136 lines
4.2 KiB
C++
136 lines
4.2 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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QM DSP Library
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Centre for Digital Music, Queen Mary, University of London.
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This file copyright 2008-2009 Matthew Davies and QMUL.
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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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*/
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#ifndef DOWNBEAT_H
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#define DOWNBEAT_H
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#include <vector>
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#include "dsp/rateconversion/Decimator.h"
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using std::vector;
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class FFTReal;
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/**
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* This class takes an input audio signal and a sequence of beat
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* locations (calculated e.g. by TempoTrackV2) and estimates which of
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* the beat locations are downbeats (first beat of the bar).
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*
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* The input audio signal is expected to have been downsampled to a
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* very low sampling rate (e.g. 2700Hz). A utility function for
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* downsampling and buffering incoming block-by-block audio is
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* provided.
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*/
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class DownBeat
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{
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public:
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/**
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* Construct a downbeat locator that will operate on audio at the
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* downsampled by the given decimation factor from the given
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* original sample rate, plus beats extracted from the same audio
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* at the given original sample rate with the given frame
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* increment.
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*
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* decimationFactor must be a power of two no greater than 64, and
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* dfIncrement must be a multiple of decimationFactor.
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*/
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DownBeat(float originalSampleRate,
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size_t decimationFactor,
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size_t dfIncrement);
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~DownBeat();
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void setBeatsPerBar(int bpb);
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/**
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* Estimate which beats are down-beats.
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*
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* audio contains the input audio stream after downsampling, and
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* audioLength contains the number of samples in this downsampled
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* stream.
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*
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* beats contains a series of beat positions expressed in
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* multiples of the df increment at the audio's original sample
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* rate, as described to the constructor.
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*
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* The returned downbeat array contains a series of indices to the
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* beats array.
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*/
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void findDownBeats(const float *audio, // downsampled
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size_t audioLength, // after downsampling
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const vector<double> &beats,
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vector<int> &downbeats);
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/**
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* Return the beat spectral difference function. This is
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* calculated during findDownBeats, so this function can only be
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* meaningfully called after that has completed. The returned
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* vector contains one value for each of the beat times passed in
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* to findDownBeats, less one. Each value contains the spectral
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* difference between region prior to the beat's nominal position
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* and the region following it.
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*/
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void getBeatSD(vector<double> &beatsd) const;
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/**
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* For your downsampling convenience: call this function
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* repeatedly with input audio blocks containing dfIncrement
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* samples at the original sample rate, to decimate them to the
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* downsampled rate and buffer them within the DownBeat class.
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*
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* Call getBufferedAudio() to retrieve the results after all
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* blocks have been processed.
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*/
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void pushAudioBlock(const float *audio);
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/**
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* Retrieve the accumulated audio produced by pushAudioBlock calls.
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*/
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const float *getBufferedAudio(size_t &length) const;
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/**
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* Clear any buffered downsampled audio data.
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*/
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void resetAudioBuffer();
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private:
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typedef vector<int> i_vec_t;
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typedef vector<vector<int> > i_mat_t;
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typedef vector<double> d_vec_t;
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typedef vector<vector<double> > d_mat_t;
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void makeDecimators();
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double measureSpecDiff(d_vec_t oldspec, d_vec_t newspec);
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int m_bpb;
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float m_rate;
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size_t m_factor;
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size_t m_increment;
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Decimator *m_decimator1;
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Decimator *m_decimator2;
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float *m_buffer;
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float *m_decbuf;
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size_t m_bufsiz;
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size_t m_buffill;
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size_t m_beatframesize;
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double *m_beatframe;
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FFTReal *m_fft;
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double *m_fftRealOut;
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double *m_fftImagOut;
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d_vec_t m_beatsd;
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};
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#endif
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