Integrate Intelligibility with APM

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 20 matching lines @@
 class EchoControlMobileImpl;
 class FileWrapper;
 class GainControlImpl;
 class GainControlForNewAgc;
 class HighPassFilterImpl;
 class LevelEstimatorImpl;
 class NoiseSuppressionImpl;
 class ProcessingComponent;
 class TransientSuppressor;
 class VoiceDetectionImpl;
+class IntelligibilityEnhancer;
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
 namespace audioproc {
 
 class Event;
 
 }  // namespace audioproc
 #endif
 
 class AudioRate {
@@ skipping 61 matching lines @@
   bool output_will_be_muted() const override;
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
                     int samples_per_channel,
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
                     int output_sample_rate_hz,
                     ChannelLayout output_layout,
                     float* const* dest) override;
   int AnalyzeReverseStream(AudioFrame* frame) override;
-  int AnalyzeReverseStream(const float* const* data,
+  int AnalyzeReverseStream(float* const* data,
                            int samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
   int set_stream_delay_ms(int delay) override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
   bool stream_key_pressed() const override;
@@ skipping 32 matching lines @@
   int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   bool is_data_processed() const;
   bool output_copy_needed(bool is_data_processed) const;
   bool synthesis_needed(bool is_data_processed) const;
   bool analysis_needed(bool is_data_processed) const;
   void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   EchoCancellationImpl* echo_cancellation_;
   EchoControlMobileImpl* echo_control_mobile_;
   GainControlImpl* gain_control_;
   HighPassFilterImpl* high_pass_filter_;
   LevelEstimatorImpl* level_estimator_;
   NoiseSuppressionImpl* noise_suppression_;
   VoiceDetectionImpl* voice_detection_;
   rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
@@ skipping 38 matching lines @@
   rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
   int agc_startup_min_volume_;
 
   bool transient_suppressor_enabled_;
   rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
   const bool beamformer_enabled_;
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   const bool supports_48kHz_;
+
+  bool intelligibility_enabled_;
+  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_