mirror of
https://github.com/dolphin-emu/dolphin.git
synced 2024-12-30 23:00:51 +01:00
372 lines
12 KiB
C++
372 lines
12 KiB
C++
////////////////////////////////////////////////////////////////////////////////
|
|
///
|
|
/// Beats-per-minute (BPM) detection routine.
|
|
///
|
|
/// The beat detection algorithm works as follows:
|
|
/// - Use function 'inputSamples' to input a chunks of samples to the class for
|
|
/// analysis. It's a good idea to enter a large sound file or stream in smallish
|
|
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
|
|
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
|
|
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
|
|
/// Simple averaging is used for anti-alias filtering because the resulting signal
|
|
/// quality isn't of that high importance.
|
|
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
|
|
/// taking absolute value that's smoothed by sliding average. Signal levels that
|
|
/// are below a couple of times the general RMS amplitude level are cut away to
|
|
/// leave only notable peaks there.
|
|
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
|
|
/// autocorrelation function of the enveloped signal.
|
|
/// - After whole sound data file has been analyzed as above, the bpm level is
|
|
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
|
|
/// function, calculates it's precise location and converts this reading to bpm's.
|
|
///
|
|
/// Author : Copyright (c) Olli Parviainen
|
|
/// Author e-mail : oparviai 'at' iki.fi
|
|
/// SoundTouch WWW: http://www.surina.net/soundtouch
|
|
///
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// Last changed : $Date: 2015-02-21 23:24:29 +0200 (Sat, 21 Feb 2015) $
|
|
// File revision : $Revision: 4 $
|
|
//
|
|
// $Id: BPMDetect.cpp 202 2015-02-21 21:24:29Z oparviai $
|
|
//
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
//
|
|
// License :
|
|
//
|
|
// SoundTouch audio processing library
|
|
// Copyright (c) Olli Parviainen
|
|
//
|
|
// This library is free software; you can redistribute it and/or
|
|
// modify it under the terms of the GNU Lesser General Public
|
|
// License as published by the Free Software Foundation; either
|
|
// version 2.1 of the License, or (at your option) any later version.
|
|
//
|
|
// This library is distributed in the hope that it will be useful,
|
|
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
// Lesser General Public License for more details.
|
|
//
|
|
// You should have received a copy of the GNU Lesser General Public
|
|
// License along with this library; if not, write to the Free Software
|
|
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
|
//
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
|
|
#include <math.h>
|
|
#include <assert.h>
|
|
#include <string.h>
|
|
#include <stdio.h>
|
|
#include "FIFOSampleBuffer.h"
|
|
#include "PeakFinder.h"
|
|
#include "BPMDetect.h"
|
|
|
|
using namespace soundtouch;
|
|
|
|
#define INPUT_BLOCK_SAMPLES 2048
|
|
#define DECIMATED_BLOCK_SAMPLES 256
|
|
|
|
/// decay constant for calculating RMS volume sliding average approximation
|
|
/// (time constant is about 10 sec)
|
|
const float avgdecay = 0.99986f;
|
|
|
|
/// Normalization coefficient for calculating RMS sliding average approximation.
|
|
const float avgnorm = (1 - avgdecay);
|
|
|
|
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
|
|
// Enable following define to create bpm analysis file:
|
|
|
|
// #define _CREATE_BPM_DEBUG_FILE
|
|
|
|
#ifdef _CREATE_BPM_DEBUG_FILE
|
|
|
|
#define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt"
|
|
|
|
static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
|
|
{
|
|
FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
|
|
int i;
|
|
|
|
if (fptr)
|
|
{
|
|
printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
|
|
for (i = minpos; i < maxpos; i ++)
|
|
{
|
|
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
|
|
}
|
|
fclose(fptr);
|
|
}
|
|
}
|
|
#else
|
|
#define _SaveDebugData(a,b,c,d)
|
|
#endif
|
|
|
|
////////////////////////////////////////////////////////////////////////////////
|
|
|
|
|
|
BPMDetect::BPMDetect(int numChannels, int aSampleRate)
|
|
{
|
|
this->sampleRate = aSampleRate;
|
|
this->channels = numChannels;
|
|
|
|
decimateSum = 0;
|
|
decimateCount = 0;
|
|
|
|
envelopeAccu = 0;
|
|
|
|
// Initialize RMS volume accumulator to RMS level of 1500 (out of 32768) that's
|
|
// safe initial RMS signal level value for song data. This value is then adapted
|
|
// to the actual level during processing.
|
|
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
|
// integer samples
|
|
RMSVolumeAccu = (1500 * 1500) / avgnorm;
|
|
#else
|
|
// float samples, scaled to range [-1..+1[
|
|
RMSVolumeAccu = (0.045f * 0.045f) / avgnorm;
|
|
#endif
|
|
|
|
// choose decimation factor so that result is approx. 1000 Hz
|
|
decimateBy = sampleRate / 1000;
|
|
assert(decimateBy > 0);
|
|
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
|
|
|
|
// Calculate window length & starting item according to desired min & max bpms
|
|
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
|
|
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
|
|
|
|
assert(windowLen > windowStart);
|
|
|
|
// allocate new working objects
|
|
xcorr = new float[windowLen];
|
|
memset(xcorr, 0, windowLen * sizeof(float));
|
|
|
|
// allocate processing buffer
|
|
buffer = new FIFOSampleBuffer();
|
|
// we do processing in mono mode
|
|
buffer->setChannels(1);
|
|
buffer->clear();
|
|
}
|
|
|
|
|
|
|
|
BPMDetect::~BPMDetect()
|
|
{
|
|
delete[] xcorr;
|
|
delete buffer;
|
|
}
|
|
|
|
|
|
|
|
/// convert to mono, low-pass filter & decimate to about 500 Hz.
|
|
/// return number of outputted samples.
|
|
///
|
|
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
|
|
/// the amount of data needed to be processed as calculating autocorrelation
|
|
/// function is a very-very heavy operation.
|
|
///
|
|
/// Anti-alias filtering is done simply by averaging the samples. This is really a
|
|
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
|
|
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
|
|
/// narrow band)
|
|
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
|
|
{
|
|
int count, outcount;
|
|
LONG_SAMPLETYPE out;
|
|
|
|
assert(channels > 0);
|
|
assert(decimateBy > 0);
|
|
outcount = 0;
|
|
for (count = 0; count < numsamples; count ++)
|
|
{
|
|
int j;
|
|
|
|
// convert to mono and accumulate
|
|
for (j = 0; j < channels; j ++)
|
|
{
|
|
decimateSum += src[j];
|
|
}
|
|
src += j;
|
|
|
|
decimateCount ++;
|
|
if (decimateCount >= decimateBy)
|
|
{
|
|
// Store every Nth sample only
|
|
out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
|
|
decimateSum = 0;
|
|
decimateCount = 0;
|
|
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
|
// check ranges for sure (shouldn't actually be necessary)
|
|
if (out > 32767)
|
|
{
|
|
out = 32767;
|
|
}
|
|
else if (out < -32768)
|
|
{
|
|
out = -32768;
|
|
}
|
|
#endif // SOUNDTOUCH_INTEGER_SAMPLES
|
|
dest[outcount] = (SAMPLETYPE)out;
|
|
outcount ++;
|
|
}
|
|
}
|
|
return outcount;
|
|
}
|
|
|
|
|
|
|
|
// Calculates autocorrelation function of the sample history buffer
|
|
void BPMDetect::updateXCorr(int process_samples)
|
|
{
|
|
int offs;
|
|
SAMPLETYPE *pBuffer;
|
|
|
|
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
|
|
|
|
pBuffer = buffer->ptrBegin();
|
|
#pragma omp parallel for
|
|
for (offs = windowStart; offs < windowLen; offs ++)
|
|
{
|
|
LONG_SAMPLETYPE sum;
|
|
int i;
|
|
|
|
sum = 0;
|
|
for (i = 0; i < process_samples; i ++)
|
|
{
|
|
sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
|
|
}
|
|
// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
|
|
// if it's desired that the system adapts automatically to
|
|
// various bpms, e.g. in processing continouos music stream.
|
|
// The 'xcorr_decay' should be a value that's smaller than but
|
|
// close to one, and should also depend on 'process_samples' value.
|
|
|
|
xcorr[offs] += (float)sum;
|
|
}
|
|
}
|
|
|
|
|
|
// Calculates envelope of the sample data
|
|
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
|
|
{
|
|
const static double decay = 0.7f; // decay constant for smoothing the envelope
|
|
const static double norm = (1 - decay);
|
|
|
|
int i;
|
|
LONG_SAMPLETYPE out;
|
|
double val;
|
|
|
|
for (i = 0; i < numsamples; i ++)
|
|
{
|
|
// calc average RMS volume
|
|
RMSVolumeAccu *= avgdecay;
|
|
val = (float)fabs((float)samples[i]);
|
|
RMSVolumeAccu += val * val;
|
|
|
|
// cut amplitudes that are below cutoff ~2 times RMS volume
|
|
// (we're interested in peak values, not the silent moments)
|
|
if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
|
|
{
|
|
val = 0;
|
|
}
|
|
|
|
// smooth amplitude envelope
|
|
envelopeAccu *= decay;
|
|
envelopeAccu += val;
|
|
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
|
|
|
|
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
|
// cut peaks (shouldn't be necessary though)
|
|
if (out > 32767) out = 32767;
|
|
#endif // SOUNDTOUCH_INTEGER_SAMPLES
|
|
samples[i] = (SAMPLETYPE)out;
|
|
}
|
|
}
|
|
|
|
|
|
|
|
void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
|
|
{
|
|
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
|
|
|
|
// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
|
|
while (numSamples > 0)
|
|
{
|
|
int block;
|
|
int decSamples;
|
|
|
|
block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
|
|
|
|
// decimate. note that converts to mono at the same time
|
|
decSamples = decimate(decimated, samples, block);
|
|
samples += block * channels;
|
|
numSamples -= block;
|
|
|
|
// envelope new samples and add them to buffer
|
|
calcEnvelope(decimated, decSamples);
|
|
buffer->putSamples(decimated, decSamples);
|
|
}
|
|
|
|
// when the buffer has enought samples for processing...
|
|
if ((int)buffer->numSamples() > windowLen)
|
|
{
|
|
int processLength;
|
|
|
|
// how many samples are processed
|
|
processLength = (int)buffer->numSamples() - windowLen;
|
|
|
|
// ... calculate autocorrelations for oldest samples...
|
|
updateXCorr(processLength);
|
|
// ... and remove them from the buffer
|
|
buffer->receiveSamples(processLength);
|
|
}
|
|
}
|
|
|
|
|
|
|
|
void BPMDetect::removeBias()
|
|
{
|
|
int i;
|
|
float minval = 1e12f; // arbitrary large number
|
|
|
|
for (i = windowStart; i < windowLen; i ++)
|
|
{
|
|
if (xcorr[i] < minval)
|
|
{
|
|
minval = xcorr[i];
|
|
}
|
|
}
|
|
|
|
for (i = windowStart; i < windowLen; i ++)
|
|
{
|
|
xcorr[i] -= minval;
|
|
}
|
|
}
|
|
|
|
|
|
float BPMDetect::getBpm()
|
|
{
|
|
double peakPos;
|
|
double coeff;
|
|
PeakFinder peakFinder;
|
|
|
|
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
|
|
|
|
// save bpm debug analysis data if debug data enabled
|
|
_SaveDebugData(xcorr, windowStart, windowLen, coeff);
|
|
|
|
// remove bias from xcorr data
|
|
removeBias();
|
|
|
|
// find peak position
|
|
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
|
|
|
|
assert(decimateBy != 0);
|
|
if (peakPos < 1e-9) return 0.0; // detection failed.
|
|
|
|
// calculate BPM
|
|
return (float) (coeff / peakPos);
|
|
}
|