Moved the AGC render sample queue into 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 220 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeIntelligibility()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   int InitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeLevelController() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   void EmptyQueuedRenderAudio();
   void AllocateRenderQueue()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
-  void QueueRenderAudio(const AudioBuffer* audio)
+  void QueueRenderAudio(AudioBuffer* audio)
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
 
   // Capture-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   int ProcessCaptureStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
   void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   // Render-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   // TODO(ekm): Remove once all clients updated to new interface.
@@ skipping 112 matching lines @@
     bool level_controller_enabled = false;
   } capture_nonlocked_;
 
   struct ApmRenderState {
     ApmRenderState();
     ~ApmRenderState();
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 
-  size_t float_render_queue_element_max_size_ GUARDED_BY(crit_render_)
+  size_t aec_render_queue_element_max_size_ GUARDED_BY(crit_render_)
       GUARDED_BY(crit_capture_) = 0;
-  std::vector<float> float_render_queue_buffer_ GUARDED_BY(crit_render_);
-  std::vector<float> float_capture_queue_buffer_ GUARDED_BY(crit_capture_);
+  std::vector<float> aec_render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<float> aec_capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
-  size_t int16_render_queue_element_max_size_ GUARDED_BY(crit_render_)
+  size_t aecm_render_queue_element_max_size_ GUARDED_BY(crit_render_)
       GUARDED_BY(crit_capture_) = 0;
-  std::vector<int16_t> int16_render_queue_buffer_ GUARDED_BY(crit_render_);
-  std::vector<int16_t> int16_capture_queue_buffer_ GUARDED_BY(crit_capture_);
+  std::vector<int16_t> aecm_render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<int16_t> aecm_capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  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_);
 
   // Lock protection not needed.
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
-      float_render_signal_queue_;
+      aec_render_signal_queue_;
   std::unique_ptr<
       SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
-      int16_render_signal_queue_;
+      aecm_render_signal_queue_;
+  std::unique_ptr<
+      SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
+      agc_render_signal_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_