Added histogram for the output level int the audio processing module

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 389 matching lines @@
   size_t agc_render_queue_element_max_size_ GUARDED_BY(crit_render_)
       GUARDED_BY(crit_capture_) = 0;
   std::vector<int16_t> agc_render_queue_buffer_ GUARDED_BY(crit_render_);
   std::vector<int16_t> agc_capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
   size_t red_render_queue_element_max_size_ GUARDED_BY(crit_render_)
       GUARDED_BY(crit_capture_) = 0;
   std::vector<float> red_render_queue_buffer_ GUARDED_BY(crit_render_);
   std::vector<float> red_capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
-  RmsLevel rms_ GUARDED_BY(crit_capture_);
-  int rms_interval_counter_ GUARDED_BY(crit_capture_) = 0;
+  RmsLevel capture_input_rms_ GUARDED_BY(crit_capture_);
+  RmsLevel capture_output_rms_ GUARDED_BY(crit_capture_);
+  int capture_rms_interval_counter_ GUARDED_BY(crit_capture_) = 0;
 
   // Lock protection not needed.
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       aec_render_signal_queue_;
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
       aecm_render_signal_queue_;
   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_