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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
 
 #include <list>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 
 class AgcManagerDirect;
 class AudioBuffer;
+class AudioConverter;
 
 template<typename T>
 class Beamformer;
 
 class CriticalSectionWrapper;
 class EchoCancellationImpl;
 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 AudioProcessingImpl : public AudioProcessing {
@@ skipping 30 matching lines @@
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
                     int output_sample_rate_hz,
                     ChannelLayout output_layout,
                     float* const* dest) override;
   int ProcessStream(const float* const* src,
                     const StreamConfig& input_config,
                     const StreamConfig& output_config,
                     float* const* dest) override;
   int AnalyzeReverseStream(AudioFrame* frame) override;
+  int ProcessReverseStream(AudioFrame* frame) override;
   int AnalyzeReverseStream(const float* const* data,
                            int samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
-  int AnalyzeReverseStream(const float* const* data,
-                           const StreamConfig& reverse_config) override;
+  int ProcessReverseStream(const float* const* src,
+                           const StreamConfig& reverse_input_config,
+                           const StreamConfig& reverse_output_config,
+                           float* const* dest) 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;
   int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
   int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
   void UpdateHistogramsOnCallEnd() override;
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
   HighPassFilter* high_pass_filter() const override;
   LevelEstimator* level_estimator() const override;
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
 
  protected:
   // Overridden in a mock.
   virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
  private:
   int InitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
   int MaybeInitializeLocked(const ProcessingConfig& config)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  // TODO(ekm): Remove once all clients updated to new interface.
+  int AnalyzeReverseStream(const float* const* src,
+                           const StreamConfig& input_config,
+                           const StreamConfig& output_config);
   int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
-  int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  int ProcessReverseStreamLocked() 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;
+  bool is_rev_processed() const;
+  bool rev_conversion_needed() 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_;
 
   std::list<ProcessingComponent*> component_list_;
   CriticalSectionWrapper* crit_;
   rtc::scoped_ptr<AudioBuffer> render_audio_;
   rtc::scoped_ptr<AudioBuffer> capture_audio_;
+  rtc::scoped_ptr<AudioConverter> render_converter_;
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
   int WriteMessageToDebugFile();
   int WriteInitMessage();
   rtc::scoped_ptr<FileWrapper> debug_file_;
   rtc::scoped_ptr<audioproc::Event> event_msg_;  // Protobuf message.
   std::string event_str_;  // Memory for protobuf serialization.
 #endif
 
@@ skipping 22 matching lines @@
   // Only set through the constructor's Config parameter.
   const bool use_new_agc_;
   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_;
+
+  bool intelligibility_enabled_;
+  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_