session-android/jni/webrtc/modules/audio_coding/neteq/time_stretch.h

/*
 *  Copyright (c) 2012 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_CODING_NETEQ_TIME_STRETCH_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_

#include <assert.h>
#include <string.h>  // memset, size_t

#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
#include "webrtc/typedefs.h"

namespace webrtc {

// Forward declarations.
class BackgroundNoise;

// This is the base class for Accelerate and PreemptiveExpand. This class
// cannot be instantiated, but must be used through either of the derived
// classes.
class TimeStretch {
 public:
  enum ReturnCodes {
    kSuccess = 0,
    kSuccessLowEnergy = 1,
    kNoStretch = 2,
    kError = -1
  };

  TimeStretch(int sample_rate_hz, size_t num_channels,
              const BackgroundNoise& background_noise)
      : sample_rate_hz_(sample_rate_hz),
        fs_mult_(sample_rate_hz / 8000),
        num_channels_(static_cast<int>(num_channels)),
        master_channel_(0),  // First channel is master.
        background_noise_(background_noise),
        max_input_value_(0) {
    assert(sample_rate_hz_ == 8000 ||
           sample_rate_hz_ == 16000 ||
           sample_rate_hz_ == 32000 ||
           sample_rate_hz_ == 48000);
    assert(num_channels_ > 0);
    assert(static_cast<int>(master_channel_) < num_channels_);
    memset(auto_correlation_, 0, sizeof(auto_correlation_));
  }

  virtual ~TimeStretch() {}

  // This method performs the processing common to both Accelerate and
  // PreemptiveExpand.
  ReturnCodes Process(const int16_t* input,
                      size_t input_len,
                      AudioMultiVector* output,
                      int16_t* length_change_samples);

 protected:
  // Sets the parameters |best_correlation| and |peak_index| to suitable
  // values when the signal contains no active speech. This method must be
  // implemented by the sub-classes.
  virtual void SetParametersForPassiveSpeech(size_t input_length,
                                             int16_t* best_correlation,
                                             int* peak_index) const = 0;

  // Checks the criteria for performing the time-stretching operation and,
  // if possible, performs the time-stretching. This method must be implemented
  // by the sub-classes.
  virtual ReturnCodes CheckCriteriaAndStretch(
      const int16_t* input, size_t input_length, size_t peak_index,
      int16_t best_correlation, bool active_speech,
      AudioMultiVector* output) const = 0;

  static const int kCorrelationLen = 50;
  static const int kLogCorrelationLen = 6;  // >= log2(kCorrelationLen).
  static const int kMinLag = 10;
  static const int kMaxLag = 60;
  static const int kDownsampledLen = kCorrelationLen + kMaxLag;
  static const int kCorrelationThreshold = 14746;  // 0.9 in Q14.

  const int sample_rate_hz_;
  const int fs_mult_;  // Sample rate multiplier = sample_rate_hz_ / 8000.
  const int num_channels_;
  const size_t master_channel_;
  const BackgroundNoise& background_noise_;
  int16_t max_input_value_;
  int16_t downsampled_input_[kDownsampledLen];
  // Adding 1 to the size of |auto_correlation_| because of how it is used
  // by the peak-detection algorithm.
  int16_t auto_correlation_[kCorrelationLen + 1];

 private:
  // Calculates the auto-correlation of |downsampled_input_| and writes the
  // result to |auto_correlation_|.
  void AutoCorrelation();

  // Performs a simple voice-activity detection based on the input parameters.
  bool SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
                       int peak_index, int scaling) const;

  DISALLOW_COPY_AND_ASSIGN(TimeStretch);
};

}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_