Initial SIE commit: migrating existing code

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 "webrtc/common_audio/lapped_transform.h"
+#include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+struct WebRtcVadInst;
+typedef struct WebRtcVadInst VadInst;
+
+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.
+class IntelligibilityEnhancer {
+ public:
+  // Construct a new instance with the given filter bank resolution,
+  // sampling rate, number of channels and analysis rates.
+  // |analysis_rate| sets the number of input blocks (containing speech!)
+  // to elapse before a new gain computation is made. |variance_rate| specifies
+  // the number of gain recomputations after which the variances are reset.
+  // |cv_*| are parameters for the VarianceArray constructor for the
+  // lear speech stream.
+  // TODO(bercic): the |cv_*|, |*_rate| and |gain_limit| parameters should
+  // probably go away once fine tuning is done. They override the internal
+  // constants in the class (kGainChangeLimit, kAnalyzeRate, kVarianceRate).
+  IntelligibilityEnhancer(int erb_resolution, int sample_rate_hz, int channels,
+                          int cv_type, float cv_alpha, int cv_win,
+                          int analysis_rate, int variance_rate,
+                          float gain_limit);
+  ~IntelligibilityEnhancer();
+
+  void ProcessRenderAudio(float* const* audio);
+  void ProcessCaptureAudio(float* const* audio);
+
+ private:
+  enum AudioSource {
+    kRenderStream = 0,
+    kCaptureStream,
+  };
+
+  class TransformCallback : public LappedTransform::Callback {
+   public:
+    TransformCallback(IntelligibilityEnhancer* parent, AudioSource source);
+    virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
+                                   int in_channels, int frames,
+                                   int out_channels,
+                                   std::complex<float>* const* out_block);
+
+   private:
+    IntelligibilityEnhancer* parent_;
+    AudioSource source_;
+  };
+  friend class TransformCallback;
+
+  void DispatchAudio(AudioSource source, const std::complex<float>* in_block,
+                     std::complex<float>* out_block);
+  void ProcessClearBlock(const std::complex<float>* in_block,
+                         std::complex<float>* out_block);
+  void AnalyzeClearBlock(float power_target);
+  void ProcessNoiseBlock(const std::complex<float>* in_block,
+                         std::complex<float>* out_block);
+
+  static int GetBankSize(int sample_rate, int erb_resolution);
+  void CreateErbBank();
+  void SolveEquation14(float lambda, int start_freq, float* sols);
+  void FilterVariance(const float* var, float* result);
+  static float DotProduct(const float* a, const float* b, int length);
+
+  static const int kErbResolution;
+  static const int kWindowSizeMs;
+  static const int kChunkSizeMs;
+  static const int kAnalyzeRate;
+  static const int kVarianceRate;
+  static const float kClipFreq;
+  static const float kConfigRho;
+  static const float kKbdAlpha;
+  static const float kGainChangeLimit;
+
+  const int freqs_;
+  const int window_size_;  // window size in samples; also the block size
+  const int chunk_length_;  // chunk size in samples
+  const int bank_size_;
+  const int sample_rate_hz_;
+  const int erb_resolution_;
+  const int channels_;
+  const int analysis_rate_;
+  const int variance_rate_;
+
+  intelligibility::VarianceArray clear_variance_;
+  intelligibility::VarianceArray noise_variance_;
+  scoped_ptr<float[]> filtered_clear_var_;
+  scoped_ptr<float[]> filtered_noise_var_;
+  float** filter_bank_;
+  scoped_ptr<float[]> center_freqs_;
+  int start_freq_;
+  scoped_ptr<float[]> rho_;
+  scoped_ptr<float[]> gains_eq_;
+  intelligibility::GainApplier gain_applier_;
+
+  // Destination buffer used to reassemble blocked chunks before overwriting
+  // the original input array with modifications.
+  float** temp_out_buffer_;
+  scoped_ptr<float*[]> input_audio_;
+  scoped_ptr<float[]> kbd_window_;
+  TransformCallback render_callback_;
+  TransformCallback capture_callback_;
+  scoped_ptr<LappedTransform> render_mangler_;
+  scoped_ptr<LappedTransform> capture_mangler_;
+  int block_count_;
+  int analysis_step_;
+
+  // TODO(bercic): Quick stopgap measure for voice detection in the clear
+  // and noise streams.
+  VadInst* vad_high_;
+  VadInst* vad_low_;
+  scoped_ptr<int16_t[]> vad_tmp_buffer_;
+  bool has_voice_low_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_
+