Added basic framework for AEC3 in 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 150 matching lines @@
     // Updates the submodule state and returns true if it has changed.
     bool Update(bool low_cut_filter_enabled,
                 bool echo_canceller_enabled,
                 bool mobile_echo_controller_enabled,
                 bool residual_echo_detector_enabled,
                 bool noise_suppressor_enabled,
                 bool intelligibility_enhancer_enabled,
                 bool beamformer_enabled,
                 bool adaptive_gain_controller_enabled,
                 bool level_controller_enabled,
+                bool echo_canceller3_enabled,
                 bool voice_activity_detector_enabled,
                 bool level_estimator_enabled,
                 bool transient_suppressor_enabled);
     bool CaptureMultiBandSubModulesActive() const;
     bool CaptureMultiBandProcessingActive() const;
     bool RenderMultiBandSubModulesActive() const;
     bool RenderMultiBandProcessingActive() const;
 
    private:
     bool low_cut_filter_enabled_ = false;
     bool echo_canceller_enabled_ = false;
     bool mobile_echo_controller_enabled_ = false;
     bool residual_echo_detector_enabled_ = false;
     bool noise_suppressor_enabled_ = false;
     bool intelligibility_enhancer_enabled_ = false;
     bool beamformer_enabled_ = false;
     bool adaptive_gain_controller_enabled_ = false;
     bool level_controller_enabled_ = false;
+    bool echo_canceller3_enabled_ = false;
     bool level_estimator_enabled_ = false;
     bool voice_activity_detector_enabled_ = false;
     bool transient_suppressor_enabled_ = false;
     bool first_update_ = true;
   };
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // State for the debug dump.
   struct ApmDebugDumpThreadState {
     ApmDebugDumpThreadState();
@@ skipping 45 matching lines @@
   void InitializeBeamformer()
       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 InitializeResidualEchoDetector()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeLowCutFilter() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
+  void InitializeEchoCanceller3() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   void EmptyQueuedRenderAudio();
   void AllocateRenderQueue()
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   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_);
@@ skipping 111 matching lines @@
           intelligibility_enabled(intelligibility_enabled) {}
     // 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;
   } 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 aec_render_queue_element_max_size_ GUARDED_BY(crit_render_)
@@ skipping 28 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_