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livetrax/libs/qm-dsp/dsp/tempotracking/DownBeat.cpp

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/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
/*
QM DSP Library
Centre for Digital Music, Queen Mary, University of London.
This file copyright 2008-2009 Matthew Davies and QMUL.
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License as
published by the Free Software Foundation; either version 2 of the
License, or (at your option) any later version. See the file
COPYING included with this distribution for more information.
*/
#include "DownBeat.h"
#include "maths/MathAliases.h"
#include "maths/MathUtilities.h"
#include "maths/KLDivergence.h"
#include "dsp/transforms/FFT.h"
#include <iostream>
#include <cstdlib>
DownBeat::DownBeat(float originalSampleRate,
size_t decimationFactor,
size_t dfIncrement) :
m_bpb(0),
m_rate(originalSampleRate),
m_factor(decimationFactor),
m_increment(dfIncrement),
m_decimator1(0),
m_decimator2(0),
m_buffer(0),
m_decbuf(0),
m_bufsiz(0),
m_buffill(0),
m_beatframesize(0),
m_beatframe(0)
{
// beat frame size is next power of two up from 1.3 seconds at the
// downsampled rate (happens to produce 4096 for 44100 or 48000 at
// 16x decimation, which is our expected normal situation)
m_beatframesize = MathUtilities::nextPowerOfTwo
(int((m_rate / decimationFactor) * 1.3));
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if (m_beatframesize < 2) {
m_beatframesize = 2;
}
// std::cerr << "rate = " << m_rate << ", dec = " << decimationFactor << ", bfs = " << m_beatframesize << std::endl;
m_beatframe = new double[m_beatframesize];
m_fftRealOut = new double[m_beatframesize];
m_fftImagOut = new double[m_beatframesize];
m_fft = new FFTReal(m_beatframesize);
}
DownBeat::~DownBeat()
{
delete m_decimator1;
delete m_decimator2;
if (m_buffer) free(m_buffer);
delete[] m_decbuf;
delete[] m_beatframe;
delete[] m_fftRealOut;
delete[] m_fftImagOut;
delete m_fft;
}
void
DownBeat::setBeatsPerBar(int bpb)
{
m_bpb = bpb;
}
void
DownBeat::makeDecimators()
{
// std::cerr << "m_factor = " << m_factor << std::endl;
if (m_factor < 2) return;
size_t highest = Decimator::getHighestSupportedFactor();
if (m_factor <= highest) {
m_decimator1 = new Decimator(m_increment, m_factor);
// std::cerr << "DownBeat: decimator 1 factor " << m_factor << ", size " << m_increment << std::endl;
return;
}
m_decimator1 = new Decimator(m_increment, highest);
// std::cerr << "DownBeat: decimator 1 factor " << highest << ", size " << m_increment << std::endl;
m_decimator2 = new Decimator(m_increment / highest, m_factor / highest);
// std::cerr << "DownBeat: decimator 2 factor " << m_factor / highest << ", size " << m_increment / highest << std::endl;
m_decbuf = new float[m_increment / highest];
}
void
DownBeat::pushAudioBlock(const float *audio)
{
if (m_buffill + (m_increment / m_factor) > m_bufsiz) {
if (m_bufsiz == 0) m_bufsiz = m_increment * 16;
else m_bufsiz = m_bufsiz * 2;
if (!m_buffer) {
m_buffer = (float *)malloc(m_bufsiz * sizeof(float));
} else {
// std::cerr << "DownBeat::pushAudioBlock: realloc m_buffer to " << m_bufsiz << std::endl;
m_buffer = (float *)realloc(m_buffer, m_bufsiz * sizeof(float));
}
}
if (!m_decimator1 && m_factor > 1) makeDecimators();
// float rmsin = 0, rmsout = 0;
// for (int i = 0; i < m_increment; ++i) {
// rmsin += audio[i] * audio[i];
// }
if (m_decimator2) {
m_decimator1->process(audio, m_decbuf);
m_decimator2->process(m_decbuf, m_buffer + m_buffill);
} else if (m_decimator1) {
m_decimator1->process(audio, m_buffer + m_buffill);
} else {
// just copy across (m_factor is presumably 1)
for (size_t i = 0; i < m_increment; ++i) {
(m_buffer + m_buffill)[i] = audio[i];
}
}
// for (int i = 0; i < m_increment / m_factor; ++i) {
// rmsout += m_buffer[m_buffill + i] * m_buffer[m_buffill + i];
// }
// std::cerr << "pushAudioBlock: rms in " << sqrt(rmsin) << ", out " << sqrt(rmsout) << std::endl;
m_buffill += m_increment / m_factor;
}
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const float *
DownBeat::getBufferedAudio(size_t &length) const
{
length = m_buffill;
return m_buffer;
}
void
DownBeat::resetAudioBuffer()
{
if (m_buffer) free(m_buffer);
m_buffer = 0;
m_buffill = 0;
m_bufsiz = 0;
}
void
DownBeat::findDownBeats(const float *audio,
size_t audioLength,
const d_vec_t &beats,
i_vec_t &downbeats)
{
// FIND DOWNBEATS BY PARTITIONING THE INPUT AUDIO FILE INTO BEAT SEGMENTS
// WHERE THE AUDIO FRAMES ARE DOWNSAMPLED BY A FACTOR OF 16 (fs ~= 2700Hz)
// THEN TAKING THE JENSEN-SHANNON DIVERGENCE BETWEEN BEAT SYNCHRONOUS SPECTRAL FRAMES
// IMPLEMENTATION (MOSTLY) FOLLOWS:
// DAVIES AND PLUMBLEY "A SPECTRAL DIFFERENCE APPROACH TO EXTRACTING DOWNBEATS IN MUSICAL AUDIO"
// EUSIPCO 2006, FLORENCE, ITALY
d_vec_t newspec(m_beatframesize / 2); // magnitude spectrum of current beat
d_vec_t oldspec(m_beatframesize / 2); // magnitude spectrum of previous beat
m_beatsd.clear();
if (audioLength == 0) return;
for (size_t i = 0; i + 1 < beats.size(); ++i) {
// Copy the extents of the current beat from downsampled array
// into beat frame buffer
size_t beatstart = (beats[i] * m_increment) / m_factor;
size_t beatend = (beats[i+1] * m_increment) / m_factor;
if (beatend >= audioLength) beatend = audioLength - 1;
if (beatend < beatstart) beatend = beatstart;
size_t beatlen = beatend - beatstart;
// Also apply a Hanning window to the beat frame buffer, sized
// to the beat extents rather than the frame size. (Because
// the size varies, it's easier to do this by hand than use
// our Window abstraction.)
// std::cerr << "beatlen = " << beatlen << std::endl;
// float rms = 0;
for (size_t j = 0; j < beatlen && j < m_beatframesize; ++j) {
double mul = 0.5 * (1.0 - cos(TWO_PI * (double(j) / double(beatlen))));
m_beatframe[j] = audio[beatstart + j] * mul;
// rms += m_beatframe[j] * m_beatframe[j];
}
// rms = sqrt(rms);
// std::cerr << "beat " << i << ": audio rms " << rms << std::endl;
for (size_t j = beatlen; j < m_beatframesize; ++j) {
m_beatframe[j] = 0.0;
}
// Now FFT beat frame
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m_fft->forward(m_beatframe, m_fftRealOut, m_fftImagOut);
// Calculate magnitudes
for (size_t j = 0; j < m_beatframesize/2; ++j) {
newspec[j] = sqrt(m_fftRealOut[j] * m_fftRealOut[j] +
m_fftImagOut[j] * m_fftImagOut[j]);
}
// Preserve peaks by applying adaptive threshold
MathUtilities::adaptiveThreshold(newspec);
// Calculate JS divergence between new and old spectral frames
if (i > 0) { // otherwise we have no previous frame
m_beatsd.push_back(measureSpecDiff(oldspec, newspec));
// std::cerr << "specdiff: " << m_beatsd[m_beatsd.size()-1] << std::endl;
}
// Copy newspec across to old
for (size_t j = 0; j < m_beatframesize/2; ++j) {
oldspec[j] = newspec[j];
}
}
// We now have all spectral difference measures in specdiff
int timesig = m_bpb;
if (timesig == 0) timesig = 4;
d_vec_t dbcand(timesig); // downbeat candidates
for (int beat = 0; beat < timesig; ++beat) {
dbcand[beat] = 0;
}
// look for beat transition which leads to greatest spectral change
for (int beat = 0; beat < timesig; ++beat) {
int count = 0;
for (int example = beat-1; example < (int)m_beatsd.size(); example += timesig) {
if (example < 0) continue;
dbcand[beat] += (m_beatsd[example]) / timesig;
++count;
}
if (count > 0) dbcand[beat] /= count;
// std::cerr << "dbcand[" << beat << "] = " << dbcand[beat] << std::endl;
}
// first downbeat is beat at index of maximum value of dbcand
int dbind = MathUtilities::getMax(dbcand);
// remaining downbeats are at timesig intervals from the first
for (int i = dbind; i < (int)beats.size(); i += timesig) {
downbeats.push_back(i);
}
}
double
DownBeat::measureSpecDiff(d_vec_t oldspec, d_vec_t newspec)
{
// JENSEN-SHANNON DIVERGENCE BETWEEN SPECTRAL FRAMES
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unsigned int SPECSIZE = 512; // ONLY LOOK AT FIRST 512 SAMPLES OF SPECTRUM.
if (SPECSIZE > oldspec.size()/4) {
SPECSIZE = oldspec.size()/4;
}
double SD = 0.;
double sd1 = 0.;
double sumnew = 0.;
double sumold = 0.;
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for (unsigned int i = 0;i < SPECSIZE;i++)
{
newspec[i] +=EPS;
oldspec[i] +=EPS;
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sumnew+=newspec[i];
sumold+=oldspec[i];
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}
for (unsigned int i = 0;i < SPECSIZE;i++)
{
newspec[i] /= (sumnew);
oldspec[i] /= (sumold);
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// IF ANY SPECTRAL VALUES ARE 0 (SHOULDN'T BE ANY!) SET THEM TO 1
if (newspec[i] == 0)
{
newspec[i] = 1.;
}
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if (oldspec[i] == 0)
{
oldspec[i] = 1.;
}
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// JENSEN-SHANNON CALCULATION
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sd1 = 0.5*oldspec[i] + 0.5*newspec[i];
SD = SD + (-sd1*log(sd1)) + (0.5*(oldspec[i]*log(oldspec[i]))) + (0.5*(newspec[i]*log(newspec[i])));
}
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return SD;
}
void
DownBeat::getBeatSD(vector<double> &beatsd) const
{
for (int i = 0; i < (int)m_beatsd.size(); ++i) beatsd.push_back(m_beatsd[i]);
}