| Index: webrtc/modules/audio_processing/audio_processing_impl.h | 
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h | 
| index a44b5a8f41320d10ac6587a30937dd43cb430531..7e89ec281b33eaff19163408ed913548efb54285 100644 | 
| --- a/webrtc/modules/audio_processing/audio_processing_impl.h | 
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h | 
| @@ -23,6 +23,7 @@ namespace webrtc { | 
|  | 
| class AgcManagerDirect; | 
| class AudioBuffer; | 
| +class AudioConverter; | 
|  | 
| template<typename T> | 
| class Beamformer; | 
| @@ -39,6 +40,7 @@ class NoiseSuppressionImpl; | 
| class ProcessingComponent; | 
| class TransientSuppressor; | 
| class VoiceDetectionImpl; | 
| +class IntelligibilityEnhancer; | 
|  | 
| #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP | 
| namespace audioproc { | 
| @@ -89,12 +91,15 @@ class AudioProcessingImpl : public AudioProcessing { | 
| 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; | 
| @@ -124,16 +129,24 @@ class AudioProcessingImpl : public AudioProcessing { | 
| 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, | 
| +                           const float* const* dest); | 
| 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_; | 
| @@ -149,6 +162,7 @@ class AudioProcessingImpl : public AudioProcessing { | 
| 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. | 
| @@ -191,6 +205,9 @@ class AudioProcessingImpl : public AudioProcessing { | 
| 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 | 
|  |