blob: 00a141508b684ca1f3bb0f8691b9ff2441cfdb76 [file] [log] [blame]
/*
* 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_(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(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,
bool fast_mode,
AudioMultiVector* output,
size_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,
size_t* 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,
bool fast_mode,
AudioMultiVector* output) const = 0;
static const size_t kCorrelationLen = 50;
static const size_t kLogCorrelationLen = 6; // >= log2(kCorrelationLen).
static const size_t kMinLag = 10;
static const size_t kMaxLag = 60;
static const size_t 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 size_t 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,
size_t peak_index, int scaling) const;
RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_