AGC2 as a new APM sub-module operating with hard-coded gain.

webrtc/modules/audio_processing/agc2/digital_gain_applier.h

Index: webrtc/modules/audio_processing/agc2/digital_gain_applier.h
diff --git a/webrtc/modules/audio_processing/agc2/digital_gain_applier.h b/webrtc/modules/audio_processing/agc2/digital_gain_applier.h
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index 0000000000000000000000000000000000000000..fb7ecbbb88c51328316bc3a70dfd885586a0d1b6
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+++ b/webrtc/modules/audio_processing/agc2/digital_gain_applier.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 WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+class DigitalGainApplier {
+ public:
+  DigitalGainApplier();
+
+  // Applies the specified gain to an array of samples.
+  void Process(float gain, rtc::ArrayView<float> samples);
+
+ private:
+  void LimitToAllowedRange(rtc::ArrayView<float> x);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_