135 lines
4.4 KiB
C++
135 lines
4.4 KiB
C++
/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
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/*
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Rubber Band Library
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An audio time-stretching and pitch-shifting library.
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Copyright 2007-2022 Particular Programs Ltd.
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This program is free software; you can redistribute it and/or
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modify it under the terms of the GNU General Public License as
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published by the Free Software Foundation; either version 2 of the
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License, or (at your option) any later version. See the file
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COPYING included with this distribution for more information.
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Alternatively, if you have a valid commercial licence for the
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Rubber Band Library obtained by agreement with the copyright
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holders, you may redistribute and/or modify it under the terms
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described in that licence.
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If you wish to distribute code using the Rubber Band Library
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under terms other than those of the GNU General Public License,
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you must obtain a valid commercial licence before doing so.
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*/
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#ifndef RUBBERBAND_AUDIO_CURVE_CALCULATOR_H
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#define RUBBERBAND_AUDIO_CURVE_CALCULATOR_H
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#include <sys/types.h>
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#include "../system/sysutils.h"
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namespace RubberBand
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{
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/**
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* AudioCurveCalculator turns a sequence of audio "columns" --
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* short-time spectrum magnitude blocks -- into a sequence of numbers
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* representing some quality of the input such as power or likelihood
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* of an onset occurring.
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*
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* These are typically low-level building-blocks: AudioCurveCalculator
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* is a simple causal interface in which each input column corresponds
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* to exactly one output value which is returned immediately. They
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* have far less power (because of the causal interface and
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* magnitude-only input) and flexibility (because of the limited
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* return types) than for example the Vamp plugin interface.
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*
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* AudioCurveCalculator implementations typically remember the history
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* of their processing data, and the caller must call reset() before
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* resynchronising to an unrelated piece of input audio.
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*/
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class AudioCurveCalculator
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{
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public:
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struct Parameters {
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Parameters(int _sampleRate, int _fftSize) :
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sampleRate(_sampleRate),
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fftSize(_fftSize)
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{ }
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int sampleRate;
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int fftSize;
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};
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AudioCurveCalculator(Parameters parameters);
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virtual ~AudioCurveCalculator();
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int getSampleRate() const { return m_sampleRate; }
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int getFftSize() const { return m_fftSize; }
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virtual void setSampleRate(int newRate);
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virtual void setFftSize(int newSize);
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Parameters getParameters() const {
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return Parameters(m_sampleRate, m_fftSize);
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}
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void setParameters(Parameters p) {
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setSampleRate(p.sampleRate);
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setFftSize(p.fftSize);
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}
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// You may not mix calls to the various process functions on a
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// given instance
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/**
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* Process the given magnitude spectrum block and return the curve
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* value for it. The mag input contains (fftSize/2 + 1) values
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* corresponding to the magnitudes of the complex FFT output bins
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* for a windowed input of size fftSize. The hop (expressed in
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* time-domain audio samples) from the previous to the current
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* input block is given by increment.
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*/
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virtual float processFloat(const float *R__ mag, int increment) = 0;
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/**
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* Process the given magnitude spectrum block and return the curve
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* value for it. The mag input contains (fftSize/2 + 1) values
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* corresponding to the magnitudes of the complex FFT output bins
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* for a windowed input of size fftSize. The hop (expressed in
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* time-domain audio samples) from the previous to the current
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* input block is given by increment.
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*/
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virtual double processDouble(const double *R__ mag, int increment) = 0;
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/**
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* Obtain a confidence for the curve value (if applicable). A
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* value of 1.0 indicates perfect confidence in the curve
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* calculation, 0.0 indicates none.
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*/
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virtual double getConfidence() const { return 1.0; }
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/**
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* Reset the calculator, forgetting the history of the audio input
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* so far.
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*/
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virtual void reset() = 0;
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/**
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* If the output of this calculator has a known unit, return it as
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* text. For example, "Hz" or "V".
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*/
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virtual const char *getUnit() const { return ""; }
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protected:
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int m_sampleRate;
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int m_fftSize;
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int m_lastPerceivedBin;
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void recalculateLastPerceivedBin();
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};
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}
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#endif
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