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Merge pull request #11706 from Minty-Meeo/update-soundtouch 

Update SoundTouch to 2.3.2 commit 1eda9c0b01039f29d230a46cda9f2290bbd1f62b
This commit is contained in:
Admiral H. Curtiss 2023-04-02 15:04:37 +02:00 committed by GitHub
parent aaeaa9c6b6 0e7e8d26f0
commit 0cd70c2aa5
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GPG Key ID: 4AEE18F83AFDEB23
30 changed files with 1912 additions and 1690 deletions
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18
Externals/soundtouch/AAFilter.cpp vendored
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@ -12,13 +12,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2014-01-05 23:40:22 +0200 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -49,7 +42,7 @@
using namespace soundtouch; using namespace soundtouch;
#define PI 3.141592655357989 #define PI 3.14159265358979323846
#define TWOPI (2 * PI) #define TWOPI (2 * PI)
// define this to save AA filter coefficients to a file // define this to save AA filter coefficients to a file
@ -61,7 +54,7 @@ using namespace soundtouch;
static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len) static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
{ {
FILE *fptr = fopen("aa_filter_coeffs.txt", "wt"); FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
if (fptr == NULL) return; if (fptr == nullptr) return;
for (int i = 0; i < len; i ++) for (int i = 0; i < len; i ++)
{ {
@ -75,7 +68,6 @@ using namespace soundtouch;
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y) #define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif #endif
/***************************************************************************** /*****************************************************************************
* *
* Implementation of the class 'AAFilter' * Implementation of the class 'AAFilter'
@ -90,14 +82,12 @@ AAFilter::AAFilter(uint len)
} }
AAFilter::~AAFilter() AAFilter::~AAFilter()
{ {
delete pFIR; delete pFIR;
} }
// Sets new anti-alias filter cut-off edge frequency, scaled to // Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5). // sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency. // The filter will cut frequencies higher than the given frequency.
@ -108,7 +98,6 @@ void AAFilter::setCutoffFreq(double newCutoffFreq)
} }
// Sets number of FIR filter taps // Sets number of FIR filter taps
void AAFilter::setLength(uint newLength) void AAFilter::setLength(uint newLength)
{ {
@ -117,7 +106,6 @@ void AAFilter::setLength(uint newLength)
} }
// Calculates coefficients for a low-pass FIR filter using Hamming window // Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs() void AAFilter::calculateCoeffs()
{ {
@ -177,12 +165,10 @@ void AAFilter::calculateCoeffs()
for (i = 0; i < length; i ++) for (i = 0; i < length; i ++)
{ {
temp = work[i] * scaleCoeff; temp = work[i] * scaleCoeff;
//#if SOUNDTOUCH_INTEGER_SAMPLES
// scale & round to nearest integer // scale & round to nearest integer
temp += (temp >= 0) ? 0.5 : -0.5; temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods // ensure no overfloods
assert(temp >= -32768 && temp <= 32767); assert(temp >= -32768 && temp <= 32767);
//#endif
coeffs[i] = (SAMPLETYPE)temp; coeffs[i] = (SAMPLETYPE)temp;
} }

7
Externals/soundtouch/AAFilter.h vendored
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@ -13,13 +13,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2014-01-07 21:41:23 +0200 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library

397
Externals/soundtouch/BPMDetect.cpp vendored
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@ -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 : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -58,22 +51,40 @@
#include <assert.h> #include <assert.h>
#include <string.h> #include <string.h>
#include <stdio.h> #include <stdio.h>
#include <cfloat>
#include "FIFOSampleBuffer.h" #include "FIFOSampleBuffer.h"
#include "PeakFinder.h" #include "PeakFinder.h"
#include "BPMDetect.h" #include "BPMDetect.h"
using namespace soundtouch; using namespace soundtouch;
#define INPUT_BLOCK_SAMPLES 2048 // algorithm input sample block size
#define DECIMATED_BLOCK_SAMPLES 256 static const int INPUT_BLOCK_SIZE = 2048;
/// decay constant for calculating RMS volume sliding average approximation // decimated sample block size
/// (time constant is about 10 sec) static const int DECIMATED_BLOCK_SIZE = 256;
const float avgdecay = 0.99986f;
/// Normalization coefficient for calculating RMS sliding average approximation. /// Target sample rate after decimation
const float avgnorm = (1 - avgdecay); 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;
#define PI 3.14159265358979323846
#define TWOPI (2 * PI)
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
@ -83,16 +94,14 @@ const float avgnorm = (1 - avgdecay);
#ifdef _CREATE_BPM_DEBUG_FILE #ifdef _CREATE_BPM_DEBUG_FILE
#define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt" static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
{ {
FILE *fptr = fopen(DEBUGFILE_NAME, "wt"); FILE *fptr = fopen(name, "wt");
int i; int i;
if (fptr) 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 ++) for (i = minpos; i < maxpos; i ++)
{ {
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]); fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
@ -100,42 +109,90 @@ const float avgnorm = (1 - avgdecay);
fclose(fptr); 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 #else
#define _SaveDebugData(a,b,c,d) #define _SaveDebugData(name, a,b,c,d)
#define _SaveDebugBeatPos(name, b)
#endif #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->sampleRate = aSampleRate;
this->channels = numChannels; this->channels = numChannels;
decimateSum = 0; decimateSum = 0;
decimateCount = 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 // choose decimation factor so that result is approx. 1000 Hz
decimateBy = sampleRate / 1000; decimateBy = sampleRate / TARGET_SRATE;
assert(decimateBy > 0); if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES); {
ST_THROW_RT_ERROR("Too small samplerate");
}
// Calculate window length & starting item according to desired min & max bpms // Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM); windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM); windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE);
assert(windowLen > windowStart); assert(windowLen > windowStart);
@ -143,23 +200,38 @@ BPMDetect::BPMDetect(int numChannels, int aSampleRate)
xcorr = new float[windowLen]; xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float)); 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 // allocate processing buffer
buffer = new FIFOSampleBuffer(); buffer = new FIFOSampleBuffer();
// we do processing in mono mode // we do processing in mono mode
buffer->setChannels(1); buffer->setChannels(1);
buffer->clear(); 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() BPMDetect::~BPMDetect()
{ {
delete[] xcorr; delete[] xcorr;
delete[] beatcorr_ringbuff;
delete[] hamw;
delete[] hamw2;
delete buffer; delete buffer;
} }
/// convert to mono, low-pass filter & decimate to about 500 Hz. /// convert to mono, low-pass filter & decimate to about 500 Hz.
/// return number of outputted samples. /// return number of outputted samples.
/// ///
@ -216,7 +288,6 @@ int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
} }
// Calculates autocorrelation function of the sample history buffer // Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples) void BPMDetect::updateXCorr(int process_samples)
{ {
@ -224,72 +295,122 @@ void BPMDetect::updateXCorr(int process_samples)
SAMPLETYPE *pBuffer; SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen)); assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
assert(process_samples == XCORR_UPDATE_SEQUENCE);
pBuffer = buffer->ptrBegin(); 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 #pragma omp parallel for
for (offs = windowStart; offs < windowLen; offs ++) for (offs = windowStart; offs < windowLen; offs ++)
{ {
LONG_SAMPLETYPE sum; float sum;
int i; int i;
sum = 0; sum = 0;
for (i = 0; i < process_samples; i ++) 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 xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable time constant.
// 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; xcorr[offs] += (float)fabs(sum);
} }
} }
// Calculates envelope of the sample data // Detect individual beat positions
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples) void BPMDetect::updateBeatPos(int process_samples)
{ {
const static double decay = 0.7f; // decay constant for smoothing the envelope SAMPLETYPE *pBuffer;
const static double norm = (1 - decay);
int i; assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
LONG_SAMPLETYPE out;
double val;
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 tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
RMSVolumeAccu *= avgdecay; }
val = (float)fabs((float)samples[i]);
RMSVolumeAccu += val * val;
// cut amplitudes that are below cutoff ~2 times RMS volume #pragma omp parallel for
// (we're interested in peak values, not the silent moments) for (int offs = windowStart; offs < windowLen; offs++)
if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
{ {
val = 0; float sum = 0;
for (int i = 0; i < process_samples; i++)
{
sum += tmp[i] * pBuffer[offs + i];
}
beatcorr_ringbuff[(beatcorr_ringbuffpos + offs) % windowLen] += (float)((sum > 0) ? sum : 0); // accumulate only positive correlations
} }
// smooth amplitude envelope int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
#ifdef SOUNDTOUCH_INTEGER_SAMPLES // compensate empty buffer at beginning by scaling coefficient
// cut peaks (shouldn't be necessary though) float scale = (float)windowLen / (float)(skipstep * init_scaler);
if (out > 32767) out = 32767; if (scale > 1.0f)
#endif // SOUNDTOUCH_INTEGER_SAMPLES {
samples[i] = (SAMPLETYPE)out; 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;
}
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) 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 // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
while (numSamples > 0) while (numSamples > 0)
@ -297,48 +418,70 @@ void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
int block; int block;
int decSamples; 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 // decimate. note that converts to mono at the same time
decSamples = decimate(decimated, samples, block); decSamples = decimate(decimated, samples, block);
samples += block * channels; samples += block * channels;
numSamples -= block; numSamples -= block;
// envelope new samples and add them to buffer
calcEnvelope(decimated, decSamples);
buffer->putSamples(decimated, decSamples); buffer->putSamples(decimated, decSamples);
} }
// when the buffer has enought samples for processing... // when the buffer has enough samples for processing...
if ((int)buffer->numSamples() > windowLen) int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
while ((int)buffer->numSamples() >= req)
{ {
int processLength; // ... update autocorrelations...
updateXCorr(XCORR_UPDATE_SEQUENCE);
// how many samples are processed // ...update beat position calculation...
processLength = (int)buffer->numSamples() - windowLen; updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
// ... and remove proceessed samples from the buffer
// ... calculate autocorrelations for oldest samples... int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
updateXCorr(processLength); buffer->receiveSamples(n);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
} }
} }
void BPMDetect::removeBias() void BPMDetect::removeBias()
{ {
int i; 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++) 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) if (xcorr[i] < minval)
{ {
minval = xcorr[i]; minval = xcorr[i];
} }
} }
// subtract min.value
for (i = windowStart; i < windowLen; i ++) for (i = windowStart; i < windowLen; i ++)
{ {
xcorr[i] -= minval; xcorr[i] -= minval;
@ -346,26 +489,82 @@ 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() float BPMDetect::getBpm()
{ {
double peakPos; double peakPos;
double coeff; double coeff;
PeakFinder peakFinder; 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 // remove bias from xcorr data
removeBias(); 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 // 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); assert(decimateBy != 0);
if (peakPos < 1e-9) return 0.0; // detection failed. if (peakPos < 1e-9) return 0.0; // detection failed.
_SaveDebugBeatPos("soundtouch-detected-beats.txt", beats);
// calculate BPM // 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;
} }

79
Externals/soundtouch/BPMDetect.h vendored
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@ -26,13 +26,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2012-08-30 22:53:44 +0300 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -57,6 +50,7 @@
#ifndef _BPMDetect_H_ #ifndef _BPMDetect_H_
#define _BPMDetect_H_ #define _BPMDetect_H_
#include <vector>
#include "STTypes.h" #include "STTypes.h"
#include "FIFOSampleBuffer.h" #include "FIFOSampleBuffer.h"
@ -64,10 +58,32 @@ namespace soundtouch
{ {
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit. /// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 29 #define MIN_BPM 45
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit. /// Maximum allowed BPM rate range. Used for calculating algorithm parametrs
#define MAX_BPM 200 #define MAX_BPM_RANGE 200
/// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM_VALID 190
////////////////////////////////////////////////////////////////////////////////
typedef struct
{
float pos;
float strength;
} BEAT;
class IIR2_filter
{
double coeffs[5];
double prev[5];
public:
IIR2_filter(const double *lpf_coeffs);
float update(float x);
};
/// Class for calculating BPM rate for audio data. /// Class for calculating BPM rate for audio data.
@ -77,12 +93,6 @@ protected:
/// Auto-correlation accumulator bins. /// Auto-correlation accumulator bins.
float *xcorr; float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
double envelopeAccu;
/// RMS volume sliding average approximation level accumulator
double RMSVolumeAccu;
/// Sample average counter. /// Sample average counter.
int decimateCount; int decimateCount;
@ -105,9 +115,28 @@ protected:
/// the first these many correlation bins. /// the first these many correlation bins.
int windowStart; int windowStart;
/// window functions for data preconditioning
float *hamw;
float *hamw2;
// beat detection variables
int pos;
int peakPos;
int beatcorr_ringbuffpos;
int init_scaler;
float peakVal;
float *beatcorr_ringbuff;
/// FIFO-buffer for decimated processing samples. /// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer; soundtouch::FIFOSampleBuffer *buffer;
/// Collection of detected beat positions
//BeatCollection beats;
std::vector<BEAT> beats;
// 2nd order low-pass-filter
IIR2_filter beat_lpf;
/// Updates auto-correlation function for given number of decimated samples that /// Updates auto-correlation function for given number of decimated samples that
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though). /// though).
@ -131,6 +160,10 @@ protected:
/// remove constant bias from xcorr data /// remove constant bias from xcorr data
void removeBias(); void removeBias();
// Detect individual beat positions
void updateBeatPos(int process_samples);
public: public:
/// Constructor. /// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data. BPMDetect(int numChannels, ///< Number of channels in sample data.
@ -150,15 +183,23 @@ public:
int numSamples ///< Number of samples in buffer int numSamples ///< Number of samples in buffer
); );
/// Analyzes the results and returns the BPM rate. Use this function to read result /// Analyzes the results and returns the BPM rate. Use this function to read result
/// after whole song data has been input to the class by consecutive calls of /// after whole song data has been input to the class by consecutive calls of
/// 'inputSamples' function. /// 'inputSamples' function.
/// ///
/// \return Beats-per-minute rate, or zero if detection failed. /// \return Beats-per-minute rate, or zero if detection failed.
float getBpm(); float getBpm();
/// 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 getBeats(float *pos, float *strength, int max_num);
}; };
} }
#endif // _BPMDetect_H_ #endif // _BPMDetect_H_

4
Externals/soundtouch/CMakeLists.txt vendored
View File

@ -1,3 +1,6 @@
# OSX meeds to know
check_and_add_flag(CXX11 -std=c++11)
set(SRCS set(SRCS
AAFilter.cpp AAFilter.cpp
BPMDetect.cpp BPMDetect.cpp
@ -17,4 +20,3 @@ set(SRCS
add_library(SoundTouch STATIC ${SRCS}) add_library(SoundTouch STATIC ${SRCS})
dolphin_disable_warnings_msvc(SoundTouch) dolphin_disable_warnings_msvc(SoundTouch)
add_definitions(-w)

31
Externals/soundtouch/FIFOSampleBuffer.cpp vendored
View File

@ -15,13 +15,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -57,8 +50,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
{ {
assert(numChannels > 0); assert(numChannels > 0);
sizeInBytes = 0; // reasonable initial value sizeInBytes = 0; // reasonable initial value
buffer = NULL; buffer = nullptr;
bufferUnaligned = NULL; bufferUnaligned = nullptr;
samplesInBuffer = 0; samplesInBuffer = 0;
bufferPos = 0; bufferPos = 0;
channels = (uint)numChannels; channels = (uint)numChannels;
@ -70,8 +63,8 @@ FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
FIFOSampleBuffer::~FIFOSampleBuffer() FIFOSampleBuffer::~FIFOSampleBuffer()
{ {
delete[] bufferUnaligned; delete[] bufferUnaligned;
bufferUnaligned = NULL; bufferUnaligned = nullptr;
buffer = NULL; buffer = nullptr;
} }
@ -80,7 +73,8 @@ void FIFOSampleBuffer::setChannels(int numChannels)
{ {
uint usedBytes; uint usedBytes;
assert(numChannels > 0); if (!verifyNumberOfChannels(numChannels)) return;
usedBytes = channels * samplesInBuffer; usedBytes = channels * samplesInBuffer;
channels = (uint)numChannels; channels = (uint)numChannels;
samplesInBuffer = usedBytes / channels; samplesInBuffer = usedBytes / channels;
@ -131,7 +125,7 @@ void FIFOSampleBuffer::putSamples(uint nSamples)
// //
// Parameter 'slackCapacity' tells the function how much free capacity (in // Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to // terms of samples) there _at least_ should be, in order to the caller to
// succesfully insert all the required samples to the buffer. When necessary, // successfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement. // the function grows the buffer size to comply with this requirement.
// //
// When using this function as means for inserting new samples, also remember // When using this function as means for inserting new samples, also remember
@ -158,7 +152,7 @@ SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
} }
// Ensures that the buffer has enought capacity, i.e. space for _at least_ // Ensures that the buffer has enough capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of // 'capacityRequirement' number of samples. The buffer is grown in steps of
// 4 kilobytes to eliminate the need for frequently growing up the buffer, // 4 kilobytes to eliminate the need for frequently growing up the buffer,
// as well as to round the buffer size up to the virtual memory page size. // as well as to round the buffer size up to the virtual memory page size.
@ -172,7 +166,7 @@ void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096; sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
assert(sizeInBytes % 2 == 0); assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)]; tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == NULL) if (tempUnaligned == nullptr)
{ {
ST_THROW_RT_ERROR("Couldn't allocate memory!\n"); ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
} }
@ -272,3 +266,10 @@ uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
return samplesInBuffer; return samplesInBuffer;
} }
/// Add silence to end of buffer
void FIFOSampleBuffer::addSilent(uint nSamples)
{
memset(ptrEnd(nSamples), 0, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}

30
Externals/soundtouch/FIFOSampleBuffer.h vendored
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@ -15,13 +15,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2014-01-05 23:40:22 +0200 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -98,7 +91,7 @@ public:
); );
/// destructor /// destructor
~FIFOSampleBuffer(); ~FIFOSampleBuffer() override;
/// Returns a pointer to the beginning of the output samples. /// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly. /// This function is provided for accessing the output samples directly.
@ -107,7 +100,7 @@ public:
/// When using this function to output samples, also remember to 'remove' the /// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the /// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function /// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin(); virtual SAMPLETYPE *ptrBegin() override;
/// Returns a pointer to the end of the used part of the sample buffer (i.e. /// Returns a pointer to the end of the used part of the sample buffer (i.e.
/// where the new samples are to be inserted). This function may be used for /// where the new samples are to be inserted). This function may be used for
@ -119,7 +112,7 @@ public:
/// 'putSamples(numSamples)' function. /// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd( SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_ uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can succesfully insert the ///< should be so that the caller can successfully insert the
///< desired samples to the buffer. If necessary, the function ///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement. ///< grows the buffer size to comply with this requirement.
); );
@ -128,7 +121,7 @@ public:
/// the sample buffer. /// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples. virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert. uint numSamples ///< Number of samples to insert.
); ) override;
/// Adjusts the book-keeping to increase number of samples in the buffer without /// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples. /// copying any actual samples.
@ -146,7 +139,7 @@ public:
/// \return Number of samples returned. /// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples. virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max. uint maxSamples ///< How many samples to receive at max.
); ) override;
/// Adjusts book-keeping so that given number of samples are removed from beginning of the /// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere. /// sample buffer without copying them anywhere.
@ -154,10 +147,10 @@ public:
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function. /// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe. virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
); ) override;
/// Returns number of samples currently available. /// Returns number of samples currently available.
virtual uint numSamples() const; virtual uint numSamples() const override;
/// Sets number of channels, 1 = mono, 2 = stereo. /// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(int numChannels); void setChannels(int numChannels);
@ -169,14 +162,17 @@ public:
} }
/// Returns nonzero if there aren't any samples available for outputting. /// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const; virtual int isEmpty() const override;
/// Clears all the samples. /// Clears all the samples.
virtual void clear(); virtual void clear() override;
/// allow trimming (downwards) amount of samples in pipeline. /// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples /// Returns adjusted amount of samples
uint adjustAmountOfSamples(uint numSamples); uint adjustAmountOfSamples(uint numSamples) override;
/// Add silence to end of buffer
void addSilent(uint nSamples);
}; };
} }

48
Externals/soundtouch/FIFOSamplePipe.h vendored
View File

@ -17,13 +17,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -58,6 +51,18 @@ namespace soundtouch
/// Abstract base class for FIFO (first-in-first-out) sample processing classes. /// Abstract base class for FIFO (first-in-first-out) sample processing classes.
class FIFOSamplePipe class FIFOSamplePipe
{ {
protected:
bool verifyNumberOfChannels(int nChannels) const
{
if ((nChannels > 0) && (nChannels <= SOUNDTOUCH_MAX_CHANNELS))
{
return true;
}
ST_THROW_RT_ERROR("Error: Illegal number of channels");
return false;
}
public: public:
// virtual default destructor // virtual default destructor
virtual ~FIFOSamplePipe() {} virtual ~FIFOSamplePipe() {}
@ -122,7 +127,6 @@ public:
}; };
/// Base-class for sound processing routines working in FIFO principle. With this base /// Base-class for sound processing routines working in FIFO principle. With this base
/// class it's easy to implement sound processing stages that can be chained together, /// class it's easy to implement sound processing stages that can be chained together,
/// so that samples that are fed into beginning of the pipe automatically go through /// so that samples that are fed into beginning of the pipe automatically go through
@ -140,20 +144,18 @@ protected:
/// Sets output pipe. /// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput) void setOutPipe(FIFOSamplePipe *pOutput)
{ {
assert(output == NULL); assert(output == nullptr);
assert(pOutput != NULL); assert(pOutput != nullptr);
output = pOutput; output = pOutput;
} }
/// Constructor. Doesn't define output pipe; it has to be set be /// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function. /// 'setOutPipe' function.
FIFOProcessor() FIFOProcessor()
{ {
output = NULL; output = nullptr;
} }
/// Constructor. Configures output pipe. /// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe. FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
) )
@ -161,13 +163,11 @@ protected:
output = pOutput; output = pOutput;
} }
/// Destructor. /// Destructor.
virtual ~FIFOProcessor() virtual ~FIFOProcessor() override
{ {
} }
/// Returns a pointer to the beginning of the output samples. /// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly. /// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping! /// Please be careful for not to corrupt the book-keeping!
@ -175,7 +175,7 @@ protected:
/// When using this function to output samples, also remember to 'remove' the /// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the /// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function /// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() virtual SAMPLETYPE *ptrBegin() override
{ {
return output->ptrBegin(); return output->ptrBegin();
} }
@ -189,44 +189,40 @@ public:
/// \return Number of samples returned. /// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples. virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max. uint maxSamples ///< How many samples to receive at max.
) ) override
{ {
return output->receiveSamples(outBuffer, maxSamples); return output->receiveSamples(outBuffer, maxSamples);
} }
/// Adjusts book-keeping so that given number of samples are removed from beginning of the /// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere. /// sample buffer without copying them anywhere.
/// ///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function. /// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe. virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) ) override
{ {
return output->receiveSamples(maxSamples); return output->receiveSamples(maxSamples);
} }
/// Returns number of samples currently available. /// Returns number of samples currently available.
virtual uint numSamples() const virtual uint numSamples() const override
{ {
return output->numSamples(); return output->numSamples();
} }
/// Returns nonzero if there aren't any samples available for outputting. /// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const virtual int isEmpty() const override
{ {
return output->isEmpty(); return output->isEmpty();
} }
/// allow trimming (downwards) amount of samples in pipeline. /// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples /// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples) virtual uint adjustAmountOfSamples(uint numSamples) override
{ {
return output->adjustAmountOfSamples(numSamples); return output->adjustAmountOfSamples(numSamples);
} }
}; };
} }

128
Externals/soundtouch/FIRFilter.cpp vendored
View File

@ -2,22 +2,21 @@
/// ///
/// General FIR digital filter routines with MMX optimization. /// General FIR digital filter routines with MMX optimization.
/// ///
/// Note : MMX optimized functions reside in a separate, platform-specific file, /// Notes : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp' /// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
/// ///
/// This source file contains OpenMP optimizations that allow speeding up the
/// corss-correlation algorithm by executing it in several threads / CPU cores
/// in parallel. See the following article link for more detailed discussion
/// about SoundTouch OpenMP optimizations:
/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
///
/// Author : Copyright (c) Olli Parviainen /// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi /// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch /// 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: FIRFilter.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -60,53 +59,43 @@ FIRFilter::FIRFilter()
resultDivider = 0; resultDivider = 0;
length = 0; length = 0;
lengthDiv8 = 0; lengthDiv8 = 0;
filterCoeffs = NULL; filterCoeffs = nullptr;
filterCoeffsStereo = nullptr;
} }
FIRFilter::~FIRFilter() FIRFilter::~FIRFilter()
{ {
delete[] filterCoeffs; delete[] filterCoeffs;
delete[] filterCoeffsStereo;
} }
// Usual C-version of the filter routine for stereo sound // Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{ {
int j, end; int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES // hint compiler autovectorization that loop length is divisible by 8
// when using floating point samples, use a scaler instead of a divider uint ilength = length & -8;
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0); assert((length != 0) && (length == ilength) && (src != nullptr) && (dest != nullptr) && (filterCoeffs != nullptr));
assert(src != NULL); assert(numSamples > ilength);
assert(dest != NULL);
assert(filterCoeffs != NULL);
end = 2 * (numSamples - length); end = 2 * (numSamples - ilength);
#pragma omp parallel for #pragma omp parallel for
for (j = 0; j < end; j += 2) for (j = 0; j < end; j += 2)
{ {
const SAMPLETYPE *ptr; const SAMPLETYPE *ptr;
LONG_SAMPLETYPE suml, sumr; LONG_SAMPLETYPE suml, sumr;
uint i;
suml = sumr = 0; suml = sumr = 0;
ptr = src + j; ptr = src + j;
for (i = 0; i < length; i += 4) for (uint i = 0; i < ilength; i ++)
{ {
// loop is unrolled by factor of 4 here for efficiency suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] + sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
} }
#ifdef SOUNDTOUCH_INTEGER_SAMPLES #ifdef SOUNDTOUCH_INTEGER_SAMPLES
@ -116,54 +105,41 @@ uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml; suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits // saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr; sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES #endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml; dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr; dest[j + 1] = (SAMPLETYPE)sumr;
} }
return numSamples - length; return numSamples - ilength;
} }
// Usual C-version of the filter routine for mono sound // Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{ {
int j, end; int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0); // hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numSamples - length; assert(ilength != 0);
end = numSamples - ilength;
#pragma omp parallel for #pragma omp parallel for
for (j = 0; j < end; j ++) for (j = 0; j < end; j ++)
{ {
const SAMPLETYPE *pSrc = src + j; const SAMPLETYPE *pSrc = src + j;
LONG_SAMPLETYPE sum; LONG_SAMPLETYPE sum;
uint i; int i;
sum = 0; sum = 0;
for (i = 0; i < length; i += 4) for (i = 0; i < ilength; i ++)
{ {
// loop is unrolled by factor of 4 here for efficiency sum += pSrc[i] * filterCoeffs[i];
sum += pSrc[i + 0] * filterCoeffs[i + 0] +
pSrc[i + 1] * filterCoeffs[i + 1] +
pSrc[i + 2] * filterCoeffs[i + 2] +
pSrc[i + 3] * filterCoeffs[i + 3];
} }
#ifdef SOUNDTOUCH_INTEGER_SAMPLES #ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor; sum >>= resultDivFactor;
// saturate to 16 bit integer limits // saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum; sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES #endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum; dest[j] = (SAMPLETYPE)sum;
} }
@ -175,26 +151,24 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
{ {
int j, end; int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0); assert(length != 0);
assert(src != NULL); assert(src != nullptr);
assert(dest != NULL); assert(dest != nullptr);
assert(filterCoeffs != NULL); assert(filterCoeffs != nullptr);
assert(numChannels < 16); assert(numChannels < 16);
end = numChannels * (numSamples - length); // hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numChannels * (numSamples - ilength);
#pragma omp parallel for #pragma omp parallel for
for (j = 0; j < end; j += numChannels) for (j = 0; j < end; j += numChannels)
{ {
const SAMPLETYPE *ptr; const SAMPLETYPE *ptr;
LONG_SAMPLETYPE sums[16]; LONG_SAMPLETYPE sums[16];
uint c, i; uint c;
int i;
for (c = 0; c < numChannels; c ++) for (c = 0; c < numChannels; c ++)
{ {
@ -203,7 +177,7 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
ptr = src + j; ptr = src + j;
for (i = 0; i < length; i ++) for (i = 0; i < ilength; i ++)
{ {
SAMPLETYPE coef=filterCoeffs[i]; SAMPLETYPE coef=filterCoeffs[i];
for (c = 0; c < numChannels; c ++) for (c = 0; c < numChannels; c ++)
@ -217,13 +191,11 @@ uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uin
{ {
#ifdef SOUNDTOUCH_INTEGER_SAMPLES #ifdef SOUNDTOUCH_INTEGER_SAMPLES
sums[c] >>= resultDivFactor; sums[c] >>= resultDivFactor;
#else
sums[c] *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES #endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j+c] = (SAMPLETYPE)sums[c]; dest[j+c] = (SAMPLETYPE)sums[c];
} }
} }
return numSamples - length; return numSamples - ilength;
} }
@ -235,6 +207,13 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
assert(newLength > 0); assert(newLength > 0);
if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8"); if (newLength % 8) ST_THROW_RT_ERROR("FIR filter length not divisible by 8");
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// scale coefficients already here if using floating samples
double scale = 1.0 / resultDivider;
#else
short scale = 1;
#endif
lengthDiv8 = newLength / 8; lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8; length = lengthDiv8 * 8;
assert(length == newLength); assert(length == newLength);
@ -244,7 +223,16 @@ void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint u
delete[] filterCoeffs; delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length]; filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE)); delete[] filterCoeffsStereo;
filterCoeffsStereo = new SAMPLETYPE[length*2];
for (uint i = 0; i < length; i ++)
{
filterCoeffs[i] = (SAMPLETYPE)(coeffs[i] * scale);
// create also stereo set of filter coefficients: this allows compiler
// to autovectorize filter evaluation much more efficiently
filterCoeffsStereo[2 * i] = (SAMPLETYPE)(coeffs[i] * scale);
filterCoeffsStereo[2 * i + 1] = (SAMPLETYPE)(coeffs[i] * scale);
}
} }
@ -254,7 +242,6 @@ uint FIRFilter::getLength() const
} }
// Applies the filter to the given sequence of samples. // Applies the filter to the given sequence of samples.
// //
// Note : The amount of outputted samples is by value of 'filter_length' // Note : The amount of outputted samples is by value of 'filter_length'
@ -284,10 +271,9 @@ uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSample
} }
// Operator 'new' is overloaded so that it automatically creates a suitable instance // Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not. // depending on if we've a MMX-capable CPU available or not.
void * FIRFilter::operator new(size_t s) void * FIRFilter::operator new(size_t)
{ {
// Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead! // Notice! don't use "new FIRFilter" directly, use "newInstance" to create a new instance instead!
ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!"); ST_THROW_RT_ERROR("Error in FIRFilter::new: Don't use 'new FIRFilter', use 'newInstance' member instead!");

16
Externals/soundtouch/FIRFilter.h vendored
View File

@ -11,13 +11,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-02-21 23:24:29 +0200 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -64,6 +57,7 @@ protected:
// Memory for filter coefficients // Memory for filter coefficients
SAMPLETYPE *filterCoeffs; SAMPLETYPE *filterCoeffs;
SAMPLETYPE *filterCoeffsStereo;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest, virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
@ -112,12 +106,12 @@ public:
short *filterCoeffsUnalign; short *filterCoeffsUnalign;
short *filterCoeffsAlign; short *filterCoeffsAlign;
virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const; virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const override;
public: public:
FIRFilterMMX(); FIRFilterMMX();
~FIRFilterMMX(); ~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor); virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor) override;
}; };
#endif // SOUNDTOUCH_ALLOW_MMX #endif // SOUNDTOUCH_ALLOW_MMX
@ -131,12 +125,12 @@ public:
float *filterCoeffsUnalign; float *filterCoeffsUnalign;
float *filterCoeffsAlign; float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const; virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const override;
public: public:
FIRFilterSSE(); FIRFilterSSE();
~FIRFilterSSE(); ~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor); virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor) override;
}; };
#endif // SOUNDTOUCH_ALLOW_SSE #endif // SOUNDTOUCH_ALLOW_SSE

4
Externals/soundtouch/InterpolateCubic.cpp vendored
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@ -8,10 +8,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library

18
Externals/soundtouch/InterpolateCubic.h vendored
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@ -8,10 +8,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// $Id: InterpolateCubic.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -45,21 +41,27 @@ namespace soundtouch
class InterpolateCubic : public TransposerBase class InterpolateCubic : public TransposerBase
{ {
protected: protected:
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest, virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
virtual int transposeStereo(SAMPLETYPE *dest, virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
virtual int transposeMulti(SAMPLETYPE *dest, virtual int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
double fract; double fract;
public: public:
InterpolateCubic(); InterpolateCubic();
virtual void resetRegisters() override;
virtual int getLatency() const override
{
return 1;
}
}; };
} }

6
Externals/soundtouch/InterpolateLinear.cpp vendored
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@ -8,10 +8,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// $Id: InterpolateLinear.cpp 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -146,7 +142,7 @@ int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE
LONG_SAMPLETYPE temp, vol1; LONG_SAMPLETYPE temp, vol1;
assert(iFract < SCALE); assert(iFract < SCALE);
vol1 = (SCALE - iFract); vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
for (int c = 0; c < numChannels; c ++) for (int c = 0; c < numChannels; c ++)
{ {
temp = vol1 * src[c] + iFract * src[c + numChannels]; temp = vol1 * src[c] + iFract * src[c + numChannels];

34
Externals/soundtouch/InterpolateLinear.h vendored
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@ -8,10 +8,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// $Id: InterpolateLinear.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -42,39 +38,42 @@
namespace soundtouch namespace soundtouch
{ {
/// Linear transposer class that uses integer arithmetics /// Linear transposer class that uses integer arithmetic
class InterpolateLinearInteger : public TransposerBase class InterpolateLinearInteger : public TransposerBase
{ {
protected: protected:
int iFract; int iFract;
int iRate; int iRate;
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest, virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
virtual int transposeStereo(SAMPLETYPE *dest, virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples); virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples) override;
public: public:
InterpolateLinearInteger(); InterpolateLinearInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower /// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates. /// rate, larger faster rates.
virtual void setRate(double newRate); virtual void setRate(double newRate) override;
virtual void resetRegisters() override;
virtual int getLatency() const override
{
return 0;
}
}; };
/// Linear transposer class that uses floating point arithmetics /// Linear transposer class that uses floating point arithmetic
class InterpolateLinearFloat : public TransposerBase class InterpolateLinearFloat : public TransposerBase
{ {
protected: protected:
double fract; double fract;
virtual void resetRegisters();
virtual int transposeMono(SAMPLETYPE *dest, virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples);
@ -85,6 +84,13 @@ protected:
public: public:
InterpolateLinearFloat(); InterpolateLinearFloat();
virtual void resetRegisters();
int getLatency() const
{
return 0;
}
}; };
} }

10
Externals/soundtouch/InterpolateShannon.cpp vendored
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@ -13,10 +13,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// $Id: InterpolateShannon.cpp 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -175,9 +171,9 @@ int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
/// Transpose stereo audio. Returns number of produced output samples, and /// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples /// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeMulti(SAMPLETYPE *pdest, int InterpolateShannon::transposeMulti(SAMPLETYPE *,
const SAMPLETYPE *psrc, const SAMPLETYPE *,
int &srcSamples) int &)
{ {
// not implemented // not implemented
assert(false); assert(false);

18
Externals/soundtouch/InterpolateShannon.h vendored
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@ -13,10 +13,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// $Id: InterpolateShannon.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -50,21 +46,27 @@ namespace soundtouch
class InterpolateShannon : public TransposerBase class InterpolateShannon : public TransposerBase
{ {
protected: protected:
void resetRegisters();
int transposeMono(SAMPLETYPE *dest, int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
int transposeStereo(SAMPLETYPE *dest, int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
int transposeMulti(SAMPLETYPE *dest, int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples); int &srcSamples) override;
double fract; double fract;
public: public:
InterpolateShannon(); InterpolateShannon();
void resetRegisters() override;
virtual int getLatency() const override
{
return 3;
}
}; };
} }

15
Externals/soundtouch/PeakFinder.cpp vendored
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@ -11,13 +11,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-05-18 18:22:02 +0300 (Mon, 18 May 2015) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.cpp 213 2015-05-18 15:22:02Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -149,7 +142,7 @@ int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, i
peaklevel = data[peakpos]; peaklevel = data[peakpos];
assert(peaklevel >= level); assert(peaklevel >= level);
pos = peakpos; pos = peakpos;
while ((pos >= minPos) && (pos < maxPos)) while ((pos >= minPos) && (pos + direction < maxPos))
{ {
if (data[pos + direction] < level) return pos; // crossing found if (data[pos + direction] < level) return pos; // crossing found
pos += direction; pos += direction;
@ -178,7 +171,6 @@ double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos)
} }
/// get exact center of peak near given position by calculating local mass of center /// get exact center of peak near given position by calculating local mass of center
double PeakFinder::getPeakCenter(const float *data, int peakpos) const double PeakFinder::getPeakCenter(const float *data, int peakpos) const
{ {
@ -218,7 +210,6 @@ double PeakFinder::getPeakCenter(const float *data, int peakpos) const
} }
double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos) double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
{ {
@ -249,12 +240,12 @@ double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
// - sometimes the highest peak can be Nth harmonic of the true base peak yet // - sometimes the highest peak can be Nth harmonic of the true base peak yet
// just a slightly higher than the true base // just a slightly higher than the true base
for (i = 3; i < 10; i ++) for (i = 1; i < 3; i ++)
{ {
double peaktmp, harmonic; double peaktmp, harmonic;
int i1,i2; int i1,i2;
harmonic = (double)i * 0.5; harmonic = (double)pow(2.0, i);
peakpos = (int)(highPeak / harmonic + 0.5f); peakpos = (int)(highPeak / harmonic + 0.5f);
if (peakpos < minPos) break; if (peakpos < minPos) break;
peakpos = findTop(data, peakpos); // seek true local maximum index peakpos = findTop(data, peakpos); // seek true local maximum index

11
Externals/soundtouch/PeakFinder.h vendored
View File

@ -9,13 +9,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2011-12-30 22:33:46 +0200 (Fri, 30 Dec 2011) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -51,8 +44,8 @@ protected:
/// Calculates the mass center between given vector items. /// Calculates the mass center between given vector items.
double calcMassCenter(const float *data, ///< Data vector. double calcMassCenter(const float *data, ///< Data vector.
int firstPos, ///< Index of first vector item beloging to the peak. int firstPos, ///< Index of first vector item belonging to the peak.
int lastPos ///< Index of last vector item beloging to the peak. int lastPos ///< Index of last vector item belonging to the peak.
) const; ) const;
/// Finds the data vector index where the monotoniously decreasing signal crosses the /// Finds the data vector index where the monotoniously decreasing signal crosses the

49
Externals/soundtouch/RateTransposer.cpp vendored
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@ -10,13 +10,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-07-26 17:45:48 +0300 (Sun, 26 Jul 2015) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -57,15 +50,21 @@ TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
// Constructor // Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer) RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{ {
bUseAAFilter = true; bUseAAFilter =
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
true;
#else
// Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
false;
#endif
// Instantiates the anti-alias filter // Instantiates the anti-alias filter
pAAFilter = new AAFilter(64); pAAFilter = new AAFilter(64);
pTransposer = TransposerBase::newInstance(); pTransposer = TransposerBase::newInstance();
clear();
} }
RateTransposer::~RateTransposer() RateTransposer::~RateTransposer()
{ {
delete pAAFilter; delete pAAFilter;
@ -73,11 +72,14 @@ RateTransposer::~RateTransposer()
} }
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(bool newMode) void RateTransposer::enableAAFilter(bool newMode)
{ {
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
// Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
bUseAAFilter = newMode; bUseAAFilter = newMode;
clear();
#endif
} }
@ -94,7 +96,6 @@ AAFilter *RateTransposer::getAAFilter()
} }
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower // Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates. // iRate, larger faster iRates.
void RateTransposer::setRate(double newRate) void RateTransposer::setRate(double newRate)
@ -130,8 +131,6 @@ void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
// the 'set_returnBuffer_size' function. // the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples) void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{ {
uint count;
if (nSamples == 0) return; if (nSamples == 0) return;
// Store samples to input buffer // Store samples to input buffer
@ -141,7 +140,7 @@ void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
// the filter // the filter
if (bUseAAFilter == false) if (bUseAAFilter == false)
{ {
count = pTransposer->transpose(outputBuffer, inputBuffer); (void)pTransposer->transpose(outputBuffer, inputBuffer);
return; return;
} }
@ -177,11 +176,10 @@ void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
// Sets the number of channels, 1 = mono, 2 = stereo // Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels) void RateTransposer::setChannels(int nChannels)
{ {
assert(nChannels > 0); if (!verifyNumberOfChannels(nChannels) ||
(pTransposer->numChannels == nChannels)) return;
if (pTransposer->numChannels == nChannels) return;
pTransposer->setChannels(nChannels); pTransposer->setChannels(nChannels);
inputBuffer.setChannels(nChannels); inputBuffer.setChannels(nChannels);
midBuffer.setChannels(nChannels); midBuffer.setChannels(nChannels);
outputBuffer.setChannels(nChannels); outputBuffer.setChannels(nChannels);
@ -194,6 +192,11 @@ void RateTransposer::clear()
outputBuffer.clear(); outputBuffer.clear();
midBuffer.clear(); midBuffer.clear();
inputBuffer.clear(); inputBuffer.clear();
pTransposer->resetRegisters();
// prefill buffer to avoid losing first samples at beginning of stream
int prefill = getLatency();
inputBuffer.addSilent(prefill);
} }
@ -208,6 +211,14 @@ int RateTransposer::isEmpty() const
} }
/// Return approximate initial input-output latency
int RateTransposer::getLatency() const
{
return pTransposer->getLatency() +
((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
}
////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////
// //
// TransposerBase - Base class for interpolation // TransposerBase - Base class for interpolation
@ -280,7 +291,7 @@ void TransposerBase::setRate(double newRate)
TransposerBase *TransposerBase::newInstance() TransposerBase *TransposerBase::newInstance()
{ {
#ifdef SOUNDTOUCH_INTEGER_SAMPLES #ifdef SOUNDTOUCH_INTEGER_SAMPLES
// Notice: For integer arithmetics support only linear algorithm (due to simplest calculus) // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
return ::new InterpolateLinearInteger; return ::new InterpolateLinearInteger;
#else #else
switch (algorithm) switch (algorithm)
@ -296,7 +307,7 @@ TransposerBase *TransposerBase::newInstance()
default: default:
assert(false); assert(false);
return NULL; return nullptr;
} }
#endif #endif
} }

35
Externals/soundtouch/RateTransposer.h vendored
View File

@ -14,13 +14,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-07-26 17:45:48 +0300 (Sun, 26 Jul 2015) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -66,8 +59,6 @@ public:
}; };
protected: protected:
virtual void resetRegisters() = 0;
virtual int transposeMono(SAMPLETYPE *dest, virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src, const SAMPLETYPE *src,
int &srcSamples) = 0; int &srcSamples) = 0;
@ -90,6 +81,9 @@ public:
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src); virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
virtual void setRate(double newRate); virtual void setRate(double newRate);
virtual void setChannels(int channels); virtual void setChannels(int channels);
virtual int getLatency() const = 0;
virtual void resetRegisters() = 0;
// static factory function // static factory function
static TransposerBase *newInstance(); static TransposerBase *newInstance();
@ -130,23 +124,11 @@ protected:
public: public:
RateTransposer(); RateTransposer();
virtual ~RateTransposer(); virtual ~RateTransposer() override;
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetics.
// static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// integer ot floating point arithmetics are to be used.
// static RateTransposer *newInstance();
/// Returns the output buffer object /// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; }; FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
// FIFOSamplePipe *getStore() { return &storeBuffer; };
/// Return anti-alias filter object /// Return anti-alias filter object
AAFilter *getAAFilter(); AAFilter *getAAFilter();
@ -165,13 +147,16 @@ public:
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into /// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. /// the input of the object.
void putSamples(const SAMPLETYPE *samples, uint numSamples); void putSamples(const SAMPLETYPE *samples, uint numSamples) override;
/// Clears all the samples in the object /// Clears all the samples in the object
void clear(); void clear() override;
/// Returns nonzero if there aren't any samples available for outputting. /// Returns nonzero if there aren't any samples available for outputting.
int isEmpty() const; int isEmpty() const override;
/// Return approximate initial input-output latency
int getLatency() const;
}; };
} }

35
Externals/soundtouch/STTypes.h vendored
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@ -8,13 +8,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-05-18 18:25:07 +0300 (Mon, 18 May 2015) $
// File revision : $Revision: 3 $
//
// $Id: STTypes.h 215 2015-05-18 15:25:07Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -62,6 +55,9 @@ namespace soundtouch
// Helper macro for aligning pointer up to next 16-byte boundary // Helper macro for aligning pointer up to next 16-byte boundary
#define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 ) #define SOUNDTOUCH_ALIGN_POINTER_16(x) ( ( (ulongptr)(x) + 15 ) & ~(ulongptr)15 )
/// Max allowed number of channels
#define SOUNDTOUCH_MAX_CHANNELS 16
/// Activate these undef's to overrule the possible sampletype /// Activate these undef's to overrule the possible sampletype
/// setting inherited from some other header file: /// setting inherited from some other header file:
#undef SOUNDTOUCH_INTEGER_SAMPLES #undef SOUNDTOUCH_INTEGER_SAMPLES
@ -74,7 +70,7 @@ namespace soundtouch
/// runtime performance so recommendation is to keep this off. /// runtime performance so recommendation is to keep this off.
// #define USE_MULTICH_ALWAYS // #define USE_MULTICH_ALWAYS
#if (defined(__SOFTFP__)) #if (defined(__SOFTFP__) && defined(ANDROID))
// For Android compilation: Force use of Integer samples in case that // For Android compilation: Force use of Integer samples in case that
// compilation uses soft-floating point emulation - soft-fp is way too slow // compilation uses soft-floating point emulation - soft-fp is way too slow
#undef SOUNDTOUCH_FLOAT_SAMPLES #undef SOUNDTOUCH_FLOAT_SAMPLES
@ -124,9 +120,9 @@ namespace soundtouch
#endif #endif
// If defined, allows the SIMD-optimized routines to take minor shortcuts // If defined, allows the SIMD-optimized routines to skip unevenly aligned
// for improved performance. Undefine to require faithfully similar SIMD // memory offsets that can cause performance penalty in some SIMD implementations.
// calculations as in normal C implementation. // Causes slight compromise in sound quality.
#define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1 #define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
@ -142,16 +138,19 @@ namespace soundtouch
#endif // SOUNDTOUCH_FLOAT_SAMPLES #endif // SOUNDTOUCH_FLOAT_SAMPLES
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow MMX optimizations // Allow MMX optimizations (not available in X64 mode)
#if (!_M_X64)
#define SOUNDTOUCH_ALLOW_MMX 1 #define SOUNDTOUCH_ALLOW_MMX 1
#endif #endif
#endif
#else #else
// floating point samples // floating point samples
typedef float SAMPLETYPE; typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision. // data type for sample accumulation: Use float also here to enable
typedef double LONG_SAMPLETYPE; // efficient autovectorization
typedef float LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS #ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations // Allow SSE optimizations
@ -160,7 +159,13 @@ namespace soundtouch
#endif // SOUNDTOUCH_INTEGER_SAMPLES #endif // SOUNDTOUCH_INTEGER_SAMPLES
}; #if ((SOUNDTOUCH_ALLOW_SSE) || (__SSE__) || (SOUNDTOUCH_USE_NEON))
#if SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
#define ST_SIMD_AVOID_UNALIGNED
#endif
#endif
}
// define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions: // define ST_NO_EXCEPTION_HANDLING switch to disable throwing std exceptions:
#define ST_NO_EXCEPTION_HANDLING 1 #define ST_NO_EXCEPTION_HANDLING 1

106
Externals/soundtouch/SoundTouch.cpp vendored
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@ -41,13 +41,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-07-26 17:45:48 +0300 (Sun, 26 Jul 2015) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.cpp 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -118,7 +111,6 @@ SoundTouch::SoundTouch()
} }
SoundTouch::~SoundTouch() SoundTouch::~SoundTouch()
{ {
delete pRateTransposer; delete pRateTransposer;
@ -126,7 +118,6 @@ SoundTouch::~SoundTouch()
} }
/// Get SoundTouch library version string /// Get SoundTouch library version string
const char *SoundTouch::getVersionString() const char *SoundTouch::getVersionString()
{ {
@ -146,18 +137,14 @@ uint SoundTouch::getVersionId()
// Sets the number of channels, 1 = mono, 2 = stereo // Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels) void SoundTouch::setChannels(uint numChannels)
{ {
/*if (numChannels != 1 && numChannels != 2) if (!verifyNumberOfChannels(numChannels)) return;
{
//ST_THROW_RT_ERROR("Illegal number of channels");
return;
}*/
channels = numChannels; channels = numChannels;
pRateTransposer->setChannels((int)numChannels); pRateTransposer->setChannels((int)numChannels);
pTDStretch->setChannels((int)numChannels); pTDStretch->setChannels((int)numChannels);
} }
// Sets new rate control value. Normal rate = 1.0, smaller values // Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates. // represent slower rate, larger faster rates.
void SoundTouch::setRate(double newRate) void SoundTouch::setRate(double newRate)
@ -167,7 +154,6 @@ void SoundTouch::setRate(double newRate)
} }
// Sets new rate control value as a difference in percents compared // Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %) // to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(double newRate) void SoundTouch::setRateChange(double newRate)
@ -177,7 +163,6 @@ void SoundTouch::setRateChange(double newRate)
} }
// Sets new tempo control value. Normal tempo = 1.0, smaller values // Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo. // represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(double newTempo) void SoundTouch::setTempo(double newTempo)
@ -187,7 +172,6 @@ void SoundTouch::setTempo(double newTempo)
} }
// Sets new tempo control value as a difference in percents compared // Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %) // to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(double newTempo) void SoundTouch::setTempoChange(double newTempo)
@ -197,7 +181,6 @@ void SoundTouch::setTempoChange(double newTempo)
} }
// Sets new pitch control value. Original pitch = 1.0, smaller values // Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch. // represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(double newPitch) void SoundTouch::setPitch(double newPitch)
@ -207,7 +190,6 @@ void SoundTouch::setPitch(double newPitch)
} }
// Sets pitch change in octaves compared to the original pitch // Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00) // (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(double newPitch) void SoundTouch::setPitchOctaves(double newPitch)
@ -217,7 +199,6 @@ void SoundTouch::setPitchOctaves(double newPitch)
} }
// Sets pitch change in semi-tones compared to the original pitch // Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12) // (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch) void SoundTouch::setPitchSemiTones(int newPitch)
@ -226,7 +207,6 @@ void SoundTouch::setPitchSemiTones(int newPitch)
} }
void SoundTouch::setPitchSemiTones(double newPitch) void SoundTouch::setPitchSemiTones(double newPitch)
{ {
setPitchOctaves(newPitch / 12.0); setPitchOctaves(newPitch / 12.0);
@ -286,9 +266,9 @@ void SoundTouch::calcEffectiveRateAndTempo()
// Sets sample rate. // Sets sample rate.
void SoundTouch::setSampleRate(uint srate) void SoundTouch::setSampleRate(uint srate)
{ {
bSrateSet = true;
// set sample rate, leave other tempo changer parameters as they are. // set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters((int)srate); pTDStretch->setParameters((int)srate);
bSrateSet = true;
} }
@ -305,22 +285,6 @@ void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined"); ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
} }
// Transpose the rate of the new samples if necessary
/* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
{
// yet flush the last samples in the pitch transposer buffer
// (may happen if 'rate' changes from a non-zero value to zero)
pTDStretch->moveSamples(*pRateTransposer);
}
pTDStretch->putSamples(samples, nSamples);
}
*/
// accumulate how many samples are expected out from processing, given the current // accumulate how many samples are expected out from processing, given the current
// processing setting // processing setting
samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo); samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
@ -359,6 +323,7 @@ void SoundTouch::flush()
// how many samples are still expected to output // how many samples are still expected to output
numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput); numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
if (numStillExpected < 0) numStillExpected = 0;
memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE)); memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by // "Push" the last active samples out from the processing pipeline by
@ -375,7 +340,6 @@ void SoundTouch::flush()
delete[] buff; delete[] buff;
// Clear input buffers // Clear input buffers
// pRateTransposer->clearInput();
pTDStretch->clearInput(); pTDStretch->clearInput();
// yet leave the output intouched as that's where the // yet leave the output intouched as that's where the
// flushed samples are! // flushed samples are!
@ -449,22 +413,62 @@ int SoundTouch::getSetting(int settingId) const
return (uint)pTDStretch->isQuickSeekEnabled(); return (uint)pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS: case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(NULL, &temp, NULL, NULL); pTDStretch->getParameters(nullptr, &temp, nullptr, nullptr);
return temp; return temp;
case SETTING_SEEKWINDOW_MS: case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(NULL, NULL, &temp, NULL); pTDStretch->getParameters(nullptr, nullptr, &temp, nullptr);
return temp; return temp;
case SETTING_OVERLAP_MS: case SETTING_OVERLAP_MS:
pTDStretch->getParameters(NULL, NULL, NULL, &temp); pTDStretch->getParameters(nullptr, nullptr, nullptr, &temp);
return temp; return temp;
case SETTING_NOMINAL_INPUT_SEQUENCE : case SETTING_NOMINAL_INPUT_SEQUENCE :
return pTDStretch->getInputSampleReq(); {
int size = pTDStretch->getInputSampleReq();
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0)
{
// transposing done before timestretch, which impacts latency
return (int)(size * rate + 0.5);
}
#endif
return size;
}
case SETTING_NOMINAL_OUTPUT_SEQUENCE : case SETTING_NOMINAL_OUTPUT_SEQUENCE :
return pTDStretch->getOutputBatchSize(); {
int size = pTDStretch->getOutputBatchSize();
if (rate > 1.0)
{
// transposing done after timestretch, which impacts latency
return (int)(size / rate + 0.5);
}
return size;
}
case SETTING_INITIAL_LATENCY:
{
double latency = pTDStretch->getLatency();
int latency_tr = pRateTransposer->getLatency();
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0)
{
// transposing done before timestretch, which impacts latency
latency = (latency + latency_tr) * rate;
}
else
#endif
{
latency += (double)latency_tr / rate;
}
return (int)(latency + 0.5);
}
default : default :
return 0; return 0;
@ -477,12 +481,12 @@ int SoundTouch::getSetting(int settingId) const
void SoundTouch::clear() void SoundTouch::clear()
{ {
samplesExpectedOut = 0; samplesExpectedOut = 0;
samplesOutput = 0;
pRateTransposer->clear(); pRateTransposer->clear();
pTDStretch->clear(); pTDStretch->clear();
} }
/// Returns number of samples currently unprocessed. /// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const uint SoundTouch::numUnprocessedSamples() const
{ {
@ -499,7 +503,6 @@ uint SoundTouch::numUnprocessedSamples() const
} }
/// Output samples from beginning of the sample buffer. Copies requested samples to /// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than /// output buffer and removes them from the sample buffer. If there are less than
/// 'numsample' samples in the buffer, returns all that available. /// 'numsample' samples in the buffer, returns all that available.
@ -524,3 +527,12 @@ uint SoundTouch::receiveSamples(uint maxSamples)
samplesOutput += (long)ret; samplesOutput += (long)ret;
return ret; return ret;
} }
/// Get ratio between input and output audio durations, useful for calculating
/// processed output duration: if you'll process a stream of N samples, then
/// you can expect to get out N * getInputOutputSampleRatio() samples.
double SoundTouch::getInputOutputSampleRatio()
{
return 1.0 / (tempo * rate);
}

95
Externals/soundtouch/SoundTouch.h vendored
View File

@ -41,13 +41,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-09-20 10:38:32 +0300 (Sun, 20 Sep 2015) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.h 230 2015-09-20 07:38:32Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -79,10 +72,10 @@ namespace soundtouch
{ {
/// Soundtouch library version string /// Soundtouch library version string
#define SOUNDTOUCH_VERSION "1.9.2" #define SOUNDTOUCH_VERSION "2.3.2"
/// SoundTouch library version id /// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (10902) #define SOUNDTOUCH_VERSION_ID (20302)
// //
// Available setting IDs for the 'setSetting' & 'get_setting' functions: // Available setting IDs for the 'setSetting' & 'get_setting' functions:
@ -116,15 +109,17 @@ namespace soundtouch
#define SETTING_OVERLAP_MS 5 #define SETTING_OVERLAP_MS 5
/// Call "getSetting" with this ID to query nominal average processing sequence /// Call "getSetting" with this ID to query processing sequence size in samples.
/// size in samples. This value tells approcimate value how many input samples /// This value gives approximate value of how many input samples you'll need to
/// SoundTouch needs to gather before it does DSP processing run for the sample batch. /// feed into SoundTouch after initial buffering to get out a new batch of
/// output samples.
///
/// This value does not include initial buffering at beginning of a new processing
/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size.
/// ///
/// Notices: /// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter /// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch /// - This parameter value is not constant but change depending on
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// tempo/pitch/rate/samplerate settings. /// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_INPUT_SEQUENCE 6 #define SETTING_NOMINAL_INPUT_SEQUENCE 6
@ -135,12 +130,41 @@ namespace soundtouch
/// ///
/// Notices: /// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter /// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch /// - This parameter value is not constant but change depending on
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// tempo/pitch/rate/samplerate settings. /// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7 #define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
/// Call "getSetting" with this ID to query initial processing latency, i.e.
/// approx. how many samples you'll need to enter to SoundTouch pipeline before
/// you can expect to get first batch of ready output samples out.
///
/// After the first output batch, you can then expect to get approx.
/// SETTING_NOMINAL_OUTPUT_SEQUENCE ready samples out for every
/// SETTING_NOMINAL_INPUT_SEQUENCE samples that you enter into SoundTouch.
///
/// Example:
/// processing with parameter -tempo=5
/// => initial latency = 5509 samples
/// input sequence = 4167 samples
/// output sequence = 3969 samples
///
/// Accordingly, you can expect to feed in approx. 5509 samples at beginning of
/// the stream, and then you'll get out the first 3969 samples. After that, for
/// every approx. 4167 samples that you'll put in, you'll receive again approx.
/// 3969 samples out.
///
/// This also means that average latency during stream processing is
/// INITIAL_LATENCY-OUTPUT_SEQUENCE/2, in the above example case 5509-3969/2
/// = 3524 samples
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - This parameter value is not constant but change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_INITIAL_LATENCY 8
class SoundTouch : public FIFOProcessor class SoundTouch : public FIFOProcessor
{ {
private: private:
@ -185,7 +209,7 @@ protected :
public: public:
SoundTouch(); SoundTouch();
virtual ~SoundTouch(); virtual ~SoundTouch() override;
/// Get SoundTouch library version string /// Get SoundTouch library version string
static const char *getVersionString(); static const char *getVersionString();
@ -228,6 +252,24 @@ public:
/// Sets sample rate. /// Sets sample rate.
void setSampleRate(uint srate); void setSampleRate(uint srate);
/// Get ratio between input and output audio durations, useful for calculating
/// processed output duration: if you'll process a stream of N samples, then
/// you can expect to get out N * getInputOutputSampleRatio() samples.
///
/// This ratio will give accurate target duration ratio for a full audio track,
/// given that the the whole track is processed with same processing parameters.
///
/// If this ratio is applied to calculate intermediate offsets inside a processing
/// stream, then this ratio is approximate and can deviate +- some tens of milliseconds
/// from ideal offset, yet by end of the audio stream the duration ratio will become
/// exact.
///
/// Example: if processing with parameters "-tempo=15 -pitch=-3", the function
/// will return value 0.8695652... Now, if processing an audio stream whose duration
/// is exactly one million audio samples, then you can expect the processed
/// output duration be 0.869565 * 1000000 = 869565 samples.
double getInputOutputSampleRatio();
/// Flushes the last samples from the processing pipeline to the output. /// Flushes the last samples from the processing pipeline to the output.
/// Clears also the internal processing buffers. /// Clears also the internal processing buffers.
// //
@ -245,7 +287,7 @@ public:
uint numSamples ///< Number of samples in buffer. Notice uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample ///< that in case of stereo-sound a single sample
///< contains data for both channels. ///< contains data for both channels.
); ) override;
/// Output samples from beginning of the sample buffer. Copies requested samples to /// Output samples from beginning of the sample buffer. Copies requested samples to
/// output buffer and removes them from the sample buffer. If there are less than /// output buffer and removes them from the sample buffer. If there are less than
@ -254,7 +296,7 @@ public:
/// \return Number of samples returned. /// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples. virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max. uint maxSamples ///< How many samples to receive at max.
); ) override;
/// Adjusts book-keeping so that given number of samples are removed from beginning of the /// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere. /// sample buffer without copying them anywhere.
@ -262,16 +304,16 @@ public:
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function. /// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe. virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
); ) override;
/// Clears all the samples in the object's output and internal processing /// Clears all the samples in the object's output and internal processing
/// buffers. /// buffers.
virtual void clear(); virtual void clear() override;
/// Changes a setting controlling the processing system behaviour. See the /// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's. /// 'SETTING_...' defines for available setting ID's.
/// ///
/// \return 'true' if the setting was succesfully changed /// \return 'true' if the setting was successfully changed
bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines. bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value. int value ///< New setting value.
); );
@ -286,6 +328,11 @@ public:
/// Returns number of samples currently unprocessed. /// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const; virtual uint numUnprocessedSamples() const;
/// Return number of channels
uint numChannels() const
{
return channels;
}
/// Other handy functions that are implemented in the ancestor classes (see /// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe') /// classes 'FIFOProcessor' and 'FIFOSamplePipe')

247
Externals/soundtouch/TDStretch.cpp vendored
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@ -1,11 +1,17 @@
//////////////////////////////////////////////////////////////////////////////// ///////////////////////////////////////////////////////////////////////////////
/// ///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo /// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like /// while maintaining the original pitch by using a time domain WSOLA-like
/// method with several performance-increasing tweaks. /// method with several performance-increasing tweaks.
/// ///
/// Note : MMX optimized functions reside in a separate, platform-specific /// Notes : MMX optimized functions reside in a separate, platform-specific
/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp' /// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'.
///
/// This source file contains OpenMP optimizations that allow speeding up the
/// corss-correlation algorithm by executing it in several threads / CPU cores
/// in parallel. See the following article link for more detailed discussion
/// about SoundTouch OpenMP optimizations:
/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices
/// ///
/// Author : Copyright (c) Olli Parviainen /// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi /// Author e-mail : oparviai 'at' iki.fi
@ -13,13 +19,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 1.12 $
//
// $Id: TDStretch.cpp 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -55,26 +54,6 @@ using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y)) #define max(x, y) (((x) > (y)) ? (x) : (y))
/*****************************************************************************
*
* Constant definitions
*
*****************************************************************************/
// Table for the hierarchical mixing position seeking algorithm
const short _scanOffsets[5][24]={
{ 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806,
868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0},
{-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
{ 121, 114, 97, 114, 98, 105, 108, 32, 104, 99, 117, 111,
116, 100, 110, 117, 111, 115, 0, 0, 0, 0, 0, 0}};
/***************************************************************************** /*****************************************************************************
* *
* Implementation of the class 'TDStretch' * Implementation of the class 'TDStretch'
@ -87,18 +66,13 @@ TDStretch::TDStretch() : FIFOProcessor(&outputBuffer)
bQuickSeek = false; bQuickSeek = false;
channels = 2; channels = 2;
pMidBuffer = NULL; pMidBuffer = nullptr;
pMidBufferUnaligned = NULL; pMidBufferUnaligned = nullptr;
overlapLength = 0; overlapLength = 0;
bAutoSeqSetting = true; bAutoSeqSetting = true;
bAutoSeekSetting = true; bAutoSeekSetting = true;
maxnorm = 0;
maxnormf = 1e8;
skipFract = 0;
tempo = 1.0f; tempo = 1.0f;
setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS); setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS);
setTempo(1.0f); setTempo(1.0f);
@ -128,7 +102,12 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
int aSeekWindowMS, int aOverlapMS) int aSeekWindowMS, int aOverlapMS)
{ {
// accept only positive parameter values - if zero or negative, use old values instead // accept only positive parameter values - if zero or negative, use old values instead
if (aSampleRate > 0) this->sampleRate = aSampleRate; if (aSampleRate > 0)
{
if (aSampleRate > 192000) ST_THROW_RT_ERROR("Error: Excessive samplerate");
this->sampleRate = aSampleRate;
}
if (aOverlapMS > 0) this->overlapMs = aOverlapMS; if (aOverlapMS > 0) this->overlapMs = aOverlapMS;
if (aSequenceMS > 0) if (aSequenceMS > 0)
@ -164,7 +143,7 @@ void TDStretch::setParameters(int aSampleRate, int aSequenceMS,
/// Get routine control parameters, see setParameters() function. /// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter /// Any of the parameters to this function can be nullptr, in such case corresponding parameter
/// value isn't returned. /// value isn't returned.
void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const void TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const
{ {
@ -219,6 +198,10 @@ void TDStretch::clearInput()
{ {
inputBuffer.clear(); inputBuffer.clear();
clearMidBuffer(); clearMidBuffer();
isBeginning = true;
maxnorm = 0;
maxnormf = 1e8;
skipFract = 0;
} }
@ -297,21 +280,23 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
int i; int i;
double norm; double norm;
bestCorr = FLT_MIN; bestCorr = -FLT_MAX;
bestOffs = 0; bestOffs = 0;
// Scans for the best correlation value by testing each possible position // Scans for the best correlation value by testing each possible position
// over the permitted range. // over the permitted range.
bestCorr = calcCrossCorr(refPos, pMidBuffer, norm); bestCorr = calcCrossCorr(refPos, pMidBuffer, norm);
bestCorr = (bestCorr + 0.1) * 0.75;
#pragma omp parallel for #pragma omp parallel for
for (i = 1; i < seekLength; i ++) for (i = 1; i < seekLength; i ++)
{ {
double corr; double corr;
// Calculates correlation value for the mixing position corresponding to 'i' // Calculates correlation value for the mixing position corresponding to 'i'
#ifdef _OPENMP #if defined(_OPENMP) || defined(ST_SIMD_AVOID_UNALIGNED)
// in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't // in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't
// iterate the loop in sequential order // iterate the loop in sequential order
// in SIMD mode, avoid accumulator version to allow avoiding unaligned positions
corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm); corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm);
#else #else
// In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same // In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same
@ -354,7 +339,7 @@ int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos)
// with improved precision // with improved precision
// //
// Based on testing: // Based on testing:
// - This algorithm gives on average 99% as good match as the full algorith // - This algorithm gives on average 99% as good match as the full algorithm
// - this quick seek algorithm finds the best match on ~90% of cases // - this quick seek algorithm finds the best match on ~90% of cases
// - on those 10% of cases when this algorithm doesn't find best match, // - on those 10% of cases when this algorithm doesn't find best match,
// it still finds on average ~90% match vs. the best possible match // it still finds on average ~90% match vs. the best possible match
@ -373,12 +358,10 @@ int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
// note: 'float' types used in this function in case that the platform would need to use software-fp // note: 'float' types used in this function in case that the platform would need to use software-fp
bestCorr = FLT_MIN; bestCorr =
bestOffs = SCANWIND; bestCorr2 = -FLT_MAX;
bestCorr2 = FLT_MIN; bestOffs =
bestOffs2 = 0; bestOffs2 = SCANWIND;
int best = 0;
// Scans for the best correlation value by testing each possible position // Scans for the best correlation value by testing each possible position
// over the permitted range. Look for two best matches on the first pass to // over the permitted range. Look for two best matches on the first pass to
@ -436,7 +419,6 @@ int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
{ {
bestCorr = corr; bestCorr = corr;
bestOffs = i; bestOffs = i;
best = 1;
} }
} }
@ -458,7 +440,6 @@ int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos)
{ {
bestCorr = corr; bestCorr = corr;
bestOffs = i; bestOffs = i;
best = 2;
} }
} }
@ -515,18 +496,18 @@ void TDStretch::clearCrossCorrState()
void TDStretch::calcSeqParameters() void TDStretch::calcSeqParameters()
{ {
// Adjust tempo param according to tempo, so that variating processing sequence length is used // Adjust tempo param according to tempo, so that variating processing sequence length is used
// at varius tempo settings, between the given low...top limits // at various tempo settings, between the given low...top limits
#define AUTOSEQ_TEMPO_LOW 0.5 // auto setting low tempo range (-50%) #define AUTOSEQ_TEMPO_LOW 0.5 // auto setting low tempo range (-50%)
#define AUTOSEQ_TEMPO_TOP 2.0 // auto setting top tempo range (+100%) #define AUTOSEQ_TEMPO_TOP 2.0 // auto setting top tempo range (+100%)
// sequence-ms setting values at above low & top tempo // sequence-ms setting values at above low & top tempo
#define AUTOSEQ_AT_MIN 125.0 #define AUTOSEQ_AT_MIN 90.0
#define AUTOSEQ_AT_MAX 50.0 #define AUTOSEQ_AT_MAX 40.0
#define AUTOSEQ_K ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW)) #define AUTOSEQ_K ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
#define AUTOSEQ_C (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW)) #define AUTOSEQ_C (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW))
// seek-window-ms setting values at above low & top tempoq // seek-window-ms setting values at above low & top tempoq
#define AUTOSEEK_AT_MIN 25.0 #define AUTOSEEK_AT_MIN 20.0
#define AUTOSEEK_AT_MAX 15.0 #define AUTOSEEK_AT_MAX 15.0
#define AUTOSEEK_K ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW)) #define AUTOSEEK_K ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW))
#define AUTOSEEK_C (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW)) #define AUTOSEEK_C (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW))
@ -586,9 +567,8 @@ void TDStretch::setTempo(double newTempo)
// Sets the number of channels, 1 = mono, 2 = stereo // Sets the number of channels, 1 = mono, 2 = stereo
void TDStretch::setChannels(int numChannels) void TDStretch::setChannels(int numChannels)
{ {
assert(numChannels > 0); if (!verifyNumberOfChannels(numChannels) ||
if (channels == numChannels) return; (channels == numChannels)) return;
// assert(numChannels == 1 || numChannels == 2);
channels = numChannels; channels = numChannels;
inputBuffer.setChannels(channels); inputBuffer.setChannels(channels);
@ -637,7 +617,8 @@ void TDStretch::processNominalTempo()
// the result into 'outputBuffer' // the result into 'outputBuffer'
void TDStretch::processSamples() void TDStretch::processSamples()
{ {
int ovlSkip, offset; int ovlSkip;
int offset = 0;
int temp; int temp;
/* Removed this small optimization - can introduce a click to sound when tempo setting /* Removed this small optimization - can introduce a click to sound when tempo setting
@ -654,8 +635,10 @@ void TDStretch::processSamples()
// to form a processing frame. // to form a processing frame.
while ((int)inputBuffer.numSamples() >= sampleReq) while ((int)inputBuffer.numSamples() >= sampleReq)
{ {
// If tempo differs from the normal ('SCALE'), scan for the best overlapping if (isBeginning == false)
// position {
// apart from the very beginning of the track,
// scan for the best overlapping position & do overlap-add
offset = seekBestOverlapPosition(inputBuffer.ptrBegin()); offset = seekBestOverlapPosition(inputBuffer.ptrBegin());
// Mix the samples in the 'inputBuffer' at position of 'offset' with the // Mix the samples in the 'inputBuffer' at position of 'offset' with the
@ -664,25 +647,50 @@ void TDStretch::processSamples()
// (that's in 'midBuffer') // (that's in 'midBuffer')
overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset); overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset);
outputBuffer.putSamples((uint)overlapLength); outputBuffer.putSamples((uint)overlapLength);
offset += overlapLength;
}
else
{
// Adjust processing offset at beginning of track by not perform initial overlapping
// and compensating that in the 'input buffer skip' calculation
isBeginning = false;
int skip = (int)(tempo * overlapLength + 0.5 * seekLength + 0.5);
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode, round the skip amount to value corresponding to aligned memory address
if (channels == 1)
{
skip &= -4;
}
else if (channels == 2)
{
skip &= -2;
}
#endif
skipFract -= skip;
if (skipFract <= -nominalSkip)
{
skipFract = -nominalSkip;
}
}
// ... then copy sequence samples from 'inputBuffer' to output: // ... then copy sequence samples from 'inputBuffer' to output:
// length of sequence
temp = (seekWindowLength - 2 * overlapLength);
// crosscheck that we don't have buffer overflow... // crosscheck that we don't have buffer overflow...
if ((int)inputBuffer.numSamples() < (offset + temp + overlapLength * 2)) if ((int)inputBuffer.numSamples() < (offset + seekWindowLength - overlapLength))
{ {
continue; // just in case, shouldn't really happen continue; // just in case, shouldn't really happen
} }
outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * (offset + overlapLength), (uint)temp); // length of sequence
temp = (seekWindowLength - 2 * overlapLength);
outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * offset, (uint)temp);
// Copies the end of the current sequence from 'inputBuffer' to // Copies the end of the current sequence from 'inputBuffer' to
// 'midBuffer' for being mixed with the beginning of the next // 'midBuffer' for being mixed with the beginning of the next
// processing sequence and so on // processing sequence and so on
assert((offset + temp + overlapLength * 2) <= (int)inputBuffer.numSamples()); assert((offset + temp + overlapLength) <= (int)inputBuffer.numSamples());
memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp + overlapLength), memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp),
channels * sizeof(SAMPLETYPE) * overlapLength); channels * sizeof(SAMPLETYPE) * overlapLength);
// Remove the processed samples from the input buffer. Update // Remove the processed samples from the input buffer. Update
@ -732,7 +740,7 @@ void TDStretch::acceptNewOverlapLength(int newOverlapLength)
// Operator 'new' is overloaded so that it automatically creates a suitable instance // Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX/SSE/etc-capable CPU available or not. // depending on if we've a MMX/SSE/etc-capable CPU available or not.
void * TDStretch::operator new(size_t s) void * TDStretch::operator new(size_t)
{ {
// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead! // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!"); ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!");
@ -776,7 +784,7 @@ TDStretch * TDStretch::newInstance()
////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////
// //
// Integer arithmetics specific algorithm implementations. // Integer arithmetic specific algorithm implementations.
// //
////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////
@ -802,21 +810,19 @@ void TDStretch::overlapStereo(short *poutput, const short *input) const
// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi' // Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi'
// version of the routine. // version of the routine.
void TDStretch::overlapMulti(SAMPLETYPE *poutput, const SAMPLETYPE *input) const void TDStretch::overlapMulti(short *poutput, const short *input) const
{ {
SAMPLETYPE m1=(SAMPLETYPE)0; short m1;
SAMPLETYPE m2;
int i = 0; int i = 0;
for (m2 = (SAMPLETYPE)overlapLength; m2; m2 --) for (m1 = 0; m1 < overlapLength; m1 ++)
{ {
short m2 = (short)(overlapLength - m1);
for (int c = 0; c < channels; c ++) for (int c = 0; c < channels; c ++)
{ {
poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength; poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength;
i++; i++;
} }
m1++;
} }
} }
@ -861,26 +867,34 @@ double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, do
unsigned long lnorm; unsigned long lnorm;
int i; int i;
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
if (((ulongptr)mixingPos) & 15) return -1e50;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
corr = lnorm = 0; corr = lnorm = 0;
// Same routine for stereo and mono. For stereo, unroll loop for better // Same routine for stereo and mono
// efficiency and gives slightly better resolution against rounding. for (i = 0; i < ilength; i += 2)
// For mono it same routine, just unrolls loop by factor of 4
for (i = 0; i < channels * overlapLength; i += 4)
{ {
corr += (mixingPos[i] * compare[i] + corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm; // notice: do intermediate division here to avoid integer overflow mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
corr += (mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBitsNorm;
lnorm += (mixingPos[i] * mixingPos[i] + lnorm += (mixingPos[i] * mixingPos[i] +
mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBitsNorm; // notice: do intermediate division here to avoid integer overflow mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBitsNorm;
lnorm += (mixingPos[i + 2] * mixingPos[i + 2] + // do intermediate scalings to avoid integer overflow
mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBitsNorm;
} }
if (lnorm > maxnorm)
{
// modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
#pragma omp critical
if (lnorm > maxnorm) if (lnorm > maxnorm)
{ {
maxnorm = lnorm; maxnorm = lnorm;
} }
}
// Normalize result by dividing by sqrt(norm) - this step is easiest // Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation // done using floating point operation
norm = (double)lnorm; norm = (double)lnorm;
@ -892,9 +906,12 @@ double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, do
double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm)
{ {
long corr; long corr;
unsigned long lnorm; long lnorm;
int i; int i;
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
// cancel first normalizer tap from previous round // cancel first normalizer tap from previous round
lnorm = 0; lnorm = 0;
for (i = 1; i <= channels; i ++) for (i = 1; i <= channels; i ++)
@ -903,15 +920,11 @@ double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *c
} }
corr = 0; corr = 0;
// Same routine for stereo and mono. For stereo, unroll loop for better // Same routine for stereo and mono.
// efficiency and gives slightly better resolution against rounding. for (i = 0; i < ilength; i += 2)
// For mono it same routine, just unrolls loop by factor of 4
for (i = 0; i < channels * overlapLength; i += 4)
{ {
corr += (mixingPos[i] * compare[i] + corr += (mixingPos[i] * compare[i] +
mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm; // notice: do intermediate division here to avoid integer overflow mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm;
corr += (mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBitsNorm;
} }
// update normalizer with last samples of this round // update normalizer with last samples of this round
@ -936,7 +949,7 @@ double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *c
////////////////////////////////////////////////////////////////////////////// //////////////////////////////////////////////////////////////////////////////
// //
// Floating point arithmetics specific algorithm implementations. // Floating point arithmetic specific algorithm implementations.
// //
#ifdef SOUNDTOUCH_FLOAT_SAMPLES #ifdef SOUNDTOUCH_FLOAT_SAMPLES
@ -1012,27 +1025,24 @@ void TDStretch::calculateOverlapLength(int overlapInMsec)
/// Calculate cross-correlation /// Calculate cross-correlation
double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm) double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm)
{ {
double corr; float corr;
double norm; float norm;
int i; int i;
#ifdef ST_SIMD_AVOID_UNALIGNED
// in SIMD mode skip 'mixingPos' positions that aren't aligned to 16-byte boundary
if (((ulongptr)mixingPos) & 15) return -1e50;
#endif
// hint compiler autovectorization that loop length is divisible by 8
int ilength = (channels * overlapLength) & -8;
corr = norm = 0; corr = norm = 0;
// Same routine for stereo and mono. For Stereo, unroll by factor of 2. // Same routine for stereo and mono
// For mono it's same routine yet unrollsd by factor of 4. for (i = 0; i < ilength; i ++)
for (i = 0; i < channels * overlapLength; i += 4)
{ {
corr += mixingPos[i] * compare[i] + corr += mixingPos[i] * compare[i];
mixingPos[i + 1] * compare[i + 1]; norm += mixingPos[i] * mixingPos[i];
norm += mixingPos[i] * mixingPos[i] +
mixingPos[i + 1] * mixingPos[i + 1];
// unroll the loop for better CPU efficiency:
corr += mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3];
norm += mixingPos[i + 2] * mixingPos[i + 2] +
mixingPos[i + 3] * mixingPos[i + 3];
} }
anorm = norm; anorm = norm;
@ -1043,7 +1053,7 @@ double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, do
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value /// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm)
{ {
double corr; float corr;
int i; int i;
corr = 0; corr = 0;
@ -1054,14 +1064,13 @@ double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *c
norm -= mixingPos[-i] * mixingPos[-i]; norm -= mixingPos[-i] * mixingPos[-i];
} }
// Same routine for stereo and mono. For Stereo, unroll by factor of 2. // hint compiler autovectorization that loop length is divisible by 8
// For mono it's same routine yet unrollsd by factor of 4. int ilength = (channels * overlapLength) & -8;
for (i = 0; i < channels * overlapLength; i += 4)
// Same routine for stereo and mono
for (i = 0; i < ilength; i ++)
{ {
corr += mixingPos[i] * compare[i] + corr += mixingPos[i] * compare[i];
mixingPos[i + 1] * compare[i + 1] +
mixingPos[i + 2] * compare[i + 2] +
mixingPos[i + 3] * compare[i + 3];
} }
// update normalizer with last samples of this round // update normalizer with last samples of this round

36
Externals/soundtouch/TDStretch.h vendored
View File

@ -13,13 +13,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.h 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -134,6 +127,7 @@ protected:
bool bQuickSeek; bool bQuickSeek;
bool bAutoSeqSetting; bool bAutoSeqSetting;
bool bAutoSeekSetting; bool bAutoSeekSetting;
bool isBeginning;
SAMPLETYPE *pMidBuffer; SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pMidBufferUnaligned; SAMPLETYPE *pMidBufferUnaligned;
@ -163,7 +157,6 @@ protected:
void calcSeqParameters(); void calcSeqParameters();
void adaptNormalizer(); void adaptNormalizer();
/// Changes the tempo of the given sound samples. /// Changes the tempo of the given sound samples.
/// Returns amount of samples returned in the "output" buffer. /// Returns amount of samples returned in the "output" buffer.
/// The maximum amount of samples that can be returned at a time is set by /// The maximum amount of samples that can be returned at a time is set by
@ -172,7 +165,7 @@ protected:
public: public:
TDStretch(); TDStretch();
virtual ~TDStretch(); virtual ~TDStretch() override;
/// Operator 'new' is overloaded so that it automatically creates a suitable instance /// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX/SSE/etc-capable CPU available or not. /// depending on if we've a MMX/SSE/etc-capable CPU available or not.
@ -194,7 +187,7 @@ public:
void setTempo(double newTempo); void setTempo(double newTempo);
/// Returns nonzero if there aren't any samples available for outputting. /// Returns nonzero if there aren't any samples available for outputting.
virtual void clear(); virtual void clear() override;
/// Clears the input buffer /// Clears the input buffer
void clearInput(); void clearInput();
@ -224,7 +217,7 @@ public:
); );
/// Get routine control parameters, see setParameters() function. /// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be NULL, in such case corresponding parameter /// Any of the parameters to this function can be nullptr, in such case corresponding parameter
/// value isn't returned. /// value isn't returned.
void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const; void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
@ -234,7 +227,7 @@ public:
const SAMPLETYPE *samples, ///< Input sample data const SAMPLETYPE *samples, ///< Input sample data
uint numSamples ///< Number of samples in 'samples' so that one sample uint numSamples ///< Number of samples in 'samples' so that one sample
///< contains both channels if stereo ///< contains both channels if stereo
); ) override;
/// return nominal input sample requirement for triggering a processing batch /// return nominal input sample requirement for triggering a processing batch
int getInputSampleReq() const int getInputSampleReq() const
@ -247,8 +240,13 @@ public:
{ {
return seekWindowLength - overlapLength; return seekWindowLength - overlapLength;
} }
};
/// return approximate initial input-output latency
int getLatency() const
{
return sampleReq;
}
};
// Implementation-specific class declarations: // Implementation-specific class declarations:
@ -258,10 +256,10 @@ public:
class TDStretchMMX : public TDStretch class TDStretchMMX : public TDStretch
{ {
protected: protected:
double calcCrossCorr(const short *mixingPos, const short *compare, double &norm); double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) override;
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm); double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) override;
virtual void overlapStereo(short *output, const short *input) const; virtual void overlapStereo(short *output, const short *input) const override;
virtual void clearCrossCorrState(); virtual void clearCrossCorrState() override;
}; };
#endif /// SOUNDTOUCH_ALLOW_MMX #endif /// SOUNDTOUCH_ALLOW_MMX
@ -271,8 +269,8 @@ public:
class TDStretchSSE : public TDStretch class TDStretchSSE : public TDStretch
{ {
protected: protected:
double calcCrossCorr(const float *mixingPos, const float *compare, double &norm); double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) override;
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm); double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) override;
}; };
#endif /// SOUNDTOUCH_ALLOW_SSE #endif /// SOUNDTOUCH_ALLOW_SSE

7
Externals/soundtouch/cpu_detect.h vendored
View File

@ -12,13 +12,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library

8
Externals/soundtouch/cpu_detect_x86.cpp vendored
View File

@ -11,13 +11,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2014-01-07 20:24:28 +0200 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86.cpp 183 2014-01-07 18:24:28Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -75,7 +68,6 @@ void disableExtensions(uint dwDisableMask)
} }
/// Checks which instruction set extensions are supported by the CPU. /// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void) uint detectCPUextensions(void)
{ {

23
Externals/soundtouch/mmx_optimized.cpp vendored
View File

@ -20,13 +20,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 4 $
//
// $Id: mmx_optimized.cpp 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -123,10 +116,15 @@ double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &d
// Clear MMS state // Clear MMS state
_m_empty(); _m_empty();
if (norm > (long)maxnorm)
{
// modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode
#pragma omp critical
if (norm > (long)maxnorm) if (norm > (long)maxnorm)
{ {
maxnorm = norm; maxnorm = norm;
} }
}
// Normalize result by dividing by sqrt(norm) - this step is easiest // Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation // done using floating point operation
@ -219,7 +217,6 @@ void TDStretchMMX::clearCrossCorrState()
} }
// MMX-optimized version of the function overlapStereo // MMX-optimized version of the function overlapStereo
void TDStretchMMX::overlapStereo(short *output, const short *input) const void TDStretchMMX::overlapStereo(short *output, const short *input) const
{ {
@ -297,8 +294,8 @@ void TDStretchMMX::overlapStereo(short *output, const short *input) const
FIRFilterMMX::FIRFilterMMX() : FIRFilter() FIRFilterMMX::FIRFilterMMX() : FIRFilter()
{ {
filterCoeffsAlign = NULL; filterCoeffsAlign = nullptr;
filterCoeffsUnalign = NULL; filterCoeffsUnalign = nullptr;
} }
@ -335,7 +332,6 @@ void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uRe
} }
// mmx-optimized version of the filter routine for stereo sound // mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
{ {
@ -392,4 +388,9 @@ uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numS
return (numSamples & 0xfffffffe) - length; return (numSamples & 0xfffffffe) - length;
} }
#else
// workaround to not complain about empty module
bool _dontcomplain_mmx_empty;
#endif // SOUNDTOUCH_ALLOW_MMX #endif // SOUNDTOUCH_ALLOW_MMX

23
Externals/soundtouch/sse_optimized.cpp vendored
View File

@ -23,13 +23,6 @@
/// ///
//////////////////////////////////////////////////////////////////////////////// ////////////////////////////////////////////////////////////////////////////////
// //
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 4 $
//
// $Id: sse_optimized.cpp 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License : // License :
// //
// SoundTouch audio processing library // SoundTouch audio processing library
@ -87,7 +80,7 @@ double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &a
// Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided // Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
// for choosing if this little cheating is allowed. // for choosing if this little cheating is allowed.
#ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION #ifdef ST_SIMD_AVOID_UNALIGNED
// Little cheating allowed, return valid correlation only for // Little cheating allowed, return valid correlation only for
// aligned locations, meaning every second round for stereo sound. // aligned locations, meaning every second round for stereo sound.
@ -202,16 +195,16 @@ double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2,
FIRFilterSSE::FIRFilterSSE() : FIRFilter() FIRFilterSSE::FIRFilterSSE() : FIRFilter()
{ {
filterCoeffsAlign = NULL; filterCoeffsAlign = nullptr;
filterCoeffsUnalign = NULL; filterCoeffsUnalign = nullptr;
} }
FIRFilterSSE::~FIRFilterSSE() FIRFilterSSE::~FIRFilterSSE()
{ {
delete[] filterCoeffsUnalign; delete[] filterCoeffsUnalign;
filterCoeffsAlign = NULL; filterCoeffsAlign = nullptr;
filterCoeffsUnalign = NULL; filterCoeffsUnalign = nullptr;
} }
@ -252,10 +245,10 @@ uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint n
if (count < 2) return 0; if (count < 2) return 0;
assert(source != NULL); assert(source != nullptr);
assert(dest != NULL); assert(dest != nullptr);
assert((length % 8) == 0); assert((length % 8) == 0);
assert(filterCoeffsAlign != NULL); assert(filterCoeffsAlign != nullptr);
assert(((ulongptr)filterCoeffsAlign) % 16 == 0); assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2' // filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'