qm-dsp 1.8
DownBeat.cpp
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1/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
2
3/*
4 QM DSP Library
5
6 Centre for Digital Music, Queen Mary, University of London.
7 This file copyright 2008-2009 Matthew Davies and QMUL.
8
9 This program is free software; you can redistribute it and/or
10 modify it under the terms of the GNU General Public License as
11 published by the Free Software Foundation; either version 2 of the
12 License, or (at your option) any later version. See the file
13 COPYING included with this distribution for more information.
14*/
15
16#include "DownBeat.h"
17
18#include "maths/MathAliases.h"
19#include "maths/MathUtilities.h"
20#include "maths/KLDivergence.h"
21#include "dsp/transforms/FFT.h"
22
23#include <iostream>
24#include <cstdlib>
25
26DownBeat::DownBeat(float originalSampleRate,
27 size_t decimationFactor,
28 size_t dfIncrement) :
29 m_bpb(0),
30 m_rate(originalSampleRate),
31 m_factor(decimationFactor),
32 m_increment(dfIncrement),
33 m_decimator1(0),
34 m_decimator2(0),
35 m_buffer(0),
36 m_decbuf(0),
37 m_bufsiz(0),
38 m_buffill(0),
39 m_beatframesize(0),
40 m_beatframe(0)
41{
42 // beat frame size is next power of two up from 1.3 seconds at the
43 // downsampled rate (happens to produce 4096 for 44100 or 48000 at
44 // 16x decimation, which is our expected normal situation)
46 (int((m_rate / decimationFactor) * 1.3));
47 if (m_beatframesize < 2) {
49 }
50// std::cerr << "rate = " << m_rate << ", dec = " << decimationFactor << ", bfs = " << m_beatframesize << std::endl;
51 m_beatframe = new double[m_beatframesize];
52 m_fftRealOut = new double[m_beatframesize];
53 m_fftImagOut = new double[m_beatframesize];
55}
56
58{
59 delete m_decimator1;
60 delete m_decimator2;
61 if (m_buffer) free(m_buffer);
62 delete[] m_decbuf;
63 delete[] m_beatframe;
64 delete[] m_fftRealOut;
65 delete[] m_fftImagOut;
66 delete m_fft;
67}
68
69void
71{
72 m_bpb = bpb;
73}
74
75void
77{
78// std::cerr << "m_factor = " << m_factor << std::endl;
79 if (m_factor < 2) return;
80 size_t highest = Decimator::getHighestSupportedFactor();
81 if (m_factor <= highest) {
83// std::cerr << "DownBeat: decimator 1 factor " << m_factor << ", size " << m_increment << std::endl;
84 return;
85 }
86 m_decimator1 = new Decimator(m_increment, highest);
87// std::cerr << "DownBeat: decimator 1 factor " << highest << ", size " << m_increment << std::endl;
88 m_decimator2 = new Decimator(m_increment / highest, m_factor / highest);
89// std::cerr << "DownBeat: decimator 2 factor " << m_factor / highest << ", size " << m_increment / highest << std::endl;
90 m_decbuf = new float[m_increment / highest];
91}
92
93void
94DownBeat::pushAudioBlock(const float *audio)
95{
97 if (m_bufsiz == 0) m_bufsiz = m_increment * 16;
98 else m_bufsiz = m_bufsiz * 2;
99 if (!m_buffer) {
100 m_buffer = (float *)malloc(m_bufsiz * sizeof(float));
101 } else {
102// std::cerr << "DownBeat::pushAudioBlock: realloc m_buffer to " << m_bufsiz << std::endl;
103 m_buffer = (float *)realloc(m_buffer, m_bufsiz * sizeof(float));
104 }
105 }
106 if (!m_decimator1 && m_factor > 1) makeDecimators();
107// float rmsin = 0, rmsout = 0;
108// for (int i = 0; i < m_increment; ++i) {
109// rmsin += audio[i] * audio[i];
110// }
111 if (m_decimator2) {
114 } else if (m_decimator1) {
116 } else {
117 // just copy across (m_factor is presumably 1)
118 for (size_t i = 0; i < m_increment; ++i) {
119 (m_buffer + m_buffill)[i] = audio[i];
120 }
121 }
122// for (int i = 0; i < m_increment / m_factor; ++i) {
123// rmsout += m_buffer[m_buffill + i] * m_buffer[m_buffill + i];
124// }
125// std::cerr << "pushAudioBlock: rms in " << sqrt(rmsin) << ", out " << sqrt(rmsout) << std::endl;
127}
128
129const float *
130DownBeat::getBufferedAudio(size_t &length) const
131{
132 length = m_buffill;
133 return m_buffer;
134}
135
136void
138{
139 if (m_buffer) free(m_buffer);
140 m_buffer = 0;
141 m_buffill = 0;
142 m_bufsiz = 0;
143}
144
145void
146DownBeat::findDownBeats(const float *audio,
147 size_t audioLength,
148 const d_vec_t &beats,
149 i_vec_t &downbeats)
150{
151 // FIND DOWNBEATS BY PARTITIONING THE INPUT AUDIO FILE INTO BEAT SEGMENTS
152 // WHERE THE AUDIO FRAMES ARE DOWNSAMPLED BY A FACTOR OF 16 (fs ~= 2700Hz)
153 // THEN TAKING THE JENSEN-SHANNON DIVERGENCE BETWEEN BEAT SYNCHRONOUS SPECTRAL FRAMES
154
155 // IMPLEMENTATION (MOSTLY) FOLLOWS:
156 // DAVIES AND PLUMBLEY "A SPECTRAL DIFFERENCE APPROACH TO EXTRACTING DOWNBEATS IN MUSICAL AUDIO"
157 // EUSIPCO 2006, FLORENCE, ITALY
158
159 d_vec_t newspec(m_beatframesize / 2); // magnitude spectrum of current beat
160 d_vec_t oldspec(m_beatframesize / 2); // magnitude spectrum of previous beat
161
162 m_beatsd.clear();
163
164 if (audioLength == 0) return;
165
166 for (size_t i = 0; i + 1 < beats.size(); ++i) {
167
168 // Copy the extents of the current beat from downsampled array
169 // into beat frame buffer
170
171 size_t beatstart = (beats[i] * m_increment) / m_factor;
172 size_t beatend = (beats[i+1] * m_increment) / m_factor;
173 if (beatend >= audioLength) beatend = audioLength - 1;
174 if (beatend < beatstart) beatend = beatstart;
175 size_t beatlen = beatend - beatstart;
176
177 // Also apply a Hanning window to the beat frame buffer, sized
178 // to the beat extents rather than the frame size. (Because
179 // the size varies, it's easier to do this by hand than use
180 // our Window abstraction.)
181
182// std::cerr << "beatlen = " << beatlen << std::endl;
183
184// float rms = 0;
185 for (size_t j = 0; j < beatlen && j < m_beatframesize; ++j) {
186 double mul = 0.5 * (1.0 - cos(TWO_PI * (double(j) / double(beatlen))));
187 m_beatframe[j] = audio[beatstart + j] * mul;
188// rms += m_beatframe[j] * m_beatframe[j];
189 }
190// rms = sqrt(rms);
191// std::cerr << "beat " << i << ": audio rms " << rms << std::endl;
192
193 for (size_t j = beatlen; j < m_beatframesize; ++j) {
194 m_beatframe[j] = 0.0;
195 }
196
197 // Now FFT beat frame
198
200
201 // Calculate magnitudes
202
203 for (size_t j = 0; j < m_beatframesize/2; ++j) {
204 newspec[j] = sqrt(m_fftRealOut[j] * m_fftRealOut[j] +
205 m_fftImagOut[j] * m_fftImagOut[j]);
206 }
207
208 // Preserve peaks by applying adaptive threshold
209
211
212 // Calculate JS divergence between new and old spectral frames
213
214 if (i > 0) { // otherwise we have no previous frame
215 m_beatsd.push_back(measureSpecDiff(oldspec, newspec));
216// std::cerr << "specdiff: " << m_beatsd[m_beatsd.size()-1] << std::endl;
217 }
218
219 // Copy newspec across to old
220
221 for (size_t j = 0; j < m_beatframesize/2; ++j) {
222 oldspec[j] = newspec[j];
223 }
224 }
225
226 // We now have all spectral difference measures in specdiff
227
228 int timesig = m_bpb;
229 if (timesig == 0) timesig = 4;
230
231 d_vec_t dbcand(timesig); // downbeat candidates
232
233 for (int beat = 0; beat < timesig; ++beat) {
234 dbcand[beat] = 0;
235 }
236
237 // look for beat transition which leads to greatest spectral change
238 for (int beat = 0; beat < timesig; ++beat) {
239 int count = 0;
240 for (int example = beat-1; example < (int)m_beatsd.size(); example += timesig) {
241 if (example < 0) continue;
242 dbcand[beat] += (m_beatsd[example]) / timesig;
243 ++count;
244 }
245 if (count > 0) dbcand[beat] /= count;
246// std::cerr << "dbcand[" << beat << "] = " << dbcand[beat] << std::endl;
247 }
248
249 // first downbeat is beat at index of maximum value of dbcand
250 int dbind = MathUtilities::getMax(dbcand);
251
252 // remaining downbeats are at timesig intervals from the first
253 for (int i = dbind; i < (int)beats.size(); i += timesig) {
254 downbeats.push_back(i);
255 }
256}
257
258double
260{
261 // JENSEN-SHANNON DIVERGENCE BETWEEN SPECTRAL FRAMES
262
263 unsigned int SPECSIZE = 512; // ONLY LOOK AT FIRST 512 SAMPLES OF SPECTRUM.
264 if (SPECSIZE > oldspec.size()/4) {
265 SPECSIZE = oldspec.size()/4;
266 }
267 double SD = 0.;
268 double sd1 = 0.;
269
270 double sumnew = 0.;
271 double sumold = 0.;
272
273 for (unsigned int i = 0;i < SPECSIZE;i++)
274 {
275 newspec[i] +=EPS;
276 oldspec[i] +=EPS;
277
278 sumnew+=newspec[i];
279 sumold+=oldspec[i];
280 }
281
282 for (unsigned int i = 0;i < SPECSIZE;i++)
283 {
284 newspec[i] /= (sumnew);
285 oldspec[i] /= (sumold);
286
287 // IF ANY SPECTRAL VALUES ARE 0 (SHOULDN'T BE ANY!) SET THEM TO 1
288 if (newspec[i] == 0)
289 {
290 newspec[i] = 1.;
291 }
292
293 if (oldspec[i] == 0)
294 {
295 oldspec[i] = 1.;
296 }
297
298 // JENSEN-SHANNON CALCULATION
299 sd1 = 0.5*oldspec[i] + 0.5*newspec[i];
300 SD = SD + (-sd1*log(sd1)) + (0.5*(oldspec[i]*log(oldspec[i]))) + (0.5*(newspec[i]*log(newspec[i])));
301 }
302
303 return SD;
304}
305
306void
307DownBeat::getBeatSD(vector<double> &beatsd) const
308{
309 for (int i = 0; i < (int)m_beatsd.size(); ++i) beatsd.push_back(m_beatsd[i]);
310}
311
#define TWO_PI
Definition MathAliases.h:30
#define EPS
Decimator carries out a fast downsample by a power-of-two factor.
Definition Decimator.h:25
static int getHighestSupportedFactor()
Definition Decimator.h:57
void process(const double *src, double *dst)
Process inLength samples (as supplied to constructor) from src and write inLength / decFactor samples...
double * m_beatframe
Definition DownBeat.h:129
double * m_fftRealOut
Definition DownBeat.h:131
FFTReal * m_fft
Definition DownBeat.h:130
int m_bpb
Definition DownBeat.h:118
DownBeat(float originalSampleRate, size_t decimationFactor, size_t dfIncrement)
Construct a downbeat locator that will operate on audio at the downsampled by the given decimation fa...
Definition DownBeat.cpp:26
float m_rate
Definition DownBeat.h:119
vector< double > d_vec_t
Definition DownBeat.h:112
float * m_decbuf
Definition DownBeat.h:125
Decimator * m_decimator1
Definition DownBeat.h:122
const float * getBufferedAudio(size_t &length) const
Retrieve the accumulated audio produced by pushAudioBlock calls.
Definition DownBeat.cpp:130
size_t m_increment
Definition DownBeat.h:121
double * m_fftImagOut
Definition DownBeat.h:132
size_t m_buffill
Definition DownBeat.h:127
float * m_buffer
Definition DownBeat.h:124
void pushAudioBlock(const float *audio)
For your downsampling convenience: call this function repeatedly with input audio blocks containing d...
Definition DownBeat.cpp:94
void getBeatSD(vector< double > &beatsd) const
Return the beat spectral difference function.
Definition DownBeat.cpp:307
Decimator * m_decimator2
Definition DownBeat.h:123
void makeDecimators()
Definition DownBeat.cpp:76
size_t m_beatframesize
Definition DownBeat.h:128
vector< int > i_vec_t
Definition DownBeat.h:110
void findDownBeats(const float *audio, size_t audioLength, const vector< double > &beats, vector< int > &downbeats)
Estimate which beats are down-beats.
Definition DownBeat.cpp:146
size_t m_factor
Definition DownBeat.h:120
size_t m_bufsiz
Definition DownBeat.h:126
void setBeatsPerBar(int bpb)
Definition DownBeat.cpp:70
d_vec_t m_beatsd
Definition DownBeat.h:133
double measureSpecDiff(d_vec_t oldspec, d_vec_t newspec)
Definition DownBeat.cpp:259
void resetAudioBuffer()
Clear any buffered downsampled audio data.
Definition DownBeat.cpp:137
Definition FFT.h:47
void forward(const double *realIn, double *realOut, double *imagOut)
Carry out a forward real-to-complex transform of size nsamples, where nsamples is the value provided ...
Definition FFT.cpp:184
static int nextPowerOfTwo(int x)
Return the next higher integer power of two from x, e.g.
static void adaptiveThreshold(std::vector< double > &data)
Threshold the input/output vector data against a moving-mean average filter.
static int getMax(double *data, unsigned int length, double *max=0)