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

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 249 matching lines @@
       bool enabled = false;
     } high_pass_filter;
 
     // Enables the next generation AEC functionality. This feature replaces the
     // standard methods for echo removal in the AEC.
     // The functionality is not yet activated in the code and turning this on
     // does not yet have the desired behavior.
     struct EchoCanceller3 {
       bool enabled = false;
     } echo_canceller3;
+
+    // AGC2 enables the next generation AGC functionality. This feature replaces
+    // the standard methods of adaptive gain control in the previous AGC.
+    // The functionality is not yet activated in the code and turning this on
+    // does not yet have the desired behavior.
+    struct Agc2 {

[peah-webrtc, 2017/05/02 11:01:08]

Along the lines of the comments from aleloi@, GainController2 and
gain_controller2 would be more propriate names here.

[AleBzk, 2017/05/02 11:31:55]

Acknowledged.

+      bool enabled = false;
+    } agc2;
   };
 
   // TODO(mgraczyk): Remove once all methods that use ChannelLayout are gone.
   enum ChannelLayout {
     kMono,
     // Left, right.
     kStereo,
     // Mono, keyboard, and mic.
     kMonoAndKeyboard,
     // Left, right, keyboard, and mic.
@@ skipping 794 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_