Compensate for the IntelligibilityEnhancer processing delay in high bands

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h

 /*
  *  Copyright (c) 2014 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_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
 
 #include <complex>
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/swap_queue.h"
+#include "webrtc/common_audio/audio_ring_buffer.h"
+#include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/common_audio/lapped_transform.h"
-#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
 #include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
 
 namespace webrtc {
 
 // Speech intelligibility enhancement module. Reads render and capture
 // audio streams and modifies the render stream with a set of gains per
 // frequency bin to enhance speech against the noise background.
 // Details of the model and algorithm can be found in the original paper:
 // http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788
 class IntelligibilityEnhancer : public LappedTransform::Callback {
  public:
   IntelligibilityEnhancer(int sample_rate_hz,
                           size_t num_render_channels,
+                          size_t num_bands,
                           size_t num_noise_bins);
 
   ~IntelligibilityEnhancer() override;
 
   // Sets the capture noise magnitude spectrum estimate.
   void SetCaptureNoiseEstimate(std::vector<float> noise, float gain);
 
   // Reads chunk of speech in time domain and updates with modified signal.
-  void ProcessRenderAudio(float* const* audio,
-                          int sample_rate_hz,
-                          size_t num_channels);
+  void ProcessRenderAudio(AudioBuffer* audio);
   bool active() const;
 
  protected:
   // All in frequency domain, receives input |in_block|, applies
   // intelligibility enhancement, and writes result to |out_block|.
   void ProcessAudioBlock(const std::complex<float>* const* in_block,
                          size_t in_channels,
                          size_t frames,
                          size_t out_channels,
                          std::complex<float>* const* out_block) override;
 
  private:
+  FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestRenderUpdate);
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation);
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains);
   FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest,
                            TestNoiseGainHasExpectedResult);
 
   // Updates the SNR estimation and enables or disables this component using a
   // hysteresis.
   void SnrBasedEffectActivation();
 
   // Bisection search for optimal |lambda|.
   void SolveForLambda(float power_target);
 
   // Transforms freq gains to ERB gains.
   void UpdateErbGains();
 
   // Returns number of ERB filters.
   static size_t GetBankSize(int sample_rate, size_t erb_resolution);
 
   // Initializes ERB filterbank.
   std::vector<std::vector<float>> CreateErbBank(size_t num_freqs);
 
   // Analytically solves quadratic for optimal gains given |lambda|.
   // Negative gains are set to 0. Stores the results in |sols|.
   void SolveForGainsGivenLambda(float lambda, size_t start_freq, float* sols);
 
   // Returns true if the audio is speech.
   bool IsSpeech(const float* audio);
 
+  // Delays the high bands to compensate for the processing delay in the low
+  // band.
+  void DelayHighBands(AudioBuffer* audio);
+
   static const size_t kMaxNumNoiseEstimatesToBuffer = 5;
 
   const size_t freqs_;         // Num frequencies in frequency domain.
   const size_t num_noise_bins_;
   const size_t chunk_length_;  // Chunk size in samples.
   const size_t bank_size_;     // Num ERB filters.
   const int sample_rate_hz_;
   const size_t num_render_channels_;
 
   intelligibility::PowerEstimator<std::complex<float>> clear_power_estimator_;
@@ skipping 16 matching lines @@
   bool is_speech_;
   float snr_;
   bool is_active_;
 
   unsigned long int num_chunks_;
   unsigned long int num_active_chunks_;
 
   std::vector<float> noise_estimation_buffer_;
   SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>
       noise_estimation_queue_;
+
+  std::vector<std::unique_ptr<AudioRingBuffer>> high_bands_buffers_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_