AGC2 dummy module: fixed gain param, APM integration, audioproc_f adaptation

webrtc/modules/audio_processing/audio_processing_impl.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 348 matching lines @@
     // Only the rate and samples fields of capture_processing_format_ are used
     // because the forward processing number of channels is mutable and is
     // tracked by the capture_audio_.
     StreamConfig capture_processing_format;
     int split_rate;
     int stream_delay_ms;
     bool beamformer_enabled;
     bool intelligibility_enabled;
     bool level_controller_enabled = false;
     bool echo_canceller3_enabled = false;
-    bool gain_controller2_enabled = false;
   } capture_nonlocked_;
 
   struct ApmRenderState {
     ApmRenderState();
     ~ApmRenderState();
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ RTC_GUARDED_BY(crit_render_);
 
   size_t aec_render_queue_element_max_size_ RTC_GUARDED_BY(crit_render_)
@@ skipping 29 matching lines @@
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       agc_render_signal_queue_;
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       red_render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_