| Index: webrtc/modules/audio_processing/audio_processing_impl.h
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| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
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| index a44b5a8f41320d10ac6587a30937dd43cb430531..7e89ec281b33eaff19163408ed913548efb54285 100644
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| --- a/webrtc/modules/audio_processing/audio_processing_impl.h
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| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h
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| @@ -23,6 +23,7 @@ namespace webrtc {
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|  
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|  class AgcManagerDirect;
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|  class AudioBuffer;
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| +class AudioConverter;
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|  
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|  template<typename T>
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|  class Beamformer;
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| @@ -39,6 +40,7 @@ class NoiseSuppressionImpl;
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|  class ProcessingComponent;
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|  class TransientSuppressor;
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|  class VoiceDetectionImpl;
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| +class IntelligibilityEnhancer;
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|  
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|  #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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|  namespace audioproc {
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| @@ -89,12 +91,15 @@ class AudioProcessingImpl : public AudioProcessing {
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|                      const StreamConfig& output_config,
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|                      float* const* dest) override;
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|    int AnalyzeReverseStream(AudioFrame* frame) override;
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| +  int ProcessReverseStream(AudioFrame* frame) override;
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|    int AnalyzeReverseStream(const float* const* data,
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|                             int samples_per_channel,
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|                             int sample_rate_hz,
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|                             ChannelLayout layout) override;
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| -  int AnalyzeReverseStream(const float* const* data,
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| -                           const StreamConfig& reverse_config) override;
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| +  int ProcessReverseStream(const float* const* src,
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| +                           const StreamConfig& reverse_input_config,
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| +                           const StreamConfig& reverse_output_config,
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| +                           float* const* dest) override;
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|    int set_stream_delay_ms(int delay) override;
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|    int stream_delay_ms() const override;
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|    bool was_stream_delay_set() const override;
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| @@ -124,16 +129,24 @@ class AudioProcessingImpl : public AudioProcessing {
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|        EXCLUSIVE_LOCKS_REQUIRED(crit_);
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|    int MaybeInitializeLocked(const ProcessingConfig& config)
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|        EXCLUSIVE_LOCKS_REQUIRED(crit_);
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| +  // TODO(ekm): Remove once all clients updated to new interface.
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| +  int AnalyzeReverseStream(const float* const* src,
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| +                           const StreamConfig& input_config,
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| +                           const StreamConfig& output_config,
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| +                           const float* const* dest);
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|    int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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| -  int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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| +  int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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|  
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|    bool is_data_processed() const;
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|    bool output_copy_needed(bool is_data_processed) const;
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|    bool synthesis_needed(bool is_data_processed) const;
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|    bool analysis_needed(bool is_data_processed) const;
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| +  bool is_rev_processed() const;
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| +  bool rev_conversion_needed() const;
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|    void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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|    void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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|    void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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| +  void InitializeIntelligibility() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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|    void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
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|  
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|    EchoCancellationImpl* echo_cancellation_;
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| @@ -149,6 +162,7 @@ class AudioProcessingImpl : public AudioProcessing {
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|    CriticalSectionWrapper* crit_;
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|    rtc::scoped_ptr<AudioBuffer> render_audio_;
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|    rtc::scoped_ptr<AudioBuffer> capture_audio_;
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| +  rtc::scoped_ptr<AudioConverter> render_converter_;
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|  #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
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|    // TODO(andrew): make this more graceful. Ideally we would split this stuff
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|    // out into a separate class with an "enabled" and "disabled" implementation.
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| @@ -191,6 +205,9 @@ class AudioProcessingImpl : public AudioProcessing {
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|    const bool beamformer_enabled_;
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|    rtc::scoped_ptr<Beamformer<float>> beamformer_;
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|    const std::vector<Point> array_geometry_;
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| +
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| +  bool intelligibility_enabled_;
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| +  rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
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|  };
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|  
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|  }  // namespace webrtc
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| 
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