New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/level_controller/gain_applier.h

Index: webrtc/modules/audio_processing/level_controller/gain_applier.h
diff --git a/webrtc/modules/audio_processing/level_controller/gain_applier.h b/webrtc/modules/audio_processing/level_controller/gain_applier.h
new file mode 100644
index 0000000000000000000000000000000000000000..decd1eb58c1c1d0a9b785be8c19c3cd83fdb9d92
--- /dev/null
+++ b/webrtc/modules/audio_processing/level_controller/gain_applier.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2016 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_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
+
+#include "webrtc/base/constructormagic.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class GainApplier {
+ public:
+  explicit GainApplier(ApmDataDumper* data_dumper);
+  void Initialize(int sample_rate_hz);
+
+  // Applies the specified gain to the audio frame and returns the resulting
+  // number of saturated sample values.
+  int Process(float new_gain, AudioBuffer* audio);
+
+ private:
+  ApmDataDumper* const data_dumper_;
+  float old_gain_ = 1.f;
+  float gain_change_step_size_ = 0.f;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_