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diff --git a/webrtc/modules/audio_processing/gain_controller2.h b/webrtc/modules/audio_processing/gain_controller2.h
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+/*
+ *  Copyright (c) 2017 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 MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_processing/agc2/adaptive_agc.h"
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/agc2/limiter.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+// Gain Controller 2 aims to automatically adjust levels by acting on the
+// microphone gain and/or applying digital gain.
+class GainController2 {
+ public:
+  GainController2();
+  ~GainController2();
+
+  void Initialize(int sample_rate_hz);
+  void Process(AudioBuffer* audio);
+  void NotifyAnalogLevel(int level);
+
+  void ApplyConfig(const AudioProcessing::Config::GainController2& config);
+  static bool Validate(const AudioProcessing::Config::GainController2& config);
+  static std::string ToString(
+      const AudioProcessing::Config::GainController2& config);
+
+ private:
+  static int instance_count_;
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+  AudioProcessing::Config::GainController2 config_;
+  GainApplier gain_applier_;
+  std::unique_ptr<AdaptiveAgc> adaptive_agc_;
+  Limiter limiter_;
+  int analog_level_ = -1;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_