mirror of
https://github.com/dolphin-emu/dolphin.git
synced 2025-07-23 14:19:46 -06:00
Update SoundTouch to 2.3.2 commit 1eda9c0b01039f29d230a46cda9f2290bbd1f62b
This commit is contained in:
get
402
Externals/soundtouch/BPMDetect.cpp
vendored
402
Externals/soundtouch/BPMDetect.cpp
vendored
@ -26,13 +26,6 @@
|
||||
///
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// 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
|
||||
@ -54,45 +47,62 @@
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#define _USE_MATH_DEFINES
|
||||
|
||||
#include <math.h>
|
||||
#include <assert.h>
|
||||
#include <string.h>
|
||||
#include <stdio.h>
|
||||
#include <cfloat>
|
||||
#include "FIFOSampleBuffer.h"
|
||||
#include "PeakFinder.h"
|
||||
#include "BPMDetect.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
#define INPUT_BLOCK_SAMPLES 2048
|
||||
#define DECIMATED_BLOCK_SAMPLES 256
|
||||
// algorithm input sample block size
|
||||
static const int INPUT_BLOCK_SIZE = 2048;
|
||||
|
||||
/// decay constant for calculating RMS volume sliding average approximation
|
||||
/// (time constant is about 10 sec)
|
||||
const float avgdecay = 0.99986f;
|
||||
// decimated sample block size
|
||||
static const int DECIMATED_BLOCK_SIZE = 256;
|
||||
|
||||
/// Normalization coefficient for calculating RMS sliding average approximation.
|
||||
const float avgnorm = (1 - avgdecay);
|
||||
/// Target sample rate after decimation
|
||||
static const int TARGET_SRATE = 1000;
|
||||
|
||||
/// XCorr update sequence size, update in about 200msec chunks
|
||||
static const int XCORR_UPDATE_SEQUENCE = (int)(TARGET_SRATE / 5);
|
||||
|
||||
/// Moving average N size
|
||||
static const int MOVING_AVERAGE_N = 15;
|
||||
|
||||
/// XCorr decay time constant, decay to half in 30 seconds
|
||||
/// If it's desired to have the system adapt quicker to beat rate
|
||||
/// changes within a continuing music stream, then the
|
||||
/// 'xcorr_decay_time_constant' value can be reduced, yet that
|
||||
/// can increase possibility of glitches in bpm detection.
|
||||
static const double XCORR_DECAY_TIME_CONSTANT = 30.0;
|
||||
|
||||
/// Data overlap factor for beat detection algorithm
|
||||
static const int OVERLAP_FACTOR = 4;
|
||||
|
||||
static const double TWOPI = (2 * M_PI);
|
||||
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// Enable following define to create bpm analysis file:
|
||||
|
||||
// #define _CREATE_BPM_DEBUG_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)
|
||||
static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
|
||||
{
|
||||
FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
|
||||
FILE *fptr = fopen(name, "wt");
|
||||
int i;
|
||||
|
||||
if (fptr)
|
||||
{
|
||||
printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
|
||||
printf("\nWriting BPM debug data into file %s\n", name);
|
||||
for (i = minpos; i < maxpos; i ++)
|
||||
{
|
||||
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
|
||||
@ -100,42 +110,90 @@ const float avgnorm = (1 - avgdecay);
|
||||
fclose(fptr);
|
||||
}
|
||||
}
|
||||
|
||||
void _SaveDebugBeatPos(const char *name, const std::vector<BEAT> &beats)
|
||||
{
|
||||
printf("\nWriting beat detections data into file %s\n", name);
|
||||
|
||||
FILE *fptr = fopen(name, "wt");
|
||||
if (fptr)
|
||||
{
|
||||
for (uint i = 0; i < beats.size(); i++)
|
||||
{
|
||||
BEAT b = beats[i];
|
||||
fprintf(fptr, "%lf\t%lf\n", b.pos, b.strength);
|
||||
}
|
||||
fclose(fptr);
|
||||
}
|
||||
}
|
||||
#else
|
||||
#define _SaveDebugData(a,b,c,d)
|
||||
#define _SaveDebugData(name, a,b,c,d)
|
||||
#define _SaveDebugBeatPos(name, b)
|
||||
#endif
|
||||
|
||||
// Hamming window
|
||||
void hamming(float *w, int N)
|
||||
{
|
||||
for (int i = 0; i < N; i++)
|
||||
{
|
||||
w[i] = (float)(0.54 - 0.46 * cos(TWOPI * i / (N - 1)));
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// IIR2_filter - 2nd order IIR filter
|
||||
|
||||
IIR2_filter::IIR2_filter(const double *lpf_coeffs)
|
||||
{
|
||||
memcpy(coeffs, lpf_coeffs, 5 * sizeof(double));
|
||||
memset(prev, 0, sizeof(prev));
|
||||
}
|
||||
|
||||
|
||||
float IIR2_filter::update(float x)
|
||||
{
|
||||
prev[0] = x;
|
||||
double y = x * coeffs[0];
|
||||
|
||||
for (int i = 4; i >= 1; i--)
|
||||
{
|
||||
y += coeffs[i] * prev[i];
|
||||
prev[i] = prev[i - 1];
|
||||
}
|
||||
|
||||
prev[3] = y;
|
||||
return (float)y;
|
||||
}
|
||||
|
||||
|
||||
// IIR low-pass filter coefficients, calculated with matlab/octave cheby2(2,40,0.05)
|
||||
const double _LPF_coeffs[5] = { 0.00996655391939, -0.01944529148401, 0.00996655391939, 1.96867605796247, -0.96916387431724 };
|
||||
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
|
||||
BPMDetect::BPMDetect(int numChannels, int aSampleRate)
|
||||
BPMDetect::BPMDetect(int numChannels, int aSampleRate) :
|
||||
beat_lpf(_LPF_coeffs)
|
||||
{
|
||||
beats.reserve(250); // initial reservation to prevent frequent reallocation
|
||||
|
||||
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);
|
||||
decimateBy = sampleRate / TARGET_SRATE;
|
||||
if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
|
||||
{
|
||||
ST_THROW_RT_ERROR("Too small samplerate");
|
||||
}
|
||||
|
||||
// Calculate window length & starting item according to desired min & max bpms
|
||||
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
|
||||
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
|
||||
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE);
|
||||
|
||||
assert(windowLen > windowStart);
|
||||
|
||||
@ -143,23 +201,38 @@ BPMDetect::BPMDetect(int numChannels, int aSampleRate)
|
||||
xcorr = new float[windowLen];
|
||||
memset(xcorr, 0, windowLen * sizeof(float));
|
||||
|
||||
pos = 0;
|
||||
peakPos = 0;
|
||||
peakVal = 0;
|
||||
init_scaler = 1;
|
||||
beatcorr_ringbuffpos = 0;
|
||||
beatcorr_ringbuff = new float[windowLen];
|
||||
memset(beatcorr_ringbuff, 0, windowLen * sizeof(float));
|
||||
|
||||
// allocate processing buffer
|
||||
buffer = new FIFOSampleBuffer();
|
||||
// we do processing in mono mode
|
||||
buffer->setChannels(1);
|
||||
buffer->clear();
|
||||
}
|
||||
|
||||
// calculate hamming windows
|
||||
hamw = new float[XCORR_UPDATE_SEQUENCE];
|
||||
hamming(hamw, XCORR_UPDATE_SEQUENCE);
|
||||
hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2];
|
||||
hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2);
|
||||
}
|
||||
|
||||
|
||||
BPMDetect::~BPMDetect()
|
||||
{
|
||||
delete[] xcorr;
|
||||
delete[] beatcorr_ringbuff;
|
||||
delete[] hamw;
|
||||
delete[] hamw2;
|
||||
delete buffer;
|
||||
}
|
||||
|
||||
|
||||
|
||||
/// convert to mono, low-pass filter & decimate to about 500 Hz.
|
||||
/// return number of outputted samples.
|
||||
///
|
||||
@ -216,7 +289,6 @@ int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Calculates autocorrelation function of the sample history buffer
|
||||
void BPMDetect::updateXCorr(int process_samples)
|
||||
{
|
||||
@ -224,72 +296,122 @@ void BPMDetect::updateXCorr(int process_samples)
|
||||
SAMPLETYPE *pBuffer;
|
||||
|
||||
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
|
||||
assert(process_samples == XCORR_UPDATE_SEQUENCE);
|
||||
|
||||
pBuffer = buffer->ptrBegin();
|
||||
|
||||
// calculate decay factor for xcorr filtering
|
||||
float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples));
|
||||
|
||||
// prescale pbuffer
|
||||
float tmp[XCORR_UPDATE_SEQUENCE];
|
||||
for (int i = 0; i < process_samples; i++)
|
||||
{
|
||||
tmp[i] = hamw[i] * hamw[i] * pBuffer[i];
|
||||
}
|
||||
|
||||
#pragma omp parallel for
|
||||
for (offs = windowStart; offs < windowLen; offs ++)
|
||||
{
|
||||
LONG_SAMPLETYPE sum;
|
||||
float 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
|
||||
sum += tmp[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] *= xcorr_decay; // decay 'xcorr' here with suitable time constant.
|
||||
|
||||
xcorr[offs] += (float)sum;
|
||||
xcorr[offs] += (float)fabs(sum);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// Calculates envelope of the sample data
|
||||
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
|
||||
// Detect individual beat positions
|
||||
void BPMDetect::updateBeatPos(int process_samples)
|
||||
{
|
||||
const static double decay = 0.7f; // decay constant for smoothing the envelope
|
||||
const static double norm = (1 - decay);
|
||||
SAMPLETYPE *pBuffer;
|
||||
|
||||
int i;
|
||||
LONG_SAMPLETYPE out;
|
||||
double val;
|
||||
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
|
||||
|
||||
for (i = 0; i < numsamples; i ++)
|
||||
pBuffer = buffer->ptrBegin();
|
||||
assert(process_samples == XCORR_UPDATE_SEQUENCE / 2);
|
||||
|
||||
// static double thr = 0.0003;
|
||||
double posScale = (double)this->decimateBy / (double)this->sampleRate;
|
||||
int resetDur = (int)(0.12 / posScale + 0.5);
|
||||
|
||||
// prescale pbuffer
|
||||
float tmp[XCORR_UPDATE_SEQUENCE / 2];
|
||||
for (int i = 0; i < process_samples; i++)
|
||||
{
|
||||
// calc average RMS volume
|
||||
RMSVolumeAccu *= avgdecay;
|
||||
val = (float)fabs((float)samples[i]);
|
||||
RMSVolumeAccu += val * val;
|
||||
tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
|
||||
}
|
||||
|
||||
// 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))
|
||||
#pragma omp parallel for
|
||||
for (int offs = windowStart; offs < windowLen; offs++)
|
||||
{
|
||||
float sum = 0;
|
||||
for (int i = 0; i < process_samples; i++)
|
||||
{
|
||||
val = 0;
|
||||
sum += tmp[i] * pBuffer[offs + i];
|
||||
}
|
||||
beatcorr_ringbuff[(beatcorr_ringbuffpos + offs) % windowLen] += (float)((sum > 0) ? sum : 0); // accumulate only positive correlations
|
||||
}
|
||||
|
||||
int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
|
||||
|
||||
// compensate empty buffer at beginning by scaling coefficient
|
||||
float scale = (float)windowLen / (float)(skipstep * init_scaler);
|
||||
if (scale > 1.0f)
|
||||
{
|
||||
init_scaler++;
|
||||
}
|
||||
else
|
||||
{
|
||||
scale = 1.0f;
|
||||
}
|
||||
|
||||
// detect beats
|
||||
for (int i = 0; i < skipstep; i++)
|
||||
{
|
||||
float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
|
||||
sum -= beat_lpf.update(sum);
|
||||
|
||||
if (sum > peakVal)
|
||||
{
|
||||
// found new local largest value
|
||||
peakVal = sum;
|
||||
peakPos = pos;
|
||||
}
|
||||
if (pos > peakPos + resetDur)
|
||||
{
|
||||
// largest value not updated for 200msec => accept as beat
|
||||
peakPos += skipstep;
|
||||
if (peakVal > 0)
|
||||
{
|
||||
// add detected beat to end of "beats" vector
|
||||
BEAT temp = { (float)(peakPos * posScale), (float)(peakVal * scale) };
|
||||
beats.push_back(temp);
|
||||
}
|
||||
|
||||
peakVal = 0;
|
||||
peakPos = pos;
|
||||
}
|
||||
|
||||
// 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;
|
||||
beatcorr_ringbuff[beatcorr_ringbuffpos] = 0;
|
||||
pos++;
|
||||
beatcorr_ringbuffpos = (beatcorr_ringbuffpos + 1) % windowLen;
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
#define max(x,y) ((x) > (y) ? (x) : (y))
|
||||
|
||||
void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
|
||||
{
|
||||
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
|
||||
SAMPLETYPE decimated[DECIMATED_BLOCK_SIZE];
|
||||
|
||||
// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
|
||||
while (numSamples > 0)
|
||||
@ -297,48 +419,70 @@ void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
|
||||
int block;
|
||||
int decSamples;
|
||||
|
||||
block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
|
||||
block = (numSamples > INPUT_BLOCK_SIZE) ? INPUT_BLOCK_SIZE : 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)
|
||||
// when the buffer has enough samples for processing...
|
||||
int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
|
||||
while ((int)buffer->numSamples() >= req)
|
||||
{
|
||||
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);
|
||||
// ... update autocorrelations...
|
||||
updateXCorr(XCORR_UPDATE_SEQUENCE);
|
||||
// ...update beat position calculation...
|
||||
updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
|
||||
// ... and remove proceessed samples from the buffer
|
||||
int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
|
||||
buffer->receiveSamples(n);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
|
||||
void BPMDetect::removeBias()
|
||||
{
|
||||
int i;
|
||||
float minval = 1e12f; // arbitrary large number
|
||||
|
||||
// Remove linear bias: calculate linear regression coefficient
|
||||
// 1. calc mean of 'xcorr' and 'i'
|
||||
double mean_i = 0;
|
||||
double mean_x = 0;
|
||||
for (i = windowStart; i < windowLen; i++)
|
||||
{
|
||||
mean_x += xcorr[i];
|
||||
}
|
||||
mean_x /= (windowLen - windowStart);
|
||||
mean_i = 0.5 * (windowLen - 1 + windowStart);
|
||||
|
||||
// 2. calculate linear regression coefficient
|
||||
double b = 0;
|
||||
double div = 0;
|
||||
for (i = windowStart; i < windowLen; i++)
|
||||
{
|
||||
double xt = xcorr[i] - mean_x;
|
||||
double xi = i - mean_i;
|
||||
b += xt * xi;
|
||||
div += xi * xi;
|
||||
}
|
||||
b /= div;
|
||||
|
||||
// subtract linear regression and resolve min. value bias
|
||||
float minval = FLT_MAX; // arbitrary large number
|
||||
for (i = windowStart; i < windowLen; i ++)
|
||||
{
|
||||
xcorr[i] -= (float)(b * i);
|
||||
if (xcorr[i] < minval)
|
||||
{
|
||||
minval = xcorr[i];
|
||||
}
|
||||
}
|
||||
|
||||
// subtract min.value
|
||||
for (i = windowStart; i < windowLen; i ++)
|
||||
{
|
||||
xcorr[i] -= minval;
|
||||
@ -346,26 +490,82 @@ void BPMDetect::removeBias()
|
||||
}
|
||||
|
||||
|
||||
// Calculate N-point moving average for "source" values
|
||||
void MAFilter(float *dest, const float *source, int start, int end, int N)
|
||||
{
|
||||
for (int i = start; i < end; i++)
|
||||
{
|
||||
int i1 = i - N / 2;
|
||||
int i2 = i + N / 2 + 1;
|
||||
if (i1 < start) i1 = start;
|
||||
if (i2 > end) i2 = end;
|
||||
|
||||
double sum = 0;
|
||||
for (int j = i1; j < i2; j ++)
|
||||
{
|
||||
sum += source[j];
|
||||
}
|
||||
dest[i] = (float)(sum / (i2 - i1));
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
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();
|
||||
|
||||
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
|
||||
|
||||
// save bpm debug data if debug data writing enabled
|
||||
_SaveDebugData("soundtouch-bpm-xcorr.txt", xcorr, windowStart, windowLen, coeff);
|
||||
|
||||
// Smoothen by N-point moving-average
|
||||
float *data = new float[windowLen];
|
||||
memset(data, 0, sizeof(float) * windowLen);
|
||||
MAFilter(data, xcorr, windowStart, windowLen, MOVING_AVERAGE_N);
|
||||
|
||||
// find peak position
|
||||
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
|
||||
peakPos = peakFinder.detectPeak(data, windowStart, windowLen);
|
||||
|
||||
// save bpm debug data if debug data writing enabled
|
||||
_SaveDebugData("soundtouch-bpm-smoothed.txt", data, windowStart, windowLen, coeff);
|
||||
|
||||
delete[] data;
|
||||
|
||||
assert(decimateBy != 0);
|
||||
if (peakPos < 1e-9) return 0.0; // detection failed.
|
||||
|
||||
_SaveDebugBeatPos("soundtouch-detected-beats.txt", beats);
|
||||
|
||||
// calculate BPM
|
||||
return (float) (coeff / peakPos);
|
||||
float bpm = (float)(coeff / peakPos);
|
||||
return (bpm >= MIN_BPM && bpm <= MAX_BPM_VALID) ? bpm : 0;
|
||||
}
|
||||
|
||||
|
||||
/// Get beat position arrays. Note: The array includes also really low beat detection values
|
||||
/// in absence of clear strong beats. Consumer may wish to filter low values away.
|
||||
/// - "pos" receive array of beat positions
|
||||
/// - "values" receive array of beat detection strengths
|
||||
/// - max_num indicates max.size of "pos" and "values" array.
|
||||
///
|
||||
/// You can query a suitable array sized by calling this with nullptr in "pos" & "values".
|
||||
///
|
||||
/// \return number of beats in the arrays.
|
||||
int BPMDetect::getBeats(float *pos, float *values, int max_num)
|
||||
{
|
||||
int num = (int)beats.size();
|
||||
if ((!pos) || (!values)) return num; // pos or values nullptr, return just size
|
||||
|
||||
for (int i = 0; (i < num) && (i < max_num); i++)
|
||||
{
|
||||
pos[i] = beats[i].pos;
|
||||
values[i] = beats[i].strength;
|
||||
}
|
||||
return num;
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user