCodingModule> acm_a_;
+  rtc::scoped_ptr<AudioCodingModule> acm_b_;
+  TestPack* channel_a_to_b_;
+  PCMFile infile_a_;
+  PCMFile outfile_b_;
+  int test_count_;
+  int packet_size_samples_;
+  size_t packet_size_bytes_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_