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diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.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_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+class AudioCodingModule;
+class AudioDecoder;
+struct CodecInst;
+
+namespace test {
+class AudioSink;
+class PacketSource;
+
+class AcmReceiveTestOldApi {
+ public:
+  enum NumOutputChannels {
+    kArbitraryChannels = 0,
+    kMonoOutput = 1,
+    kStereoOutput = 2
+  };
+
+  AcmReceiveTestOldApi(PacketSource* packet_source,
+                       AudioSink* audio_sink,
+                       int output_freq_hz,
+                       NumOutputChannels exptected_output_channels);
+  virtual ~AcmReceiveTestOldApi() {}
+
+  // Registers the codecs with default parameters from ACM.
+  void RegisterDefaultCodecs();
+
+  // Registers codecs with payload types matching the pre-encoded NetEq test
+  // files.
+  void RegisterNetEqTestCodecs();
+
+  int RegisterExternalReceiveCodec(int rtp_payload_type,
+                                   AudioDecoder* external_decoder,
+                                   int sample_rate_hz,
+                                   int num_channels,
+                                   const std::string& name);
+
+  // Runs the test and returns true if successful.
+  void Run();
+
+ protected:
+  // Method is called after each block of output audio is received from ACM.
+  virtual void AfterGetAudio() {}
+
+  SimulatedClock clock_;
+  rtc::scoped_ptr<AudioCodingModule> acm_;
+  PacketSource* packet_source_;
+  AudioSink* audio_sink_;
+  int output_freq_hz_;
+  NumOutputChannels exptected_output_channels_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(AcmReceiveTestOldApi);
+};
+
+// This test toggles the output frequency every |toggle_period_ms|. The test
+// starts with |output_freq_hz_1|. Except for the toggling, it does the same
+// thing as AcmReceiveTestOldApi.
+class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
+ public:
+  AcmReceiveTestToggleOutputFreqOldApi(
+      PacketSource* packet_source,
+      AudioSink* audio_sink,
+      int output_freq_hz_1,
+      int output_freq_hz_2,
+      int toggle_period_ms,
+      NumOutputChannels exptected_output_channels);
+
+ protected:
+  void AfterGetAudio() override;
+
+  const int output_freq_hz_1_;
+  const int output_freq_hz_2_;
+  const int toggle_period_ms_;
+  int64_t last_toggle_time_ms_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_