Add information about microphone gain changes to AEC3

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 @@
     bool key_pressed;
     bool transient_suppressor_enabled;
     std::vector<Point> array_geometry;
     SphericalPointf target_direction;
     std::unique_ptr<AudioBuffer> capture_audio;
     // Only the rate and samples fields of capture_processing_format_ are used
     // because the capture processing number of channels is mutable and is
     // tracked by the capture_audio_.
     StreamConfig capture_processing_format;
     int split_rate;
+    int previous_agc_level;
+    bool echo_path_gain_change;
   } capture_ GUARDED_BY(crit_capture_);
 
   struct ApmCaptureNonLockedState {
     ApmCaptureNonLockedState(bool beamformer_enabled,
                              bool intelligibility_enabled)
         : capture_processing_format(kSampleRate16kHz),
           split_rate(kSampleRate16kHz),
           stream_delay_ms(0),
           beamformer_enabled(beamformer_enabled),
           intelligibility_enabled(intelligibility_enabled) {}
@@ skipping 49 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_