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

webrtc/modules/audio_processing/agc2/digital_gain_applier.h

+/*
+ *  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 the audio frame.
+  void Process(float gain, AudioBuffer* audio);

[peah-webrtc, 2017/05/18 10:51:19]

I think we should avoid having AudioBuffer as an input for more than the first
layer of AGC2. The reason for that is that we'll need to change it in the not so
far future, and if it is integrated deep into AGC2 that change will be more
difficult. 

Instead let Process take arrayview objects as input.

[AleBzk, 2017/05/18 12:06:22]

Done.

+
+ private:
+  void ApplyGain(float gain, rtc::ArrayView<float> x);
+  void LimitToAllowedRange(rtc::ArrayView<float> x);
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_