Convert IntelligibilityEnhancer to multi-threaded mode

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/common_audio/lapped_transform.h"
 #include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
+#include "webrtc/modules/audio_processing/processing_component.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);
+  IntelligibilityEnhancer(int sample_rate_hz,
+                          size_t num_render_channels,
+                          size_t num_noise_bins);
 
   // Sets the capture noise magnitude spectrum estimate.
   void SetCaptureNoiseEstimate(std::vector<float> noise);
 
   // 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);
   bool active() const;
 
@@ skipping 22 matching lines @@
   // 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);
 
+  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_;
-  std::unique_ptr<intelligibility::PowerEstimator<float>>
-      noise_power_estimator_;
+  intelligibility::PowerEstimator<float> noise_power_estimator_;
   std::vector<float> filtered_clear_pow_;
   std::vector<float> filtered_noise_pow_;
   std::vector<float> center_freqs_;
   std::vector<std::vector<float>> capture_filter_bank_;
   std::vector<std::vector<float>> render_filter_bank_;
   size_t start_freq_;
 
   std::vector<float> gains_eq_;  // Pre-filter modified gains.
   intelligibility::GainApplier gain_applier_;
 
   std::unique_ptr<LappedTransform> render_mangler_;
 
   VoiceActivityDetector vad_;
   std::vector<int16_t> audio_s16_;
   size_t chunks_since_voice_;
   bool is_speech_;
+
+  std::vector<float> noise_estimation_buffer_;
+  SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>
+      noise_estimation_queue_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_