mirror/dolphin
1
0
Fork 0
mirror of https://github.com/dolphin-emu/dolphin.git synced 2025-03-14 13:21:15 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity

AudioCommon: Added Granular Synthesis

Browse Source
This commit is contained in:
Sam Belliveau 2025-03-02 20:19:20 -05:00 committed by Jordan Woyak
parent e82f03b825
commit f09ba10daa
51 changed files with 387 additions and 7928 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
CMakeLists.txt
View File

@ -696,11 +696,6 @@ if (APPLE OR WIN32)
include_directories(Externals/ed25519)
endif()
# Using static soundtouch from Externals
# Unable to use system soundtouch library: We require shorts, not floats.
add_subdirectory(Externals/soundtouch)
include_directories(Externals/soundtouch)
if(ENABLE_CUBEB)
dolphin_find_optional_system_library(CUBEB Externals/cubeb)
add_definitions(-DHAVE_CUBEB)

2
Externals/licenses.md vendored
View File

@ -68,8 +68,6 @@ Dolphin includes or links code of the following third-party software projects:
[zlib license](http://hg.libsdl.org/SDL/file/tip/COPYING.txt)
- [SFML](http://www.sfml-dev.org/):
[zlib license](http://www.sfml-dev.org/license.php)
- [SoundTouch](http://www.surina.net/soundtouch/):
[LGPLv2+](http://www.surina.net/soundtouch/license.html)
- [TAP-Windows](https://openvpn.net/):
header only
- [Windows Implementation Libraries](https://github.com/microsoft/wil):

222
Externals/soundtouch/AAFilter.cpp vendored
View File

@ -1,222 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// FIR low-pass (anti-alias) filter with filter coefficient design routine and
/// MMX optimization.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "AAFilter.h"
#include "FIRFilter.h"
using namespace soundtouch;
#define PI 3.14159265358979323846
#define TWOPI (2 * PI)
// define this to save AA filter coefficients to a file
// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS 1
#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS
#include <stdio.h>
static void _DEBUG_SAVE_AAFIR_COEFFS(SAMPLETYPE *coeffs, int len)
{
FILE *fptr = fopen("aa_filter_coeffs.txt", "wt");
if (fptr == nullptr) return;
for (int i = 0; i < len; i ++)
{
double temp = coeffs[i];
fprintf(fptr, "%lf\n", temp);
}
fclose(fptr);
}
#else
#define _DEBUG_SAVE_AAFIR_COEFFS(x, y)
#endif
/*****************************************************************************
*
* Implementation of the class 'AAFilter'
*
*****************************************************************************/
AAFilter::AAFilter(uint len)
{
pFIR = FIRFilter::newInstance();
cutoffFreq = 0.5;
setLength(len);
}
AAFilter::~AAFilter()
{
delete pFIR;
}
// Sets new anti-alias filter cut-off edge frequency, scaled to
// sampling frequency (nyquist frequency = 0.5).
// The filter will cut frequencies higher than the given frequency.
void AAFilter::setCutoffFreq(double newCutoffFreq)
{
cutoffFreq = newCutoffFreq;
calculateCoeffs();
}
// Sets number of FIR filter taps
void AAFilter::setLength(uint newLength)
{
length = newLength;
calculateCoeffs();
}
// Calculates coefficients for a low-pass FIR filter using Hamming window
void AAFilter::calculateCoeffs()
{
uint i;
double cntTemp, temp, tempCoeff,h, w;
double wc;
double scaleCoeff, sum;
double *work;
SAMPLETYPE *coeffs;
assert(length >= 2);
assert(length % 4 == 0);
assert(cutoffFreq >= 0);
assert(cutoffFreq <= 0.5);
work = new double[length];
coeffs = new SAMPLETYPE[length];
wc = 2.0 * PI * cutoffFreq;
tempCoeff = TWOPI / (double)length;
sum = 0;
for (i = 0; i < length; i ++)
{
cntTemp = (double)i - (double)(length / 2);
temp = cntTemp * wc;
if (temp != 0)
{
h = sin(temp) / temp; // sinc function
}
else
{
h = 1.0;
}
w = 0.54 + 0.46 * cos(tempCoeff * cntTemp); // hamming window
temp = w * h;
work[i] = temp;
// calc net sum of coefficients
sum += temp;
}
// ensure the sum of coefficients is larger than zero
assert(sum > 0);
// ensure we've really designed a lowpass filter...
assert(work[length/2] > 0);
assert(work[length/2 + 1] > -1e-6);
assert(work[length/2 - 1] > -1e-6);
// Calculate a scaling coefficient in such a way that the result can be
// divided by 16384
scaleCoeff = 16384.0f / sum;
for (i = 0; i < length; i ++)
{
temp = work[i] * scaleCoeff;
// scale & round to nearest integer
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
coeffs[i] = (SAMPLETYPE)temp;
}
// Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384
pFIR->setCoefficients(coeffs, length, 14);
_DEBUG_SAVE_AAFIR_COEFFS(coeffs, length);
delete[] work;
delete[] coeffs;
}
// Applies the filter to the given sequence of samples.
// Note : The amount of outputted samples is by value of 'filter length'
// smaller than the amount of input samples.
uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const
{
return pFIR->evaluate(dest, src, numSamples, numChannels);
}
/// Applies the filter to the given src & dest pipes, so that processed amount of
/// samples get removed from src, and produced amount added to dest
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
{
SAMPLETYPE *pdest;
const SAMPLETYPE *psrc;
uint numSrcSamples;
uint result;
int numChannels = src.getChannels();
assert(numChannels == dest.getChannels());
numSrcSamples = src.numSamples();
psrc = src.ptrBegin();
pdest = dest.ptrEnd(numSrcSamples);
result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
src.receiveSamples(result);
dest.putSamples(result);
return result;
}
uint AAFilter::getLength() const
{
return pFIR->getLength();
}

93
Externals/soundtouch/AAFilter.h vendored
View File

@ -1,93 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Anti-alias filter is used to prevent folding of high frequencies when
/// transposing the sample rate with interpolation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef AAFilter_H
#define AAFilter_H
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
namespace soundtouch
{
class AAFilter
{
protected:
class FIRFilter *pFIR;
/// Low-pass filter cut-off frequency, negative = invalid
double cutoffFreq;
/// num of filter taps
uint length;
/// Calculate the FIR coefficients realizing the given cutoff-frequency
void calculateCoeffs();
public:
AAFilter(uint length);
~AAFilter();
/// Sets new anti-alias filter cut-off edge frequency, scaled to sampling
/// frequency (nyquist frequency = 0.5). The filter will cut off the
/// frequencies than that.
void setCutoffFreq(double newCutoffFreq);
/// Sets number of FIR filter taps, i.e. ~filter complexity
void setLength(uint newLength);
uint getLength() const;
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels) const;
/// Applies the filter to the given src & dest pipes, so that processed amount of
/// samples get removed from src, and produced amount added to dest
/// Note : The amount of outputted samples is by value of 'filter length'
/// smaller than the amount of input samples.
uint evaluate(FIFOSampleBuffer &dest,
FIFOSampleBuffer &src) const;
};
}
#endif

570
Externals/soundtouch/BPMDetect.cpp vendored
View File

@ -1,570 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include <string.h>
#include <stdio.h>
#include <cfloat>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
// algorithm input sample block size
static const int INPUT_BLOCK_SIZE = 2048;
// decimated sample block size
static const int DECIMATED_BLOCK_SIZE = 256;
/// Target sample rate after decimation
static const int TARGET_SRATE = 1000;
/// XCorr update sequence size, update in about 200msec chunks
static const int XCORR_UPDATE_SEQUENCE = (int)(TARGET_SRATE / 5);
/// Moving average N size
static const int MOVING_AVERAGE_N = 15;
/// XCorr decay time constant, decay to half in 30 seconds
/// If it's desired to have the system adapt quicker to beat rate
/// changes within a continuing music stream, then the
/// 'xcorr_decay_time_constant' value can be reduced, yet that
/// can increase possibility of glitches in bpm detection.
static const double XCORR_DECAY_TIME_CONSTANT = 30.0;
/// Data overlap factor for beat detection algorithm
static const int OVERLAP_FACTOR = 4;
#define PI 3.14159265358979323846
#define TWOPI (2 * PI)
////////////////////////////////////////////////////////////////////////////////
// Enable following define to create bpm analysis file:
//#define _CREATE_BPM_DEBUG_FILE
#ifdef _CREATE_BPM_DEBUG_FILE
static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff)
{
FILE *fptr = fopen(name, "wt");
int i;
if (fptr)
{
printf("\nWriting BPM debug data into file %s\n", name);
for (i = minpos; i < maxpos; i ++)
{
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
}
fclose(fptr);
}
}
void _SaveDebugBeatPos(const char *name, const std::vector<BEAT> &beats)
{
printf("\nWriting beat detections data into file %s\n", name);
FILE *fptr = fopen(name, "wt");
if (fptr)
{
for (uint i = 0; i < beats.size(); i++)
{
BEAT b = beats[i];
fprintf(fptr, "%lf\t%lf\n", b.pos, b.strength);
}
fclose(fptr);
}
}
#else
#define _SaveDebugData(name, a,b,c,d)
#define _SaveDebugBeatPos(name, b)
#endif
// Hamming window
void hamming(float *w, int N)
{
for (int i = 0; i < N; i++)
{
w[i] = (float)(0.54 - 0.46 * cos(TWOPI * i / (N - 1)));
}
}
////////////////////////////////////////////////////////////////////////////////
//
// IIR2_filter - 2nd order IIR filter
IIR2_filter::IIR2_filter(const double *lpf_coeffs)
{
memcpy(coeffs, lpf_coeffs, 5 * sizeof(double));
memset(prev, 0, sizeof(prev));
}
float IIR2_filter::update(float x)
{
prev[0] = x;
double y = x * coeffs[0];
for (int i = 4; i >= 1; i--)
{
y += coeffs[i] * prev[i];
prev[i] = prev[i - 1];
}
prev[3] = y;
return (float)y;
}
// IIR low-pass filter coefficients, calculated with matlab/octave cheby2(2,40,0.05)
const double _LPF_coeffs[5] = { 0.00996655391939, -0.01944529148401, 0.00996655391939, 1.96867605796247, -0.96916387431724 };
////////////////////////////////////////////////////////////////////////////////
BPMDetect::BPMDetect(int numChannels, int aSampleRate) :
beat_lpf(_LPF_coeffs)
{
beats.reserve(250); // initial reservation to prevent frequent reallocation
this->sampleRate = aSampleRate;
this->channels = numChannels;
decimateSum = 0;
decimateCount = 0;
// choose decimation factor so that result is approx. 1000 Hz
decimateBy = sampleRate / TARGET_SRATE;
if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE))
{
ST_THROW_RT_ERROR("Too small samplerate");
}
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
pos = 0;
peakPos = 0;
peakVal = 0;
init_scaler = 1;
beatcorr_ringbuffpos = 0;
beatcorr_ringbuff = new float[windowLen];
memset(beatcorr_ringbuff, 0, windowLen * sizeof(float));
// allocate processing buffer
buffer = new FIFOSampleBuffer();
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
// calculate hamming windows
hamw = new float[XCORR_UPDATE_SEQUENCE];
hamming(hamw, XCORR_UPDATE_SEQUENCE);
hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2];
hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2);
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete[] beatcorr_ringbuff;
delete[] hamw;
delete[] hamw2;
delete buffer;
}
/// convert to mono, low-pass filter & decimate to about 500 Hz.
/// return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(channels > 0);
assert(decimateBy > 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
int j;
// convert to mono and accumulate
for (j = 0; j < channels; j ++)
{
decimateSum += src[j];
}
src += j;
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
decimateSum = 0;
decimateCount = 0;
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
assert(process_samples == XCORR_UPDATE_SEQUENCE);
pBuffer = buffer->ptrBegin();
// calculate decay factor for xcorr filtering
float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples));
// prescale pbuffer
float tmp[XCORR_UPDATE_SEQUENCE];
for (int i = 0; i < process_samples; i++)
{
tmp[i] = hamw[i] * hamw[i] * pBuffer[i];
}
#pragma omp parallel for
for (offs = windowStart; offs < windowLen; offs ++)
{
float sum;
int i;
sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += tmp[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable time constant.
xcorr[offs] += (float)fabs(sum);
}
}
// Detect individual beat positions
void BPMDetect::updateBeatPos(int process_samples)
{
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
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++)
{
tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i];
}
#pragma omp parallel for
for (int offs = windowStart; offs < windowLen; offs++)
{
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
}
int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
// compensate empty buffer at beginning by scaling coefficient
float scale = (float)windowLen / (float)(skipstep * init_scaler);
if (scale > 1.0f)
{
init_scaler++;
}
else
{
scale = 1.0f;
}
// detect beats
for (int i = 0; i < skipstep; i++)
{
float sum = beatcorr_ringbuff[beatcorr_ringbuffpos];
sum -= beat_lpf.update(sum);
if (sum > peakVal)
{
// found new local largest value
peakVal = sum;
peakPos = pos;
}
if (pos > peakPos + resetDur)
{
// largest value not updated for 200msec => accept as beat
peakPos += skipstep;
if (peakVal > 0)
{
// add detected beat to end of "beats" vector
BEAT temp = { (float)(peakPos * posScale), (float)(peakVal * scale) };
beats.push_back(temp);
}
peakVal = 0;
peakPos = pos;
}
beatcorr_ringbuff[beatcorr_ringbuffpos] = 0;
pos++;
beatcorr_ringbuffpos = (beatcorr_ringbuffpos + 1) % windowLen;
}
}
#define max(x,y) ((x) > (y) ? (x) : (y))
void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SIZE];
// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
while (numSamples > 0)
{
int block;
int decSamples;
block = (numSamples > INPUT_BLOCK_SIZE) ? INPUT_BLOCK_SIZE : numSamples;
// decimate. note that converts to mono at the same time
decSamples = decimate(decimated, samples, block);
samples += block * channels;
numSamples -= block;
buffer->putSamples(decimated, decSamples);
}
// when the buffer has enough samples for processing...
int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE);
while ((int)buffer->numSamples() >= req)
{
// ... update autocorrelations...
updateXCorr(XCORR_UPDATE_SEQUENCE);
// ...update beat position calculation...
updateBeatPos(XCORR_UPDATE_SEQUENCE / 2);
// ... and remove proceessed samples from the buffer
int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR;
buffer->receiveSamples(n);
}
}
void BPMDetect::removeBias()
{
int i;
// Remove linear bias: calculate linear regression coefficient
// 1. calc mean of 'xcorr' and 'i'
double mean_i = 0;
double mean_x = 0;
for (i = windowStart; i < windowLen; i++)
{
mean_x += xcorr[i];
}
mean_x /= (windowLen - windowStart);
mean_i = 0.5 * (windowLen - 1 + windowStart);
// 2. calculate linear regression coefficient
double b = 0;
double div = 0;
for (i = windowStart; i < windowLen; i++)
{
double xt = xcorr[i] - mean_x;
double xi = i - mean_i;
b += xt * xi;
div += xi * xi;
}
b /= div;
// subtract linear regression and resolve min. value bias
float minval = FLT_MAX; // arbitrary large number
for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= (float)(b * i);
if (xcorr[i] < minval)
{
minval = xcorr[i];
}
}
// subtract min.value
for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= minval;
}
}
// Calculate N-point moving average for "source" values
void MAFilter(float *dest, const float *source, int start, int end, int N)
{
for (int i = start; i < end; i++)
{
int i1 = i - N / 2;
int i2 = i + N / 2 + 1;
if (i1 < start) i1 = start;
if (i2 > end) i2 = end;
double sum = 0;
for (int j = i1; j < i2; j ++)
{
sum += source[j];
}
dest[i] = (float)(sum / (i2 - i1));
}
}
float BPMDetect::getBpm()
{
double peakPos;
double coeff;
PeakFinder peakFinder;
// remove bias from xcorr data
removeBias();
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
// save bpm debug data if debug data writing enabled
_SaveDebugData("soundtouch-bpm-xcorr.txt", xcorr, windowStart, windowLen, coeff);
// Smoothen by N-point moving-average
float *data = new float[windowLen];
memset(data, 0, sizeof(float) * windowLen);
MAFilter(data, xcorr, windowStart, windowLen, MOVING_AVERAGE_N);
// find peak position
peakPos = peakFinder.detectPeak(data, windowStart, windowLen);
// save bpm debug data if debug data writing enabled
_SaveDebugData("soundtouch-bpm-smoothed.txt", data, windowStart, windowLen, coeff);
delete[] data;
assert(decimateBy != 0);
if (peakPos < 1e-9) return 0.0; // detection failed.
_SaveDebugBeatPos("soundtouch-detected-beats.txt", beats);
// calculate BPM
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;
}

205
Externals/soundtouch/BPMDetect.h vendored
View File

@ -1,205 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Beats-per-minute (BPM) detection routine.
///
/// The beat detection algorithm works as follows:
/// - Use function 'inputSamples' to input a chunks of samples to the class for
/// analysis. It's a good idea to enter a large sound file or stream in smallish
/// chunks of around few kilosamples in order not to extinguish too much RAM memory.
/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden,
/// which is basically ok as low (bass) frequencies mostly determine the beat rate.
/// Simple averaging is used for anti-alias filtering because the resulting signal
/// quality isn't of that high importance.
/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by
/// taking absolute value that's smoothed by sliding average. Signal levels that
/// are below a couple of times the general RMS amplitude level are cut away to
/// leave only notable peaks there.
/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term
/// autocorrelation function of the enveloped signal.
/// - After whole sound data file has been analyzed as above, the bpm level is
/// detected by function 'getBpm' that finds the highest peak of the autocorrelation
/// function, calculates it's precise location and converts this reading to bpm's.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _BPMDetect_H_
#define _BPMDetect_H_
#include <vector>
#include "STTypes.h"
#include "FIFOSampleBuffer.h"
namespace soundtouch
{
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 45
/// Maximum allowed BPM rate range. Used for calculating algorithm parametrs
#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 BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Sample average counter.
int decimateCount;
/// Sample average accumulator for FIFO-like decimation.
soundtouch::LONG_SAMPLETYPE decimateSum;
/// Decimate sound by this coefficient to reach approx. 500 Hz.
int decimateBy;
/// Auto-correlation window length
int windowLen;
/// Number of channels (1 = mono, 2 = stereo)
int channels;
/// sample rate
int sampleRate;
/// Beginning of auto-correlation window: Autocorrelation isn't being updated for
/// the first these many correlation bins.
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.
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
/// are read from the internal 'buffer' pipe (samples aren't removed from the pipe
/// though).
void updateXCorr(int process_samples /// How many samples are processed.
);
/// Decimates samples to approx. 500 Hz.
///
/// \return Number of output samples.
int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer
const soundtouch::SAMPLETYPE *src, ///< Source sample buffer
int numsamples ///< Number of source samples.
);
/// Calculates amplitude envelope for the buffer of samples.
/// Result is output to 'samples'.
void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer
int numsamples ///< Number of samples in buffer
);
/// remove constant bias from xcorr data
void removeBias();
// Detect individual beat positions
void updateBeatPos(int process_samples);
public:
/// Constructor.
BPMDetect(int numChannels, ///< Number of channels in sample data.
int sampleRate ///< Sample rate in Hz.
);
/// Destructor.
virtual ~BPMDetect();
/// Inputs a block of samples for analyzing: Envelopes the samples and then
/// updates the autocorrelation estimation. When whole song data has been input
/// in smaller blocks using this function, read the resulting bpm with 'getBpm'
/// function.
///
/// Notice that data in 'samples' array can be disrupted in processing.
void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer
int numSamples ///< Number of samples in buffer
);
/// 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
/// 'inputSamples' function.
///
/// \return Beats-per-minute rate, or zero if detection failed.
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_

22
Externals/soundtouch/CMakeLists.txt vendored
View File

@ -1,22 +0,0 @@
# OSX meeds to know
check_and_add_flag(CXX11 -std=c++11)
set(SRCS
AAFilter.cpp
BPMDetect.cpp
cpu_detect_x86.cpp
FIFOSampleBuffer.cpp
FIRFilter.cpp
InterpolateCubic.cpp
InterpolateLinear.cpp
InterpolateShannon.cpp
mmx_optimized.cpp
PeakFinder.cpp
RateTransposer.cpp
SoundTouch.cpp
sse_optimized.cpp
TDStretch.cpp
)
add_library(SoundTouch STATIC ${SRCS})
dolphin_disable_warnings(SoundTouch)

275
Externals/soundtouch/FIFOSampleBuffer.cpp vendored
View File

@ -1,275 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// outputted samples from the buffer, as well as grows the buffer size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stdlib.h>
#include <memory.h>
#include <string.h>
#include <assert.h>
#include "FIFOSampleBuffer.h"
using namespace soundtouch;
// Constructor
FIFOSampleBuffer::FIFOSampleBuffer(int numChannels)
{
assert(numChannels > 0);
sizeInBytes = 0; // reasonable initial value
buffer = nullptr;
bufferUnaligned = nullptr;
samplesInBuffer = 0;
bufferPos = 0;
channels = (uint)numChannels;
ensureCapacity(32); // allocate initial capacity
}
// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
bufferUnaligned = nullptr;
buffer = nullptr;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;
if (!verifyNumberOfChannels(numChannels)) return;
usedBytes = channels * samplesInBuffer;
channels = (uint)numChannels;
samplesInBuffer = usedBytes / channels;
}
// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and
// zeroes this pointer by copying samples from the 'bufferPos' pointer
// location on to the beginning of the buffer.
void FIFOSampleBuffer::rewind()
{
if (buffer && bufferPos)
{
memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer);
bufferPos = 0;
}
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position to
// the sample buffer.
void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}
// Increases the number of samples in the buffer without copying any actual
// samples.
//
// This function is used to update the number of samples in the sample buffer
// when accessing the buffer directly with 'ptrEnd' function. Please be
// careful though!
void FIFOSampleBuffer::putSamples(uint nSamples)
{
uint req;
req = samplesInBuffer + nSamples;
ensureCapacity(req);
samplesInBuffer += nSamples;
}
// 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
// inserting new samples into the sample buffer directly. Please be careful!
//
// Parameter 'slackCapacity' tells the function how much free capacity (in
// terms of samples) there _at least_ should be, in order to the caller to
// successfully insert all the required samples to the buffer. When necessary,
// the function grows the buffer size to comply with this requirement.
//
// When using this function as means for inserting new samples, also remember
// to increase the sample count afterwards, by calling the
// 'putSamples(numSamples)' function.
SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity)
{
ensureCapacity(samplesInBuffer + slackCapacity);
return buffer + samplesInBuffer * channels;
}
// Returns a pointer to the beginning of the currently non-outputted samples.
// This function is provided for accessing the output samples directly.
// Please be careful!
//
// When using this function to output samples, also remember to 'remove' the
// outputted samples from the buffer by calling the
// 'receiveSamples(numSamples)' function
SAMPLETYPE *FIFOSampleBuffer::ptrBegin()
{
assert(buffer);
return buffer + bufferPos * channels;
}
// Ensures that the buffer has enough capacity, i.e. space for _at least_
// 'capacityRequirement' number of samples. The buffer is grown in steps of
// 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.
void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement)
{
SAMPLETYPE *tempUnaligned, *temp;
if (capacityRequirement > getCapacity())
{
// enlarge the buffer in 4kbyte steps (round up to next 4k boundary)
sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096;
assert(sizeInBytes % 2 == 0);
tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)];
if (tempUnaligned == nullptr)
{
ST_THROW_RT_ERROR("Couldn't allocate memory!\n");
}
// Align the buffer to begin at 16byte cache line boundary for optimal performance
temp = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(tempUnaligned);
if (samplesInBuffer)
{
memcpy(temp, ptrBegin(), samplesInBuffer * channels * sizeof(SAMPLETYPE));
}
delete[] bufferUnaligned;
buffer = temp;
bufferUnaligned = tempUnaligned;
bufferPos = 0;
}
else
{
// simply rewind the buffer (if necessary)
rewind();
}
}
// Returns the current buffer capacity in terms of samples
uint FIFOSampleBuffer::getCapacity() const
{
return sizeInBytes / (channels * sizeof(SAMPLETYPE));
}
// Returns the number of samples currently in the buffer
uint FIFOSampleBuffer::numSamples() const
{
return samplesInBuffer;
}
// Output samples from beginning of the sample buffer. Copies demanded number
// of samples to output and removes them from the sample buffer. If there
// are less than 'numsample' samples in the buffer, returns all available.
//
// Returns number of samples copied.
uint FIFOSampleBuffer::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint num;
num = (maxSamples > samplesInBuffer) ? samplesInBuffer : maxSamples;
memcpy(output, ptrBegin(), channels * sizeof(SAMPLETYPE) * num);
return receiveSamples(num);
}
// Removes samples from the beginning of the sample buffer without copying them
// anywhere. Used to reduce the number of samples in the buffer, when accessing
// the sample buffer with the 'ptrBegin' function.
uint FIFOSampleBuffer::receiveSamples(uint maxSamples)
{
if (maxSamples >= samplesInBuffer)
{
uint temp;
temp = samplesInBuffer;
samplesInBuffer = 0;
return temp;
}
samplesInBuffer -= maxSamples;
bufferPos += maxSamples;
return maxSamples;
}
// Returns nonzero if the sample buffer is empty
int FIFOSampleBuffer::isEmpty() const
{
return (samplesInBuffer == 0) ? 1 : 0;
}
// Clears the sample buffer
void FIFOSampleBuffer::clear()
{
samplesInBuffer = 0;
bufferPos = 0;
}
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint FIFOSampleBuffer::adjustAmountOfSamples(uint numSamples)
{
if (numSamples < samplesInBuffer)
{
samplesInBuffer = numSamples;
}
return samplesInBuffer;
}
/// Add silence to end of buffer
void FIFOSampleBuffer::addSilent(uint nSamples)
{
memset(ptrEnd(nSamples), 0, sizeof(SAMPLETYPE) * nSamples * channels);
samplesInBuffer += nSamples;
}

180
Externals/soundtouch/FIFOSampleBuffer.h vendored
View File

@ -1,180 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A buffer class for temporarily storaging sound samples, operates as a
/// first-in-first-out pipe.
///
/// Samples are added to the end of the sample buffer with the 'putSamples'
/// function, and are received from the beginning of the buffer by calling
/// the 'receiveSamples' function. The class automatically removes the
/// output samples from the buffer as well as grows the storage size
/// whenever necessary.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSampleBuffer_H
#define FIFOSampleBuffer_H
#include "FIFOSamplePipe.h"
namespace soundtouch
{
/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes
/// care of storage size adjustment and data moving during input/output operations.
///
/// Notice that in case of stereo audio, one sample is considered to consist of
/// both channel data.
class FIFOSampleBuffer : public FIFOSamplePipe
{
private:
/// Sample buffer.
SAMPLETYPE *buffer;
// Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first
// 16-byte aligned location of this buffer
SAMPLETYPE *bufferUnaligned;
/// Sample buffer size in bytes
uint sizeInBytes;
/// How many samples are currently in buffer.
uint samplesInBuffer;
/// Channels, 1=mono, 2=stereo.
uint channels;
/// Current position pointer to the buffer. This pointer is increased when samples are
/// removed from the pipe so that it's necessary to actually rewind buffer (move data)
/// only new data when is put to the pipe.
uint bufferPos;
/// Rewind the buffer by moving data from position pointed by 'bufferPos' to real
/// beginning of the buffer.
void rewind();
/// Ensures that the buffer has capacity for at least this many samples.
void ensureCapacity(uint capacityRequirement);
/// Returns current capacity.
uint getCapacity() const;
public:
/// Constructor
FIFOSampleBuffer(int numChannels = 2 ///< Number of channels, 1=mono, 2=stereo.
///< Default is stereo.
);
/// destructor
~FIFOSampleBuffer() override;
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() override;
/// 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
/// inserting new samples into the sample buffer directly. Please be careful
/// not corrupt the book-keeping!
///
/// When using this function as means for inserting new samples, also remember
/// to increase the sample count afterwards, by calling the
/// 'putSamples(numSamples)' function.
SAMPLETYPE *ptrEnd(
uint slackCapacity ///< How much free capacity (in samples) there _at least_
///< should be so that the caller can successfully insert the
///< desired samples to the buffer. If necessary, the function
///< grows the buffer size to comply with this requirement.
);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) override;
/// Adjusts the book-keeping to increase number of samples in the buffer without
/// copying any actual samples.
///
/// This function is used to update the number of samples in the sample buffer
/// when accessing the buffer directly with 'ptrEnd' function. Please be
/// careful though!
virtual void putSamples(uint numSamples ///< Number of samples been inserted.
);
/// 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
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
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
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) override;
/// Returns number of samples currently available.
virtual uint numSamples() const override;
/// Sets number of channels, 1 = mono, 2 = stereo.
void setChannels(int numChannels);
/// Get number of channels
int getChannels()
{
return channels;
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const override;
/// Clears all the samples.
virtual void clear() override;
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint adjustAmountOfSamples(uint numSamples) override;
/// Add silence to end of buffer
void addSilent(uint nSamples);
};
}
#endif

230
Externals/soundtouch/FIFOSamplePipe.h vendored
View File

@ -1,230 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound
/// samples by operating like a first-in-first-out pipe: New samples are fed
/// into one end of the pipe with the 'putSamples' function, and the processed
/// samples are received from the other end with the 'receiveSamples' function.
///
/// 'FIFOProcessor' : A base class for classes the do signal processing with
/// the samples while operating like a first-in-first-out pipe. When samples
/// are input with the 'putSamples' function, the class processes them
/// and moves the processed samples to the given 'output' pipe object, which
/// may be either another processing stage, or a fifo sample buffer object.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIFOSamplePipe_H
#define FIFOSamplePipe_H
#include <assert.h>
#include <stdlib.h>
#include "STTypes.h"
namespace soundtouch
{
/// Abstract base class for FIFO (first-in-first-out) sample processing classes.
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:
// virtual default destructor
virtual ~FIFOSamplePipe() {}
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() = 0;
/// Adds 'numSamples' pcs of samples from the 'samples' memory position to
/// the sample buffer.
virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples.
uint numSamples ///< Number of samples to insert.
) = 0;
// Moves samples from the 'other' pipe instance to this instance.
void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data.
)
{
int oNumSamples = other.numSamples();
putSamples(other.ptrBegin(), oNumSamples);
other.receiveSamples(oNumSamples);
};
/// 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
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
) = 0;
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) = 0;
/// Returns number of samples currently available.
virtual uint numSamples() const = 0;
// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const = 0;
/// Clears all the samples.
virtual void clear() = 0;
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples) = 0;
};
/// 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,
/// so that samples that are fed into beginning of the pipe automatically go through
/// all the processing stages.
///
/// When samples are input to this class, they're first processed and then put to
/// the FIFO pipe that's defined as output of this class. This output pipe can be
/// either other processing stage or a FIFO sample buffer.
class FIFOProcessor :public FIFOSamplePipe
{
protected:
/// Internal pipe where processed samples are put.
FIFOSamplePipe *output;
/// Sets output pipe.
void setOutPipe(FIFOSamplePipe *pOutput)
{
assert(output == nullptr);
assert(pOutput != nullptr);
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = nullptr;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
output = pOutput;
}
/// Destructor.
virtual ~FIFOProcessor() override
{
}
/// Returns a pointer to the beginning of the output samples.
/// This function is provided for accessing the output samples directly.
/// Please be careful for not to corrupt the book-keeping!
///
/// When using this function to output samples, also remember to 'remove' the
/// output samples from the buffer by calling the
/// 'receiveSamples(numSamples)' function
virtual SAMPLETYPE *ptrBegin() override
{
return output->ptrBegin();
}
public:
/// 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
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *outBuffer, ///< Buffer where to copy output samples.
uint maxSamples ///< How many samples to receive at max.
) override
{
return output->receiveSamples(outBuffer, maxSamples);
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe.
) override
{
return output->receiveSamples(maxSamples);
}
/// Returns number of samples currently available.
virtual uint numSamples() const override
{
return output->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const override
{
return output->isEmpty();
}
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples) override
{
return output->adjustAmountOfSamples(numSamples);
}
};
}
#endif

314
Externals/soundtouch/FIRFilter.cpp vendored
View File

@ -1,314 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Notes : MMX optimized functions reside in a separate, platform-specific 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 e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <math.h>
#include <stdlib.h>
#include "FIRFilter.h"
#include "cpu_detect.h"
using namespace soundtouch;
/*****************************************************************************
*
* Implementation of the class 'FIRFilter'
*
*****************************************************************************/
FIRFilter::FIRFilter()
{
resultDivFactor = 0;
resultDivider = 0;
length = 0;
lengthDiv8 = 0;
filterCoeffs = nullptr;
filterCoeffsStereo = nullptr;
}
FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
delete[] filterCoeffsStereo;
}
// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
int j, end;
// hint compiler autovectorization that loop length is divisible by 8
uint ilength = length & -8;
assert((length != 0) && (length == ilength) && (src != nullptr) && (dest != nullptr) && (filterCoeffs != nullptr));
assert(numSamples > ilength);
end = 2 * (numSamples - ilength);
#pragma omp parallel for
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE suml, sumr;
suml = sumr = 0;
ptr = src + j;
for (uint i = 0; i < ilength; i ++)
{
suml += ptr[2 * i] * filterCoeffsStereo[2 * i];
sumr += ptr[2 * i + 1] * filterCoeffsStereo[2 * i + 1];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
suml >>= resultDivFactor;
sumr >>= resultDivFactor;
// saturate to 16 bit integer limits
suml = (suml < -32768) ? -32768 : (suml > 32767) ? 32767 : suml;
// saturate to 16 bit integer limits
sumr = (sumr < -32768) ? -32768 : (sumr > 32767) ? 32767 : sumr;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - ilength;
}
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
int j, end;
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
assert(ilength != 0);
end = numSamples - ilength;
#pragma omp parallel for
for (j = 0; j < end; j ++)
{
const SAMPLETYPE *pSrc = src + j;
LONG_SAMPLETYPE sum;
int i;
sum = 0;
for (i = 0; i < ilength; i ++)
{
sum += pSrc[i] * filterCoeffs[i];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)sum;
}
return end;
}
uint FIRFilter::evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
{
int j, end;
assert(length != 0);
assert(src != nullptr);
assert(dest != nullptr);
assert(filterCoeffs != nullptr);
assert(numChannels < 16);
// hint compiler autovectorization that loop length is divisible by 8
int ilength = length & -8;
end = numChannels * (numSamples - ilength);
#pragma omp parallel for
for (j = 0; j < end; j += numChannels)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE sums[16];
uint c;
int i;
for (c = 0; c < numChannels; c ++)
{
sums[c] = 0;
}
ptr = src + j;
for (i = 0; i < ilength; i ++)
{
SAMPLETYPE coef=filterCoeffs[i];
for (c = 0; c < numChannels; c ++)
{
sums[c] += ptr[0] * coef;
ptr ++;
}
}
for (c = 0; c < numChannels; c ++)
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sums[c] >>= resultDivFactor;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j+c] = (SAMPLETYPE)sums[c];
}
}
return numSamples - ilength;
}
// Set filter coeffiecients and length.
//
// Throws an exception if filter length isn't divisible by 8
void FIRFilter::setCoefficients(const SAMPLETYPE *coeffs, uint newLength, uint uResultDivFactor)
{
assert(newLength > 0);
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;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
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);
}
}
uint FIRFilter::getLength() const
{
return length;
}
// Applies the filter to the given sequence of samples.
//
// Note : The amount of outputted samples is by value of 'filter_length'
// smaller than the amount of input samples.
uint FIRFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels)
{
assert(length > 0);
assert(lengthDiv8 * 8 == length);
if (numSamples < length) return 0;
#ifndef USE_MULTICH_ALWAYS
if (numChannels == 1)
{
return evaluateFilterMono(dest, src, numSamples);
}
else if (numChannels == 2)
{
return evaluateFilterStereo(dest, src, numSamples);
}
else
#endif // USE_MULTICH_ALWAYS
{
assert(numChannels > 0);
return evaluateFilterMulti(dest, src, numSamples, numChannels);
}
}
// Operator 'new' is overloaded so that it automatically creates a suitable instance
// depending on if we've a MMX-capable CPU available or not.
void * FIRFilter::operator new(size_t)
{
// 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!");
return newInstance();
}
FIRFilter * FIRFilter::newInstance()
{
uint uExtensions;
uExtensions = detectCPUextensions();
// Check if MMX/SSE instruction set extensions supported by CPU
#ifdef SOUNDTOUCH_ALLOW_MMX
// MMX routines available only with integer sample types
if (uExtensions & SUPPORT_MMX)
{
return ::new FIRFilterMMX;
}
else
#endif // SOUNDTOUCH_ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_SSE
if (uExtensions & SUPPORT_SSE)
{
// SSE support
return ::new FIRFilterSSE;
}
else
#endif // SOUNDTOUCH_ALLOW_SSE
{
// ISA optimizations not supported, use plain C version
return ::new FIRFilter;
}
}

140
Externals/soundtouch/FIRFilter.h vendored
View File

@ -1,140 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// General FIR digital filter routines with MMX optimization.
///
/// Note : MMX optimized functions reside in a separate, platform-specific file,
/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef FIRFilter_H
#define FIRFilter_H
#include <stddef.h>
#include "STTypes.h"
namespace soundtouch
{
class FIRFilter
{
protected:
// Number of FIR filter taps
uint length;
// Number of FIR filter taps divided by 8
uint lengthDiv8;
// Result divider factor in 2^k format
uint resultDivFactor;
// Result divider value.
SAMPLETYPE resultDivider;
// Memory for filter coefficients
SAMPLETYPE *filterCoeffs;
SAMPLETYPE *filterCoeffsStereo;
virtual uint evaluateFilterStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples) const;
virtual uint evaluateFilterMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels);
public:
FIRFilter();
virtual ~FIRFilter();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we've a MMX-capable CPU available or not.
static void * operator new(size_t s);
static FIRFilter *newInstance();
/// Applies the filter to the given sequence of samples.
/// Note : The amount of outputted samples is by value of 'filter_length'
/// smaller than the amount of input samples.
///
/// \return Number of samples copied to 'dest'.
uint evaluate(SAMPLETYPE *dest,
const SAMPLETYPE *src,
uint numSamples,
uint numChannels);
uint getLength() const;
virtual void setCoefficients(const SAMPLETYPE *coeffs,
uint newLength,
uint uResultDivFactor);
};
// Optional subclasses that implement CPU-specific optimizations:
#ifdef SOUNDTOUCH_ALLOW_MMX
/// Class that implements MMX optimized functions exclusive for 16bit integer samples type.
class FIRFilterMMX : public FIRFilter
{
protected:
short *filterCoeffsUnalign;
short *filterCoeffsAlign;
virtual uint evaluateFilterStereo(short *dest, const short *src, uint numSamples) const override;
public:
FIRFilterMMX();
~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor) override;
};
#endif // SOUNDTOUCH_ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_SSE
/// Class that implements SSE optimized functions exclusive for floating point samples type.
class FIRFilterSSE : public FIRFilter
{
protected:
float *filterCoeffsUnalign;
float *filterCoeffsAlign;
virtual uint evaluateFilterStereo(float *dest, const float *src, uint numSamples) const override;
public:
FIRFilterSSE();
~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor) override;
};
#endif // SOUNDTOUCH_ALLOW_SSE
}
#endif // FIRFilter_H

196
Externals/soundtouch/InterpolateCubic.cpp vendored
View File

@ -1,196 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Cubic interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <stddef.h>
#include <math.h>
#include "InterpolateCubic.h"
#include "STTypes.h"
using namespace soundtouch;
// cubic interpolation coefficients
static const float _coeffs[]=
{ -0.5f, 1.0f, -0.5f, 0.0f,
1.5f, -2.5f, 0.0f, 1.0f,
-1.5f, 2.0f, 0.5f, 0.0f,
0.5f, -0.5f, 0.0f, 0.0f};
InterpolateCubic::InterpolateCubic()
{
fract = 0;
}
void InterpolateCubic::resetRegisters()
{
fract = 0;
}
/// Transpose mono audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateCubic::transposeMono(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 4;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
float out;
const float x3 = 1.0f;
const float x2 = (float)fract; // x
const float x1 = x2*x2; // x^2
const float x0 = x1*x2; // x^3
float y0, y1, y2, y3;
assert(fract < 1.0);
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
out = y0 * psrc[0] + y1 * psrc[1] + y2 * psrc[2] + y3 * psrc[3];
pdest[i] = (SAMPLETYPE)out;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateCubic::transposeStereo(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 4;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
const float x3 = 1.0f;
const float x2 = (float)fract; // x
const float x1 = x2*x2; // x^2
const float x0 = x1*x2; // x^3
float y0, y1, y2, y3;
float out0, out1;
assert(fract < 1.0);
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
out0 = y0 * psrc[0] + y1 * psrc[2] + y2 * psrc[4] + y3 * psrc[6];
out1 = y0 * psrc[1] + y1 * psrc[3] + y2 * psrc[5] + y3 * psrc[7];
pdest[2*i] = (SAMPLETYPE)out0;
pdest[2*i+1] = (SAMPLETYPE)out1;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += 2*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose multi-channel audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateCubic::transposeMulti(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 4;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
const float x3 = 1.0f;
const float x2 = (float)fract; // x
const float x1 = x2*x2; // x^2
const float x0 = x1*x2; // x^3
float y0, y1, y2, y3;
assert(fract < 1.0);
y0 = _coeffs[0] * x0 + _coeffs[1] * x1 + _coeffs[2] * x2 + _coeffs[3] * x3;
y1 = _coeffs[4] * x0 + _coeffs[5] * x1 + _coeffs[6] * x2 + _coeffs[7] * x3;
y2 = _coeffs[8] * x0 + _coeffs[9] * x1 + _coeffs[10] * x2 + _coeffs[11] * x3;
y3 = _coeffs[12] * x0 + _coeffs[13] * x1 + _coeffs[14] * x2 + _coeffs[15] * x3;
for (int c = 0; c < numChannels; c ++)
{
float out;
out = y0 * psrc[c] + y1 * psrc[c + numChannels] + y2 * psrc[c + 2 * numChannels] + y3 * psrc[c + 3 * numChannels];
pdest[0] = (SAMPLETYPE)out;
pdest ++;
}
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += numChannels*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}

69
Externals/soundtouch/InterpolateCubic.h vendored
View File

@ -1,69 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Cubic interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateCubic_H_
#define _InterpolateCubic_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
class InterpolateCubic : public TransposerBase
{
protected:
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
virtual int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
double fract;
public:
InterpolateCubic();
virtual void resetRegisters() override;
virtual int getLatency() const override
{
return 1;
}
};
}
#endif

296
Externals/soundtouch/InterpolateLinear.cpp vendored
View File

@ -1,296 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Linear interpolation algorithm.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <assert.h>
#include <stdlib.h>
#include "InterpolateLinear.h"
using namespace soundtouch;
//////////////////////////////////////////////////////////////////////////////
//
// InterpolateLinearInteger - integer arithmetic implementation
//
/// fixed-point interpolation routine precision
#define SCALE 65536
// Constructor
InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
resetRegisters();
setRate(1.0f);
}
void InterpolateLinearInteger::resetRegisters()
{
iFract = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
LONG_SAMPLETYPE temp;
assert(iFract < SCALE);
temp = (SCALE - iFract) * src[0] + iFract * src[1];
dest[i] = (SAMPLETYPE)(temp / SCALE);
i++;
iFract += iRate;
int iWhole = iFract / SCALE;
iFract -= iWhole * SCALE;
srcCount += iWhole;
src += iWhole;
}
srcSamples = srcCount;
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Stereo' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
LONG_SAMPLETYPE temp0;
LONG_SAMPLETYPE temp1;
assert(iFract < SCALE);
temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
dest[0] = (SAMPLETYPE)(temp0 / SCALE);
dest[1] = (SAMPLETYPE)(temp1 / SCALE);
dest += 2;
i++;
iFract += iRate;
int iWhole = iFract / SCALE;
iFract -= iWhole * SCALE;
srcCount += iWhole;
src += 2*iWhole;
}
srcSamples = srcCount;
return i;
}
int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
LONG_SAMPLETYPE temp, vol1;
assert(iFract < SCALE);
vol1 = (LONG_SAMPLETYPE)(SCALE - iFract);
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + iFract * src[c + numChannels];
dest[0] = (SAMPLETYPE)(temp / SCALE);
dest ++;
}
i++;
iFract += iRate;
int iWhole = iFract / SCALE;
iFract -= iWhole * SCALE;
srcCount += iWhole;
src += iWhole * numChannels;
}
srcSamples = srcCount;
return i;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void InterpolateLinearInteger::setRate(double newRate)
{
iRate = (int)(newRate * SCALE + 0.5);
TransposerBase::setRate(newRate);
}
//////////////////////////////////////////////////////////////////////////////
//
// InterpolateLinearFloat - floating point arithmetic implementation
//
//////////////////////////////////////////////////////////////////////////////
// Constructor
InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
{
// Notice: use local function calling syntax for sake of clarity,
// to indicate the fact that C++ constructor can't call virtual functions.
resetRegisters();
setRate(1.0);
}
void InterpolateLinearFloat::resetRegisters()
{
fract = 0;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out;
assert(fract < 1.0);
out = (1.0 - fract) * src[0] + fract * src[1];
dest[i] = (SAMPLETYPE)out;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
src += whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
// Transposes the sample rate of the given samples using linear interpolation.
// 'Mono' version of the routine. Returns the number of samples returned in
// the "dest" buffer
int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out0, out1;
assert(fract < 1.0);
out0 = (1.0 - fract) * src[0] + fract * src[2];
out1 = (1.0 - fract) * src[1] + fract * src[3];
dest[2*i] = (SAMPLETYPE)out0;
dest[2*i+1] = (SAMPLETYPE)out1;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
src += 2*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 1;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
float temp, vol1, fract_float;
vol1 = (float)(1.0 - fract);
fract_float = (float)fract;
for (int c = 0; c < numChannels; c ++)
{
temp = vol1 * src[c] + fract_float * src[c + numChannels];
*dest = (SAMPLETYPE)temp;
dest ++;
}
i++;
fract += rate;
int iWhole = (int)fract;
fract -= iWhole;
srcCount += iWhole;
src += iWhole * numChannels;
}
srcSamples = srcCount;
return i;
}

98
Externals/soundtouch/InterpolateLinear.h vendored
View File

@ -1,98 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Linear interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateLinear_H_
#define _InterpolateLinear_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
/// Linear transposer class that uses integer arithmetic
class InterpolateLinearInteger : public TransposerBase
{
protected:
int iFract;
int iRate;
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples) override;
public:
InterpolateLinearInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(double newRate) override;
virtual void resetRegisters() override;
virtual int getLatency() const override
{
return 0;
}
};
/// Linear transposer class that uses floating point arithmetic
class InterpolateLinearFloat : public TransposerBase
{
protected:
double fract;
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples);
public:
InterpolateLinearFloat();
virtual void resetRegisters();
int getLatency() const
{
return 0;
}
};
}
#endif

181
Externals/soundtouch/InterpolateShannon.cpp vendored
View File

@ -1,181 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
/// with kaiser window.
///
/// Notice. This algorithm is remarkably much heavier than linear or cubic
/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
/// for experimental purposes
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include "InterpolateShannon.h"
#include "STTypes.h"
using namespace soundtouch;
/// Kaiser window with beta = 2.0
/// Values scaled down by 5% to avoid overflows
static const double _kaiser8[8] =
{
0.41778693317814,
0.64888025049173,
0.83508562409944,
0.93887857733412,
0.93887857733412,
0.83508562409944,
0.64888025049173,
0.41778693317814
};
InterpolateShannon::InterpolateShannon()
{
fract = 0;
}
void InterpolateShannon::resetRegisters()
{
fract = 0;
}
#define PI 3.1415926536
#define sinc(x) (sin(PI * (x)) / (PI * (x)))
/// Transpose mono audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeMono(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 8;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out;
assert(fract < 1.0);
out = psrc[0] * sinc(-3.0 - fract) * _kaiser8[0];
out += psrc[1] * sinc(-2.0 - fract) * _kaiser8[1];
out += psrc[2] * sinc(-1.0 - fract) * _kaiser8[2];
if (fract < 1e-6)
{
out += psrc[3] * _kaiser8[3]; // sinc(0) = 1
}
else
{
out += psrc[3] * sinc(- fract) * _kaiser8[3];
}
out += psrc[4] * sinc( 1.0 - fract) * _kaiser8[4];
out += psrc[5] * sinc( 2.0 - fract) * _kaiser8[5];
out += psrc[6] * sinc( 3.0 - fract) * _kaiser8[6];
out += psrc[7] * sinc( 4.0 - fract) * _kaiser8[7];
pdest[i] = (SAMPLETYPE)out;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeStereo(SAMPLETYPE *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
int i;
int srcSampleEnd = srcSamples - 8;
int srcCount = 0;
i = 0;
while (srcCount < srcSampleEnd)
{
double out0, out1, w;
assert(fract < 1.0);
w = sinc(-3.0 - fract) * _kaiser8[0];
out0 = psrc[0] * w; out1 = psrc[1] * w;
w = sinc(-2.0 - fract) * _kaiser8[1];
out0 += psrc[2] * w; out1 += psrc[3] * w;
w = sinc(-1.0 - fract) * _kaiser8[2];
out0 += psrc[4] * w; out1 += psrc[5] * w;
w = _kaiser8[3] * ((fract < 1e-5) ? 1.0 : sinc(- fract)); // sinc(0) = 1
out0 += psrc[6] * w; out1 += psrc[7] * w;
w = sinc( 1.0 - fract) * _kaiser8[4];
out0 += psrc[8] * w; out1 += psrc[9] * w;
w = sinc( 2.0 - fract) * _kaiser8[5];
out0 += psrc[10] * w; out1 += psrc[11] * w;
w = sinc( 3.0 - fract) * _kaiser8[6];
out0 += psrc[12] * w; out1 += psrc[13] * w;
w = sinc( 4.0 - fract) * _kaiser8[7];
out0 += psrc[14] * w; out1 += psrc[15] * w;
pdest[2*i] = (SAMPLETYPE)out0;
pdest[2*i+1] = (SAMPLETYPE)out1;
i ++;
// update position fraction
fract += rate;
// update whole positions
int whole = (int)fract;
fract -= whole;
psrc += 2*whole;
srcCount += whole;
}
srcSamples = srcCount;
return i;
}
/// Transpose stereo audio. Returns number of produced output samples, and
/// updates "srcSamples" to amount of consumed source samples
int InterpolateShannon::transposeMulti(SAMPLETYPE *,
const SAMPLETYPE *,
int &)
{
// not implemented
assert(false);
return 0;
}

74
Externals/soundtouch/InterpolateShannon.h vendored
View File

@ -1,74 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample interpolation routine using 8-tap band-limited Shannon interpolation
/// with kaiser window.
///
/// Notice. This algorithm is remarkably much heavier than linear or cubic
/// interpolation, and not remarkably better than cubic algorithm. Thus mostly
/// for experimental purposes
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateShannon_H_
#define _InterpolateShannon_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
class InterpolateShannon : public TransposerBase
{
protected:
int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) override;
double fract;
public:
InterpolateShannon();
void resetRegisters() override;
virtual int getLatency() const override
{
return 3;
}
};
}
#endif

277
Externals/soundtouch/PeakFinder.cpp vendored
View File

@ -1,277 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Peak detection routine.
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include "PeakFinder.h"
using namespace soundtouch;
#define max(x, y) (((x) > (y)) ? (x) : (y))
PeakFinder::PeakFinder()
{
minPos = maxPos = 0;
}
// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
int PeakFinder::findTop(const float *data, int peakpos) const
{
int i;
int start, end;
float refvalue;
refvalue = data[peakpos];
// seek within ±10 points
start = peakpos - 10;
if (start < minPos) start = minPos;
end = peakpos + 10;
if (end > maxPos) end = maxPos;
for (i = start; i <= end; i ++)
{
if (data[i] > refvalue)
{
peakpos = i;
refvalue = data[i];
}
}
// failure if max value is at edges of seek range => it's not peak, it's at slope.
if ((peakpos == start) || (peakpos == end)) return 0;
return peakpos;
}
// Finds 'ground level' of a peak hump by starting from 'peakpos' and proceeding
// to direction defined by 'direction' until next 'hump' after minimum value will
// begin
int PeakFinder::findGround(const float *data, int peakpos, int direction) const
{
int lowpos;
int pos;
int climb_count;
float refvalue;
float delta;
climb_count = 0;
refvalue = data[peakpos];
lowpos = peakpos;
pos = peakpos;
while ((pos > minPos+1) && (pos < maxPos-1))
{
int prevpos;
prevpos = pos;
pos += direction;
// calculate derivate
delta = data[pos] - data[prevpos];
if (delta <= 0)
{
// going downhill, ok
if (climb_count)
{
climb_count --; // decrease climb count
}
// check if new minimum found
if (data[pos] < refvalue)
{
// new minimum found
lowpos = pos;
refvalue = data[pos];
}
}
else
{
// going uphill, increase climbing counter
climb_count ++;
if (climb_count > 5) break; // we've been climbing too long => it's next uphill => quit
}
}
return lowpos;
}
// Find offset where the value crosses the given level, when starting from 'peakpos' and
// proceeds to direction defined in 'direction'
int PeakFinder::findCrossingLevel(const float *data, float level, int peakpos, int direction) const
{
float peaklevel;
int pos;
peaklevel = data[peakpos];
assert(peaklevel >= level);
pos = peakpos;
while ((pos >= minPos) && (pos + direction < maxPos))
{
if (data[pos + direction] < level) return pos; // crossing found
pos += direction;
}
return -1; // not found
}
// Calculates the center of mass location of 'data' array items between 'firstPos' and 'lastPos'
double PeakFinder::calcMassCenter(const float *data, int firstPos, int lastPos) const
{
int i;
float sum;
float wsum;
sum = 0;
wsum = 0;
for (i = firstPos; i <= lastPos; i ++)
{
sum += (float)i * data[i];
wsum += data[i];
}
if (wsum < 1e-6) return 0;
return sum / wsum;
}
/// get exact center of peak near given position by calculating local mass of center
double PeakFinder::getPeakCenter(const float *data, int peakpos) const
{
float peakLevel; // peak level
int crosspos1, crosspos2; // position where the peak 'hump' crosses cutting level
float cutLevel; // cutting value
float groundLevel; // ground level of the peak
int gp1, gp2; // bottom positions of the peak 'hump'
// find ground positions.
gp1 = findGround(data, peakpos, -1);
gp2 = findGround(data, peakpos, 1);
peakLevel = data[peakpos];
if (gp1 == gp2)
{
// avoid rounding errors when all are equal
assert(gp1 == peakpos);
cutLevel = groundLevel = peakLevel;
} else {
// get average of the ground levels
groundLevel = 0.5f * (data[gp1] + data[gp2]);
// calculate 70%-level of the peak
cutLevel = 0.70f * peakLevel + 0.30f * groundLevel;
}
// find mid-level crossings
crosspos1 = findCrossingLevel(data, cutLevel, peakpos, -1);
crosspos2 = findCrossingLevel(data, cutLevel, peakpos, 1);
if ((crosspos1 < 0) || (crosspos2 < 0)) return 0; // no crossing, no peak..
// calculate mass center of the peak surroundings
return calcMassCenter(data, crosspos1, crosspos2);
}
double PeakFinder::detectPeak(const float *data, int aminPos, int amaxPos)
{
int i;
int peakpos; // position of peak level
double highPeak, peak;
this->minPos = aminPos;
this->maxPos = amaxPos;
// find absolute peak
peakpos = minPos;
peak = data[minPos];
for (i = minPos + 1; i < maxPos; i ++)
{
if (data[i] > peak)
{
peak = data[i];
peakpos = i;
}
}
// Calculate exact location of the highest peak mass center
highPeak = getPeakCenter(data, peakpos);
peak = highPeak;
// Now check if the highest peak were in fact harmonic of the true base beat peak
// - sometimes the highest peak can be Nth harmonic of the true base peak yet
// just a slightly higher than the true base
for (i = 1; i < 3; i ++)
{
double peaktmp, harmonic;
int i1,i2;
harmonic = (double)pow(2.0, i);
peakpos = (int)(highPeak / harmonic + 0.5f);
if (peakpos < minPos) break;
peakpos = findTop(data, peakpos); // seek true local maximum index
if (peakpos == 0) continue; // no local max here
// calculate mass-center of possible harmonic peak
peaktmp = getPeakCenter(data, peakpos);
// accept harmonic peak if
// (a) it is found
// (b) is within ±4% of the expected harmonic interval
// (c) has at least half x-corr value of the max. peak
double diff = harmonic * peaktmp / highPeak;
if ((diff < 0.96) || (diff > 1.04)) continue; // peak too afar from expected
// now compare to highest detected peak
i1 = (int)(highPeak + 0.5);
i2 = (int)(peaktmp + 0.5);
if (data[i2] >= 0.4*data[i1])
{
// The harmonic is at least half as high primary peak,
// thus use the harmonic peak instead
peak = peaktmp;
}
}
return peak;
}

90
Externals/soundtouch/PeakFinder.h vendored
View File

@ -1,90 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// The routine detects highest value on an array of values and calculates the
/// precise peak location as a mass-center of the 'hump' around the peak value.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _PeakFinder_H_
#define _PeakFinder_H_
namespace soundtouch
{
class PeakFinder
{
protected:
/// Min, max allowed peak positions within the data vector
int minPos, maxPos;
/// Calculates the mass center between given vector items.
double calcMassCenter(const float *data, ///< Data vector.
int firstPos, ///< Index of first vector item belonging to the peak.
int lastPos ///< Index of last vector item belonging to the peak.
) const;
/// Finds the data vector index where the monotoniously decreasing signal crosses the
/// given level.
int findCrossingLevel(const float *data, ///< Data vector.
float level, ///< Goal crossing level.
int peakpos, ///< Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
// Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'.
int findTop(const float *data, int peakpos) const;
/// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right-
/// or left-hand side of the given peak position.
int findGround(const float *data, /// Data vector.
int peakpos, /// Peak position index within the data vector.
int direction /// Direction where to proceed from the peak: 1 = right, -1 = left.
) const;
/// get exact center of peak near given position by calculating local mass of center
double getPeakCenter(const float *data, int peakpos) const;
public:
/// Constructor.
PeakFinder();
/// Detect exact peak position of the data vector by finding the largest peak 'hump'
/// and calculating the mass-center location of the peak hump.
///
/// \return The location of the largest base harmonic peak hump.
double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has
/// to be at least 'maxPos' items long.
int minPos, ///< Min allowed peak location within the vector data.
int maxPos ///< Max allowed peak location within the vector data.
);
};
}
#endif // _PeakFinder_H_

313
Externals/soundtouch/RateTransposer.cpp vendored
View File

@ -1,313 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application)
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <memory.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include "RateTransposer.h"
#include "InterpolateLinear.h"
#include "InterpolateCubic.h"
#include "InterpolateShannon.h"
#include "AAFilter.h"
using namespace soundtouch;
// Define default interpolation algorithm here
TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
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
pAAFilter = new AAFilter(64);
pTransposer = TransposerBase::newInstance();
clear();
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
delete pTransposer;
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
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;
clear();
#endif
}
/// Returns nonzero if anti-alias filter is enabled.
bool RateTransposer::isAAFilterEnabled() const
{
return bUseAAFilter;
}
AAFilter *RateTransposer::getAAFilter()
{
return pAAFilter;
}
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
// iRate, larger faster iRates.
void RateTransposer::setRate(double newRate)
{
double fCutoff;
pTransposer->setRate(newRate);
// design a new anti-alias filter
if (newRate > 1.0)
{
fCutoff = 0.5 / newRate;
}
else
{
fCutoff = 0.5 * newRate;
}
pAAFilter->setCutoffFreq(fCutoff);
}
// Adds 'nSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
processSamples(samples, nSamples);
}
// Transposes sample rate by applying anti-alias filter to prevent folding.
// Returns amount of samples returned in the "dest" buffer.
// The maximum amount of samples that can be returned at a time is set by
// the 'set_returnBuffer_size' function.
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
{
if (nSamples == 0) return;
// Store samples to input buffer
inputBuffer.putSamples(src, nSamples);
// If anti-alias filter is turned off, simply transpose without applying
// the filter
if (bUseAAFilter == false)
{
(void)pTransposer->transpose(outputBuffer, inputBuffer);
return;
}
assert(pAAFilter);
// Transpose with anti-alias filter
if (pTransposer->rate < 1.0f)
{
// If the parameter 'Rate' value is smaller than 1, first transpose
// the samples and then apply the anti-alias filter to remove aliasing.
// Transpose the samples, store the result to end of "midBuffer"
pTransposer->transpose(midBuffer, inputBuffer);
// Apply the anti-alias filter for transposed samples in midBuffer
pAAFilter->evaluate(outputBuffer, midBuffer);
}
else
{
// If the parameter 'Rate' value is larger than 1, first apply the
// anti-alias filter to remove high frequencies (prevent them from folding
// over the lover frequencies), then transpose.
// Apply the anti-alias filter for samples in inputBuffer
pAAFilter->evaluate(midBuffer, inputBuffer);
// Transpose the AA-filtered samples in "midBuffer"
pTransposer->transpose(outputBuffer, midBuffer);
}
}
// Sets the number of channels, 1 = mono, 2 = stereo
void RateTransposer::setChannels(int nChannels)
{
if (!verifyNumberOfChannels(nChannels) ||
(pTransposer->numChannels == nChannels)) return;
pTransposer->setChannels(nChannels);
inputBuffer.setChannels(nChannels);
midBuffer.setChannels(nChannels);
outputBuffer.setChannels(nChannels);
}
// Clears all the samples in the object
void RateTransposer::clear()
{
outputBuffer.clear();
midBuffer.clear();
inputBuffer.clear();
pTransposer->resetRegisters();
// prefill buffer to avoid losing first samples at beginning of stream
int prefill = getLatency();
inputBuffer.addSilent(prefill);
}
// Returns nonzero if there aren't any samples available for outputting.
int RateTransposer::isEmpty() const
{
int res;
res = FIFOProcessor::isEmpty();
if (res == 0) return 0;
return inputBuffer.isEmpty();
}
/// Return approximate initial input-output latency
int RateTransposer::getLatency() const
{
return pTransposer->getLatency() +
((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
}
//////////////////////////////////////////////////////////////////////////////
//
// TransposerBase - Base class for interpolation
//
// static function to set interpolation algorithm
void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
{
TransposerBase::algorithm = a;
}
// Transposes the sample rate of the given samples using linear interpolation.
// Returns the number of samples returned in the "dest" buffer
int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
{
int numSrcSamples = src.numSamples();
int sizeDemand = (int)((double)numSrcSamples / rate) + 8;
int numOutput;
SAMPLETYPE *psrc = src.ptrBegin();
SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
#ifndef USE_MULTICH_ALWAYS
if (numChannels == 1)
{
numOutput = transposeMono(pdest, psrc, numSrcSamples);
}
else if (numChannels == 2)
{
numOutput = transposeStereo(pdest, psrc, numSrcSamples);
}
else
#endif // USE_MULTICH_ALWAYS
{
assert(numChannels > 0);
numOutput = transposeMulti(pdest, psrc, numSrcSamples);
}
dest.putSamples(numOutput);
src.receiveSamples(numSrcSamples);
return numOutput;
}
TransposerBase::TransposerBase()
{
numChannels = 0;
rate = 1.0f;
}
TransposerBase::~TransposerBase()
{
}
void TransposerBase::setChannels(int channels)
{
numChannels = channels;
resetRegisters();
}
void TransposerBase::setRate(double newRate)
{
rate = newRate;
}
// static factory function
TransposerBase *TransposerBase::newInstance()
{
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// Notice: For integer arithmetic support only linear algorithm (due to simplest calculus)
return ::new InterpolateLinearInteger;
#else
switch (algorithm)
{
case LINEAR:
return new InterpolateLinearFloat;
case CUBIC:
return new InterpolateCubic;
case SHANNON:
return new InterpolateShannon;
default:
assert(false);
return nullptr;
}
#endif
}

164
Externals/soundtouch/RateTransposer.h vendored
View File

@ -1,164 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sample rate transposer. Changes sample rate by using linear interpolation
/// together with anti-alias filtering (first order interpolation with anti-
/// alias filtering should be quite adequate for this application).
///
/// Use either of the derived classes of 'RateTransposerInteger' or
/// 'RateTransposerFloat' for corresponding integer/floating point tranposing
/// algorithm implementation.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef RateTransposer_H
#define RateTransposer_H
#include <stddef.h>
#include "AAFilter.h"
#include "FIFOSamplePipe.h"
#include "FIFOSampleBuffer.h"
#include "STTypes.h"
namespace soundtouch
{
/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
class TransposerBase
{
public:
enum ALGORITHM {
LINEAR = 0,
CUBIC,
SHANNON
};
protected:
virtual int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
virtual int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
virtual int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples) = 0;
static ALGORITHM algorithm;
public:
double rate;
int numChannels;
TransposerBase();
virtual ~TransposerBase();
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
virtual void setRate(double newRate);
virtual void setChannels(int channels);
virtual int getLatency() const = 0;
virtual void resetRegisters() = 0;
// static factory function
static TransposerBase *newInstance();
// static function to set interpolation algorithm
static void setAlgorithm(ALGORITHM a);
};
/// A common linear samplerate transposer class.
///
class RateTransposer : public FIFOProcessor
{
protected:
/// Anti-alias filter object
AAFilter *pAAFilter;
TransposerBase *pTransposer;
/// Buffer for collecting samples to feed the anti-alias filter between
/// two batches
FIFOSampleBuffer inputBuffer;
/// Buffer for keeping samples between transposing & anti-alias filter
FIFOSampleBuffer midBuffer;
/// Output sample buffer
FIFOSampleBuffer outputBuffer;
bool bUseAAFilter;
/// Transposes sample rate by applying anti-alias filter to prevent folding.
/// Returns amount of samples returned in the "dest" buffer.
/// The maximum amount of samples that can be returned at a time is set by
/// the 'set_returnBuffer_size' function.
void processSamples(const SAMPLETYPE *src,
uint numSamples);
public:
RateTransposer();
virtual ~RateTransposer() override;
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Return anti-alias filter object
AAFilter *getAAFilter();
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void enableAAFilter(bool newMode);
/// Returns nonzero if anti-alias filter is enabled.
bool isAAFilterEnabled() const;
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(double newRate);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(int channels);
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
void putSamples(const SAMPLETYPE *samples, uint numSamples) override;
/// Clears all the samples in the object
void clear() override;
/// Returns nonzero if there aren't any samples available for outputting.
int isEmpty() const override;
/// Return approximate initial input-output latency
int getLatency() const;
};
}
#endif

189
Externals/soundtouch/STTypes.h vendored
View File

@ -1,189 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Common type definitions for SoundTouch audio processing library.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef STTypes_H
#define STTypes_H
#if (defined(__GNUC__) && !defined(ANDROID))
// In GCC, include soundtouch_config.h made by config scritps.
// Skip this in Android compilation that uses GCC but without configure scripts.
//#include "soundtouch_config.h"
#endif
namespace soundtouch
{
typedef unsigned int uint;
typedef unsigned long ulong;
// Patch for MinGW: on Win64 long is 32-bit
#ifdef _WIN64
typedef unsigned long long ulongptr;
#else
typedef ulong ulongptr;
#endif
// Helper macro for aligning pointer up to next 16-byte boundary
#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
/// setting inherited from some other header file:
#undef SOUNDTOUCH_INTEGER_SAMPLES
#undef SOUNDTOUCH_FLOAT_SAMPLES
/// If following flag is defined, always uses multichannel processing
/// routines also for mono and stero sound. This is for routine testing
/// purposes; output should be same with either routines, yet disabling
/// the dedicated mono/stereo processing routines will result in slower
/// runtime performance so recommendation is to keep this off.
// #define USE_MULTICH_ALWAYS
#if (defined(__SOFTFP__) && defined(ANDROID))
// For Android compilation: Force use of Integer samples in case that
// compilation uses soft-floating point emulation - soft-fp is way too slow
#undef SOUNDTOUCH_FLOAT_SAMPLES
#define SOUNDTOUCH_INTEGER_SAMPLES 1
#endif
#if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES)
/// Choose either 32bit floating point or 16bit integer sampletype
/// by choosing one of the following defines, unless this selection
/// has already been done in some other file.
////
/// Notes:
/// - In Windows environment, choose the sample format with the
/// following defines.
/// - In GNU environment, the floating point samples are used by
/// default, but integer samples can be chosen by giving the
/// following switch to the configure script:
/// ./configure --enable-integer-samples
/// However, if you still prefer to select the sample format here
/// also in GNU environment, then please #undef the INTEGER_SAMPLE
/// and FLOAT_SAMPLE defines first as in comments above.
#define SOUNDTOUCH_INTEGER_SAMPLES 1 //< 16bit integer samples
//#define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples
#endif
#if (_M_IX86 || __i386__ || __x86_64__ || _M_X64)
/// Define this to allow X86-specific assembler/intrinsic optimizations.
/// Notice that library contains also usual C++ versions of each of these
/// these routines, so if you're having difficulties getting the optimized
/// routines compiled for whatever reason, you may disable these optimizations
/// to make the library compile.
//#define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1
/// In GNU environment, allow the user to override this setting by
/// giving the following switch to the configure script:
/// ./configure --disable-x86-optimizations
/// ./configure --enable-x86-optimizations=no
#ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
#endif
#else
/// Always disable optimizations when not using a x86 systems.
#undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
#endif
// If defined, allows the SIMD-optimized routines to skip unevenly aligned
// memory offsets that can cause performance penalty in some SIMD implementations.
// Causes slight compromise in sound quality.
#define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION 1
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// 16bit integer sample type
typedef short SAMPLETYPE;
// data type for sample accumulation: Use 32bit integer to prevent overflows
typedef long LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// check that only one sample type is defined
#error "conflicting sample types defined"
#endif // SOUNDTOUCH_FLOAT_SAMPLES
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow MMX optimizations (not available in X64 mode)
#if (!_M_X64)
#define SOUNDTOUCH_ALLOW_MMX 1
#endif
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use float also here to enable
// efficient autovectorization
typedef float LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations
#define SOUNDTOUCH_ALLOW_SSE 1
#endif
#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 1
#ifdef ST_NO_EXCEPTION_HANDLING
// Exceptions disabled. Throw asserts instead if enabled.
#include <assert.h>
#define ST_THROW_RT_ERROR(x) {assert((const char *)x);}
#else
// use c++ standard exceptions
#include <stdexcept>
#include <string>
#define ST_THROW_RT_ERROR(x) {throw std::runtime_error(x);}
#endif
// When this #define is active, eliminates a clicking sound when the "rate" or "pitch"
// parameter setting crosses from value <1 to >=1 or vice versa during processing.
// Default is off as such crossover is untypical case and involves a slight sound
// quality compromise.
//#define SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER 1
#endif

538
Externals/soundtouch/SoundTouch.cpp vendored
View File

@ -1,538 +0,0 @@
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <assert.h>
#include <stdlib.h>
#include <memory.h>
#include <math.h>
#include <stdio.h>
#include "SoundTouch.h"
#include "TDStretch.h"
#include "RateTransposer.h"
#include "cpu_detect.h"
using namespace soundtouch;
/// test if two floating point numbers are equal
#define TEST_FLOAT_EQUAL(a, b) (fabs(a - b) < 1e-10)
/// Print library version string for autoconf
extern "C" void soundtouch_ac_test()
{
printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION);
}
SoundTouch::SoundTouch()
{
// Initialize rate transposer and tempo changer instances
pRateTransposer = new RateTransposer();
pTDStretch = TDStretch::newInstance();
setOutPipe(pTDStretch);
rate = tempo = 0;
virtualPitch =
virtualRate =
virtualTempo = 1.0;
calcEffectiveRateAndTempo();
samplesExpectedOut = 0;
samplesOutput = 0;
channels = 0;
bSrateSet = false;
}
SoundTouch::~SoundTouch()
{
delete pRateTransposer;
delete pTDStretch;
}
/// Get SoundTouch library version string
const char *SoundTouch::getVersionString()
{
static const char *_version = SOUNDTOUCH_VERSION;
return _version;
}
/// Get SoundTouch library version Id
uint SoundTouch::getVersionId()
{
return SOUNDTOUCH_VERSION_ID;
}
// Sets the number of channels, 1 = mono, 2 = stereo
void SoundTouch::setChannels(uint numChannels)
{
if (!verifyNumberOfChannels(numChannels)) return;
channels = numChannels;
pRateTransposer->setChannels((int)numChannels);
pTDStretch->setChannels((int)numChannels);
}
// Sets new rate control value. Normal rate = 1.0, smaller values
// represent slower rate, larger faster rates.
void SoundTouch::setRate(double newRate)
{
virtualRate = newRate;
calcEffectiveRateAndTempo();
}
// Sets new rate control value as a difference in percents compared
// to the original rate (-50 .. +100 %)
void SoundTouch::setRateChange(double newRate)
{
virtualRate = 1.0 + 0.01 * newRate;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value. Normal tempo = 1.0, smaller values
// represent slower tempo, larger faster tempo.
void SoundTouch::setTempo(double newTempo)
{
virtualTempo = newTempo;
calcEffectiveRateAndTempo();
}
// Sets new tempo control value as a difference in percents compared
// to the original tempo (-50 .. +100 %)
void SoundTouch::setTempoChange(double newTempo)
{
virtualTempo = 1.0 + 0.01 * newTempo;
calcEffectiveRateAndTempo();
}
// Sets new pitch control value. Original pitch = 1.0, smaller values
// represent lower pitches, larger values higher pitch.
void SoundTouch::setPitch(double newPitch)
{
virtualPitch = newPitch;
calcEffectiveRateAndTempo();
}
// Sets pitch change in octaves compared to the original pitch
// (-1.00 .. +1.00)
void SoundTouch::setPitchOctaves(double newPitch)
{
virtualPitch = exp(0.69314718056 * newPitch);
calcEffectiveRateAndTempo();
}
// Sets pitch change in semi-tones compared to the original pitch
// (-12 .. +12)
void SoundTouch::setPitchSemiTones(int newPitch)
{
setPitchOctaves((double)newPitch / 12.0);
}
void SoundTouch::setPitchSemiTones(double newPitch)
{
setPitchOctaves(newPitch / 12.0);
}
// Calculates 'effective' rate and tempo values from the
// nominal control values.
void SoundTouch::calcEffectiveRateAndTempo()
{
double oldTempo = tempo;
double oldRate = rate;
tempo = virtualTempo / virtualPitch;
rate = virtualPitch * virtualRate;
if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate);
if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo);
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0f)
{
if (output != pTDStretch)
{
FIFOSamplePipe *tempoOut;
assert(output == pRateTransposer);
// move samples in the current output buffer to the output of pTDStretch
tempoOut = pTDStretch->getOutput();
tempoOut->moveSamples(*output);
// move samples in pitch transposer's store buffer to tempo changer's input
// deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
output = pTDStretch;
}
}
else
#endif
{
if (output != pRateTransposer)
{
FIFOSamplePipe *transOut;
assert(output == pTDStretch);
// move samples in the current output buffer to the output of pRateTransposer
transOut = pRateTransposer->getOutput();
transOut->moveSamples(*output);
// move samples in tempo changer's input to pitch transposer's input
pRateTransposer->moveSamples(*pTDStretch->getInput());
output = pRateTransposer;
}
}
}
// Sets sample rate.
void SoundTouch::setSampleRate(uint srate)
{
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters((int)srate);
bSrateSet = true;
}
// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples)
{
if (bSrateSet == false)
{
ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined");
}
else if (channels == 0)
{
ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined");
}
// accumulate how many samples are expected out from processing, given the current
// processing setting
samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo);
#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
if (rate <= 1.0f)
{
// transpose the rate down, output the transposed sound to tempo changer buffer
assert(output == pTDStretch);
pRateTransposer->putSamples(samples, nSamples);
pTDStretch->moveSamples(*pRateTransposer);
}
else
#endif
{
// evaluate the tempo changer, then transpose the rate up,
assert(output == pRateTransposer);
pTDStretch->putSamples(samples, nSamples);
pRateTransposer->moveSamples(*pTDStretch);
}
}
// Flushes the last samples from the processing pipeline to the output.
// Clears also the internal processing buffers.
//
// Note: This function is meant for extracting the last samples of a sound
// stream. This function may introduce additional blank samples in the end
// of the sound stream, and thus it's not recommended to call this function
// in the middle of a sound stream.
void SoundTouch::flush()
{
int i;
int numStillExpected;
SAMPLETYPE *buff = new SAMPLETYPE[128 * channels];
// how many samples are still expected to output
numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput);
if (numStillExpected < 0) numStillExpected = 0;
memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE));
// "Push" the last active samples out from the processing pipeline by
// feeding blank samples into the processing pipeline until new,
// processed samples appear in the output (not however, more than
// 24ksamples in any case)
for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++)
{
putSamples(buff, 128);
}
adjustAmountOfSamples(numStillExpected);
delete[] buff;
// Clear input buffers
pTDStretch->clearInput();
// yet leave the output intouched as that's where the
// flushed samples are!
}
// Changes a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
bool SoundTouch::setSetting(int settingId, int value)
{
int sampleRate, sequenceMs, seekWindowMs, overlapMs;
// read current tdstretch routine parameters
pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs);
switch (settingId)
{
case SETTING_USE_AA_FILTER :
// enables / disabless anti-alias filter
pRateTransposer->enableAAFilter((value != 0) ? true : false);
return true;
case SETTING_AA_FILTER_LENGTH :
// sets anti-alias filter length
pRateTransposer->getAAFilter()->setLength(value);
return true;
case SETTING_USE_QUICKSEEK :
// enables / disables tempo routine quick seeking algorithm
pTDStretch->enableQuickSeek((value != 0) ? true : false);
return true;
case SETTING_SEQUENCE_MS:
// change time-stretch sequence duration parameter
pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs);
return true;
case SETTING_SEEKWINDOW_MS:
// change time-stretch seek window length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs);
return true;
case SETTING_OVERLAP_MS:
// change time-stretch overlap length parameter
pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value);
return true;
default :
return false;
}
}
// Reads a setting controlling the processing system behaviour. See the
// 'SETTING_...' defines for available setting ID's.
//
// Returns the setting value.
int SoundTouch::getSetting(int settingId) const
{
int temp;
switch (settingId)
{
case SETTING_USE_AA_FILTER :
return (uint)pRateTransposer->isAAFilterEnabled();
case SETTING_AA_FILTER_LENGTH :
return pRateTransposer->getAAFilter()->getLength();
case SETTING_USE_QUICKSEEK :
return (uint)pTDStretch->isQuickSeekEnabled();
case SETTING_SEQUENCE_MS:
pTDStretch->getParameters(nullptr, &temp, nullptr, nullptr);
return temp;
case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(nullptr, nullptr, &temp, nullptr);
return temp;
case SETTING_OVERLAP_MS:
pTDStretch->getParameters(nullptr, nullptr, nullptr, &temp);
return temp;
case SETTING_NOMINAL_INPUT_SEQUENCE :
{
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 :
{
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 :
return 0;
}
}
// Clears all the samples in the object's output and internal processing
// buffers.
void SoundTouch::clear()
{
samplesExpectedOut = 0;
samplesOutput = 0;
pRateTransposer->clear();
pTDStretch->clear();
}
/// Returns number of samples currently unprocessed.
uint SoundTouch::numUnprocessedSamples() const
{
FIFOSamplePipe * psp;
if (pTDStretch)
{
psp = pTDStretch->getInput();
if (psp)
{
return psp->numSamples();
}
}
return 0;
}
/// 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
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples)
{
uint ret = FIFOProcessor::receiveSamples(output, maxSamples);
samplesOutput += (long)ret;
return ret;
}
/// Adjusts book-keeping so that given number of samples are removed from beginning of the
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
uint SoundTouch::receiveSamples(uint maxSamples)
{
uint ret = FIFOProcessor::receiveSamples(maxSamples);
samplesOutput += (long)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);
}

348
Externals/soundtouch/SoundTouch.h vendored
View File

@ -1,348 +0,0 @@
//////////////////////////////////////////////////////////////////////////////
///
/// SoundTouch - main class for tempo/pitch/rate adjusting routines.
///
/// Notes:
/// - Initialize the SoundTouch object instance by setting up the sound stream
/// parameters with functions 'setSampleRate' and 'setChannels', then set
/// desired tempo/pitch/rate settings with the corresponding functions.
///
/// - The SoundTouch class behaves like a first-in-first-out pipeline: The
/// samples that are to be processed are fed into one of the pipe by calling
/// function 'putSamples', while the ready processed samples can be read
/// from the other end of the pipeline with function 'receiveSamples'.
///
/// - The SoundTouch processing classes require certain sized 'batches' of
/// samples in order to process the sound. For this reason the classes buffer
/// incoming samples until there are enough of samples available for
/// processing, then they carry out the processing step and consequently
/// make the processed samples available for outputting.
///
/// - For the above reason, the processing routines introduce a certain
/// 'latency' between the input and output, so that the samples input to
/// SoundTouch may not be immediately available in the output, and neither
/// the amount of outputtable samples may not immediately be in direct
/// relationship with the amount of previously input samples.
///
/// - The tempo/pitch/rate control parameters can be altered during processing.
/// Please notice though that they aren't currently protected by semaphores,
/// so in multi-thread application external semaphore protection may be
/// required.
///
/// - This class utilizes classes 'TDStretch' for tempo change (without modifying
/// pitch) and 'RateTransposer' for changing the playback rate (that is, both
/// tempo and pitch in the same ratio) of the sound. The third available control
/// 'pitch' (change pitch but maintain tempo) is produced by a combination of
/// combining the two other controls.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef SoundTouch_H
#define SoundTouch_H
#include "FIFOSamplePipe.h"
#include "STTypes.h"
namespace soundtouch
{
/// Soundtouch library version string
#define SOUNDTOUCH_VERSION "2.3.2"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (20302)
//
// Available setting IDs for the 'setSetting' & 'get_setting' functions:
/// Enable/disable anti-alias filter in pitch transposer (0 = disable)
#define SETTING_USE_AA_FILTER 0
/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32)
#define SETTING_AA_FILTER_LENGTH 1
/// Enable/disable quick seeking algorithm in tempo changer routine
/// (enabling quick seeking lowers CPU utilization but causes a minor sound
/// quality compromising)
#define SETTING_USE_QUICKSEEK 2
/// Time-stretch algorithm single processing sequence length in milliseconds. This determines
/// to how long sequences the original sound is chopped in the time-stretch algorithm.
/// See "STTypes.h" or README for more information.
#define SETTING_SEQUENCE_MS 3
/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the
/// best possible overlapping location. This determines from how wide window the algorithm
/// may look for an optimal joining location when mixing the sound sequences back together.
/// See "STTypes.h" or README for more information.
#define SETTING_SEEKWINDOW_MS 4
/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences
/// are mixed back together, to form a continuous sound stream, this parameter defines over
/// how long period the two consecutive sequences are let to overlap each other.
/// See "STTypes.h" or README for more information.
#define SETTING_OVERLAP_MS 5
/// Call "getSetting" with this ID to query processing sequence size in samples.
/// This value gives approximate value of how many input samples you'll need to
/// 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:
/// - 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_NOMINAL_INPUT_SEQUENCE 6
/// Call "getSetting" with this ID to query nominal average processing output
/// size in samples. This value tells approcimate value how many output samples
/// SoundTouch outputs once it does DSP processing run for a batch of input 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_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
{
private:
/// Rate transposer class instance
class RateTransposer *pRateTransposer;
/// Time-stretch class instance
class TDStretch *pTDStretch;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
double virtualRate;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
double virtualTempo;
/// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters.
double virtualPitch;
/// Flag: Has sample rate been set?
bool bSrateSet;
/// Accumulator for how many samples in total will be expected as output vs. samples put in,
/// considering current processing settings.
double samplesExpectedOut;
/// Accumulator for how many samples in total have been read out from the processing so far
long samplesOutput;
/// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and
/// 'virtualPitch' parameters.
void calcEffectiveRateAndTempo();
protected :
/// Number of channels
uint channels;
/// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
double rate;
/// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch'
double tempo;
public:
SoundTouch();
virtual ~SoundTouch() override;
/// Get SoundTouch library version string
static const char *getVersionString();
/// Get SoundTouch library version Id
static uint getVersionId();
/// Sets new rate control value. Normal rate = 1.0, smaller values
/// represent slower rate, larger faster rates.
void setRate(double newRate);
/// Sets new tempo control value. Normal tempo = 1.0, smaller values
/// represent slower tempo, larger faster tempo.
void setTempo(double newTempo);
/// Sets new rate control value as a difference in percents compared
/// to the original rate (-50 .. +100 %)
void setRateChange(double newRate);
/// Sets new tempo control value as a difference in percents compared
/// to the original tempo (-50 .. +100 %)
void setTempoChange(double newTempo);
/// Sets new pitch control value. Original pitch = 1.0, smaller values
/// represent lower pitches, larger values higher pitch.
void setPitch(double newPitch);
/// Sets pitch change in octaves compared to the original pitch
/// (-1.00 .. +1.00)
void setPitchOctaves(double newPitch);
/// Sets pitch change in semi-tones compared to the original pitch
/// (-12 .. +12)
void setPitchSemiTones(int newPitch);
void setPitchSemiTones(double newPitch);
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(uint numChannels);
/// Sets sample rate.
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.
/// Clears also the internal processing buffers.
//
/// Note: This function is meant for extracting the last samples of a sound
/// stream. This function may introduce additional blank samples in the end
/// of the sound stream, and thus it's not recommended to call this function
/// in the middle of a sound stream.
void flush();
/// Adds 'numSamples' pcs of samples from the 'samples' memory position into
/// the input of the object. Notice that sample rate _has_to_ be set before
/// calling this function, otherwise throws a runtime_error exception.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Pointer to sample buffer.
uint numSamples ///< Number of samples in buffer. Notice
///< that in case of stereo-sound a single sample
///< contains data for both channels.
) override;
/// 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
/// 'numsample' samples in the buffer, returns all that available.
///
/// \return Number of samples returned.
virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples.
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
/// sample buffer without copying them anywhere.
///
/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly
/// with 'ptrBegin' function.
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
/// buffers.
virtual void clear() override;
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'true' if the setting was successfully changed
bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines.
int value ///< New setting value.
);
/// Reads a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return the setting value.
int getSetting(int settingId ///< Setting ID number, see SETTING_... defines.
) const;
/// Returns number of samples currently unprocessed.
virtual uint numUnprocessedSamples() const;
/// Return number of channels
uint numChannels() const
{
return channels;
}
/// Other handy functions that are implemented in the ancestor classes (see
/// classes 'FIFOProcessor' and 'FIFOSamplePipe')
///
/// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch.
/// - numSamples() : Get number of 'ready' samples that can be received with
/// function 'receiveSamples()'
/// - isEmpty() : Returns nonzero if there aren't any 'ready' samples.
/// - clear() : Clears all samples from ready/processing buffers.
};
}
#endif

56
Externals/soundtouch/SoundTouch.vcxproj vendored
View File

@ -1,56 +0,0 @@
<?xml version="1.0" encoding="utf-8"?>
<Project>
<Import Project="..\..\Source\VSProps\Base.Macros.props" />
<Import Project="$(VSPropsDir)Base.Targets.props" />
<PropertyGroup Label="Globals">
<ProjectGuid>{EC082900-B4D8-42E9-9663-77F02F6936AE}</ProjectGuid>
</PropertyGroup>
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.Default.props" />
<Import Project="$(VSPropsDir)Configuration.StaticLibrary.props" />
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.props" />
<ImportGroup Label="ExtensionSettings" />
<ImportGroup Label="PropertySheets">
<Import Project="$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props" Condition="exists('$(UserRootDir)\Microsoft.Cpp.$(Platform).user.props')" Label="LocalAppDataPlatform" />
<Import Project="$(VSPropsDir)Base.props" />
<Import Project="$(VSPropsDir)ClDisableAllWarnings.props" />
</ImportGroup>
<PropertyGroup Label="UserMacros" />
<ItemGroup>
<ClCompile Include="AAFilter.cpp" />
<ClCompile Include="BPMDetect.cpp" />
<ClCompile Include="cpu_detect_x86.cpp" />
<ClCompile Include="FIFOSampleBuffer.cpp" />
<ClCompile Include="FIRFilter.cpp" />
<ClCompile Include="InterpolateCubic.cpp" />
<ClCompile Include="InterpolateLinear.cpp" />
<ClCompile Include="InterpolateShannon.cpp" />
<ClCompile Include="mmx_optimized.cpp" />
<ClCompile Include="PeakFinder.cpp" />
<ClCompile Include="RateTransposer.cpp" />
<ClCompile Include="SoundTouch.cpp" />
<ClCompile Include="sse_optimized.cpp" />
<ClCompile Include="TDStretch.cpp" />
</ItemGroup>
<ItemGroup>
<ClInclude Include="AAFilter.h" />
<ClInclude Include="BPMDetect.h" />
<ClInclude Include="cpu_detect.h" />
<ClInclude Include="FIFOSampleBuffer.h" />
<ClInclude Include="FIFOSamplePipe.h" />
<ClInclude Include="FIRFilter.h" />
<ClInclude Include="InterpolateCubic.h" />
<ClInclude Include="InterpolateLinear.h" />
<ClInclude Include="InterpolateShannon.h" />
<ClInclude Include="PeakFinder.h" />
<ClInclude Include="RateTransposer.h" />
<ClInclude Include="SoundTouch.h" />
<ClInclude Include="STTypes.h" />
<ClInclude Include="TDStretch.h" />
</ItemGroup>
<ItemGroup>
<Text Include="CMakeLists.txt" />
</ItemGroup>
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />
<ImportGroup Label="ExtensionTargets">
</ImportGroup>
</Project>

1087
Externals/soundtouch/TDStretch.cpp vendored
View File

File diff suppressed because it is too large Load Diff

279
Externals/soundtouch/TDStretch.h vendored
View File

@ -1,279 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo
/// while maintaining the original pitch by using a time domain WSOLA-like method
/// with several performance-increasing tweaks.
///
/// Note : MMX/SSE optimized functions reside in separate, platform-specific files
/// 'mmx_optimized.cpp' and 'sse_optimized.cpp'
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef TDStretch_H
#define TDStretch_H
#include <stddef.h>
#include "STTypes.h"
#include "RateTransposer.h"
#include "FIFOSamplePipe.h"
namespace soundtouch
{
/// Default values for sound processing parameters:
/// Notice that the default parameters are tuned for contemporary popular music
/// processing. For speech processing applications these parameters suit better:
/// #define DEFAULT_SEQUENCE_MS 40
/// #define DEFAULT_SEEKWINDOW_MS 15
/// #define DEFAULT_OVERLAP_MS 8
///
/// Default length of a single processing sequence, in milliseconds. This determines to how
/// long sequences the original sound is chopped in the time-stretch algorithm.
///
/// The larger this value is, the lesser sequences are used in processing. In principle
/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo
/// and vice versa.
///
/// Increasing this value reduces computational burden & vice versa.
//#define DEFAULT_SEQUENCE_MS 40
#define DEFAULT_SEQUENCE_MS USE_AUTO_SEQUENCE_LEN
/// Giving this value for the sequence length sets automatic parameter value
/// according to tempo setting (recommended)
#define USE_AUTO_SEQUENCE_LEN 0
/// Seeking window default length in milliseconds for algorithm that finds the best possible
/// overlapping location. This determines from how wide window the algorithm may look for an
/// optimal joining location when mixing the sound sequences back together.
///
/// The bigger this window setting is, the higher the possibility to find a better mixing
/// position will become, but at the same time large values may cause a "drifting" artifact
/// because consequent sequences will be taken at more uneven intervals.
///
/// If there's a disturbing artifact that sounds as if a constant frequency was drifting
/// around, try reducing this setting.
///
/// Increasing this value increases computational burden & vice versa.
//#define DEFAULT_SEEKWINDOW_MS 15
#define DEFAULT_SEEKWINDOW_MS USE_AUTO_SEEKWINDOW_LEN
/// Giving this value for the seek window length sets automatic parameter value
/// according to tempo setting (recommended)
#define USE_AUTO_SEEKWINDOW_LEN 0
/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together,
/// to form a continuous sound stream, this parameter defines over how long period the two
/// consecutive sequences are let to overlap each other.
///
/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting
/// by a large amount, you might wish to try a smaller value on this.
///
/// Increasing this value increases computational burden & vice versa.
#define DEFAULT_OVERLAP_MS 8
/// Class that does the time-stretch (tempo change) effect for the processed
/// sound.
class TDStretch : public FIFOProcessor
{
protected:
int channels;
int sampleReq;
int overlapLength;
int seekLength;
int seekWindowLength;
int overlapDividerBitsNorm;
int overlapDividerBitsPure;
int slopingDivider;
int sampleRate;
int sequenceMs;
int seekWindowMs;
int overlapMs;
unsigned long maxnorm;
float maxnormf;
double tempo;
double nominalSkip;
double skipFract;
bool bQuickSeek;
bool bAutoSeqSetting;
bool bAutoSeekSetting;
bool isBeginning;
SAMPLETYPE *pMidBuffer;
SAMPLETYPE *pMidBufferUnaligned;
FIFOSampleBuffer outputBuffer;
FIFOSampleBuffer inputBuffer;
void acceptNewOverlapLength(int newOverlapLength);
virtual void clearCrossCorrState();
void calculateOverlapLength(int overlapMs);
virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm);
virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos);
virtual int seekBestOverlapPosition(const SAMPLETYPE *refPos);
virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const;
virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const;
void clearMidBuffer();
void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const;
void calcSeqParameters();
void adaptNormalizer();
/// Changes the tempo of the given sound samples.
/// 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 'set_returnBuffer_size' function.
void processSamples();
public:
TDStretch();
virtual ~TDStretch() override;
/// 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.
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 feature set depending on if the CPU
/// supports MMX/SSE/etc extensions.
static TDStretch *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the input buffer object
FIFOSamplePipe *getInput() { return &inputBuffer; };
/// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower
/// tempo, larger faster tempo.
void setTempo(double newTempo);
/// Returns nonzero if there aren't any samples available for outputting.
virtual void clear() override;
/// Clears the input buffer
void clearInput();
/// Sets the number of channels, 1 = mono, 2 = stereo
void setChannels(int numChannels);
/// Enables/disables the quick position seeking algorithm. Zero to disable,
/// nonzero to enable
void enableQuickSeek(bool enable);
/// Returns nonzero if the quick seeking algorithm is enabled.
bool isQuickSeekEnabled() const;
/// Sets routine control parameters. These control are certain time constants
/// defining how the sound is stretched to the desired duration.
//
/// 'sampleRate' = sample rate of the sound
/// 'sequenceMS' = one processing sequence length in milliseconds
/// 'seekwindowMS' = seeking window length for scanning the best overlapping
/// position
/// 'overlapMS' = overlapping length
void setParameters(int sampleRate, ///< Samplerate of sound being processed (Hz)
int sequenceMS = -1, ///< Single processing sequence length (ms)
int seekwindowMS = -1, ///< Offset seeking window length (ms)
int overlapMS = -1 ///< Sequence overlapping length (ms)
);
/// Get routine control parameters, see setParameters() function.
/// Any of the parameters to this function can be nullptr, in such case corresponding parameter
/// value isn't returned.
void getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const;
/// Adds 'numsamples' pcs of samples from the 'samples' memory position into
/// the input of the object.
virtual void putSamples(
const SAMPLETYPE *samples, ///< Input sample data
uint numSamples ///< Number of samples in 'samples' so that one sample
///< contains both channels if stereo
) override;
/// return nominal input sample requirement for triggering a processing batch
int getInputSampleReq() const
{
return (int)(nominalSkip + 0.5);
}
/// return nominal output sample amount when running a processing batch
int getOutputBatchSize() const
{
return seekWindowLength - overlapLength;
}
/// return approximate initial input-output latency
int getLatency() const
{
return sampleReq;
}
};
// Implementation-specific class declarations:
#ifdef SOUNDTOUCH_ALLOW_MMX
/// Class that implements MMX optimized routines for 16bit integer samples type.
class TDStretchMMX : public TDStretch
{
protected:
double calcCrossCorr(const short *mixingPos, const short *compare, double &norm) override;
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) override;
virtual void overlapStereo(short *output, const short *input) const override;
virtual void clearCrossCorrState() override;
};
#endif /// SOUNDTOUCH_ALLOW_MMX
#ifdef SOUNDTOUCH_ALLOW_SSE
/// Class that implements SSE optimized routines for floating point samples type.
class TDStretchSSE : public TDStretch
{
protected:
double calcCrossCorr(const float *mixingPos, const float *compare, double &norm) override;
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) override;
};
#endif /// SOUNDTOUCH_ALLOW_SSE
}
#endif /// TDStretch_H

57
Externals/soundtouch/cpu_detect.h vendored
View File

@ -1,57 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// A header file for detecting the Intel MMX instructions set extension.
///
/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the
/// routine implementations for x86 Windows, x86 gnu version and non-x86
/// platforms, respectively.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _CPU_DETECT_H_
#define _CPU_DETECT_H_
#include "STTypes.h"
#define SUPPORT_MMX 0x0001
#define SUPPORT_3DNOW 0x0002
#define SUPPORT_ALTIVEC 0x0004
#define SUPPORT_SSE 0x0008
#define SUPPORT_SSE2 0x0010
using namespace soundtouch;
/// Checks which instruction set extensions are supported by the CPU.
///
/// \return A bitmask of supported extensions, see SUPPORT_... defines.
uint detectCPUextensions(void);
/// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint wDisableMask);
#endif // _CPU_DETECT_H_

130
Externals/soundtouch/cpu_detect_x86.cpp vendored
View File

@ -1,130 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Generic version of the x86 CPU extension detection routine.
///
/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp'
/// for the Microsoft compiler version.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
#if defined(__GNUC__) && defined(__i386__)
// gcc
#include "cpuid.h"
#elif defined(_M_IX86)
// windows non-gcc
#include <intrin.h>
#endif
#define bit_MMX (1 << 23)
#define bit_SSE (1 << 25)
#define bit_SSE2 (1 << 26)
#endif
//////////////////////////////////////////////////////////////////////////////
//
// processor instructions extension detection routines
//
//////////////////////////////////////////////////////////////////////////////
// Flag variable indicating whick ISA extensions are disabled (for debugging)
static uint _dwDisabledISA = 0x00; // 0xffffffff; //<- use this to disable all extensions
// Disables given set of instruction extensions. See SUPPORT_... defines.
void disableExtensions(uint dwDisableMask)
{
_dwDisabledISA = dwDisableMask;
}
/// Checks which instruction set extensions are supported by the CPU.
uint detectCPUextensions(void)
{
/// If building for a 64bit system (no Itanium) and the user wants optimizations.
/// Return the OR of SUPPORT_{MMX,SSE,SSE2}. 11001 or 0x19.
/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
#if ((defined(__GNUC__) && defined(__x86_64__)) \
|| defined(_M_X64)) \
&& defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
return 0x19 & ~_dwDisabledISA;
/// If building for a 32bit system and the user wants optimizations.
/// Keep the _dwDisabledISA test (2 more operations, could be eliminated).
#elif ((defined(__GNUC__) && defined(__i386__)) \
|| defined(_M_IX86)) \
&& defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS)
if (_dwDisabledISA == 0xffffffff) return 0;
uint res = 0;
#if defined(__GNUC__)
// GCC version of cpuid. Requires GCC 4.3.0 or later for __cpuid intrinsic support.
uint eax, ebx, ecx, edx; // unsigned int is the standard type. uint is defined by the compiler and not guaranteed to be portable.
// Check if no cpuid support.
if (!__get_cpuid (1, &eax, &ebx, &ecx, &edx)) return 0; // always disable extensions.
if (edx & bit_MMX) res = res | SUPPORT_MMX;
if (edx & bit_SSE) res = res | SUPPORT_SSE;
if (edx & bit_SSE2) res = res | SUPPORT_SSE2;
#else
// Window / VS version of cpuid. Notice that Visual Studio 2005 or later required
// for __cpuid intrinsic support.
int reg[4] = {-1};
// Check if no cpuid support.
__cpuid(reg,0);
if ((unsigned int)reg[0] == 0) return 0; // always disable extensions.
__cpuid(reg,1);
if ((unsigned int)reg[3] & bit_MMX) res = res | SUPPORT_MMX;
if ((unsigned int)reg[3] & bit_SSE) res = res | SUPPORT_SSE;
if ((unsigned int)reg[3] & bit_SSE2) res = res | SUPPORT_SSE2;
#endif
return res & ~_dwDisabledISA;
#else
/// One of these is true:
/// 1) We don't want optimizations.
/// 2) Using an unsupported compiler.
/// 3) Running on a non-x86 platform.
return 0;
#endif
}

13
Externals/soundtouch/exports.props vendored
View File

@ -1,13 +0,0 @@
<?xml version="1.0" encoding="utf-8"?>
<Project>
<ItemDefinitionGroup>
<ClCompile>
<AdditionalIncludeDirectories>$(ExternalsDir)soundtouch;%(AdditionalIncludeDirectories)</AdditionalIncludeDirectories>
</ClCompile>
</ItemDefinitionGroup>
<ItemGroup>
<ProjectReference Include="$(ExternalsDir)soundtouch\SoundTouch.vcxproj">
<Project>{ec082900-b4d8-42e9-9663-77f02f6936ae}</Project>
</ProjectReference>
</ItemGroup>
</Project>

396
Externals/soundtouch/mmx_optimized.cpp vendored
View File

@ -1,396 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// MMX optimized routines. All MMX optimized functions have been gathered into
/// this single source code file, regardless to their class or original source
/// code file, in order to ease porting the library to other compiler and
/// processor platforms.
///
/// The MMX-optimizations are programmed using MMX compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support compiler intrinsic syntax. The update
/// is available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "STTypes.h"
#ifdef SOUNDTOUCH_ALLOW_MMX
// MMX routines available only with integer sample type
using namespace soundtouch;
//////////////////////////////////////////////////////////////////////////////
//
// implementation of MMX optimized functions of class 'TDStretchMMX'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
#include <mmintrin.h>
#include <limits.h>
#include <math.h>
// Calculates cross correlation of two buffers
double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm)
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
__m64 accu, normaccu;
long corr, norm;
int i;
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBitsNorm);
normaccu = accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
// during each round for improved CPU-level parallellization.
for (i = 0; i < channels * overlapLength / 16; i ++)
{
__m64 temp, temp2;
// dictionary of instructions:
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
// _mm_add_pi32 : 2*32bit add
// _m_psrad : 32bit right-shift
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter));
accu = _mm_add_pi32(accu, temp);
normaccu = _mm_add_pi32(normaccu, temp2);
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter));
accu = _mm_add_pi32(accu, temp);
normaccu = _mm_add_pi32(normaccu, temp2);
pVec1 += 4;
pVec2 += 4;
}
// copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
// and finally store the result into the variable "corr"
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
corr = _m_to_int(accu);
normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32));
norm = _m_to_int(normaccu);
// Clear MMS state
_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)
{
maxnorm = norm;
}
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
dnorm = (double)norm;
return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm);
// Note: Warning about the missing EMMS instruction is harmless
// as it'll be called elsewhere.
}
/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value
double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm)
{
const __m64 *pVec1, *pVec2;
__m64 shifter;
__m64 accu;
long corr, lnorm;
int i;
// cancel first normalizer tap from previous round
lnorm = 0;
for (i = 1; i <= channels; i ++)
{
lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBitsNorm;
}
pVec1 = (__m64*)pV1;
pVec2 = (__m64*)pV2;
shifter = _m_from_int(overlapDividerBitsNorm);
accu = _mm_setzero_si64();
// Process 4 parallel sets of 2 * stereo samples or 4 * mono samples
// during each round for improved CPU-level parallellization.
for (i = 0; i < channels * overlapLength / 16; i ++)
{
__m64 temp;
// dictionary of instructions:
// _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3]
// _mm_add_pi32 : 2*32bit add
// _m_psrad : 32bit right-shift
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter));
accu = _mm_add_pi32(accu, temp);
temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter),
_mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter));
accu = _mm_add_pi32(accu, temp);
pVec1 += 4;
pVec2 += 4;
}
// copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1
// and finally store the result into the variable "corr"
accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32));
corr = _m_to_int(accu);
// Clear MMS state
_m_empty();
// update normalizer with last samples of this round
pV1 = (short *)pVec1;
for (int j = 1; j <= channels; j ++)
{
lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBitsNorm;
}
dnorm += (double)lnorm;
if (lnorm > (long)maxnorm)
{
maxnorm = lnorm;
}
// Normalize result by dividing by sqrt(norm) - this step is easiest
// done using floating point operation
return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm);
}
void TDStretchMMX::clearCrossCorrState()
{
// Clear MMS state
_m_empty();
//_asm EMMS;
}
// MMX-optimized version of the function overlapStereo
void TDStretchMMX::overlapStereo(short *output, const short *input) const
{
const __m64 *pVinput, *pVMidBuf;
__m64 *pVdest;
__m64 mix1, mix2, adder, shifter;
int i;
pVinput = (const __m64*)input;
pVMidBuf = (const __m64*)pMidBuffer;
pVdest = (__m64*)output;
// mix1 = mixer values for 1st stereo sample
// mix1 = mixer values for 2nd stereo sample
// adder = adder for updating mixer values after each round
mix1 = _mm_set_pi16(0, overlapLength, 0, overlapLength);
adder = _mm_set_pi16(1, -1, 1, -1);
mix2 = _mm_add_pi16(mix1, adder);
adder = _mm_add_pi16(adder, adder);
// Overlaplength-division by shifter. "+1" is to account for "-1" deduced in
// overlapDividerBits calculation earlier.
shifter = _m_from_int(overlapDividerBitsPure + 1);
for (i = 0; i < overlapLength / 4; i ++)
{
__m64 temp1, temp2;
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r
temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r
// temp = (temp .* mix) >> shifter
temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
// update mix += adder
mix1 = _mm_add_pi16(mix1, adder);
mix2 = _mm_add_pi16(mix2, adder);
// --- second round begins here ---
// load & shuffle data so that input & mixbuffer data samples are paired
temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]); // = i2l m2l i2r m2r
temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]); // = i3l m3l i3r m3r
// temp = (temp .* mix) >> shifter
temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter);
temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter);
pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit
// update mix += adder
mix1 = _mm_add_pi16(mix1, adder);
mix2 = _mm_add_pi16(mix2, adder);
pVinput += 2;
pVMidBuf += 2;
pVdest += 2;
}
_m_empty(); // clear MMS state
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of MMX optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilterMMX::FIRFilterMMX() : FIRFilter()
{
filterCoeffsAlign = nullptr;
filterCoeffsUnalign = nullptr;
}
FIRFilterMMX::~FIRFilterMMX()
{
delete[] filterCoeffsUnalign;
}
// (overloaded) Calculates filter coefficients for MMX routine
void FIRFilterMMX::setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new short[2 * newLength + 8];
filterCoeffsAlign = (short *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
// rearrange the filter coefficients for mmx routines
for (i = 0;i < length; i += 4)
{
filterCoeffsAlign[2 * i + 0] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 1] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 2] = coeffs[i + 0];
filterCoeffsAlign[2 * i + 3] = coeffs[i + 2];
filterCoeffsAlign[2 * i + 4] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 5] = coeffs[i + 3];
filterCoeffsAlign[2 * i + 6] = coeffs[i + 1];
filterCoeffsAlign[2 * i + 7] = coeffs[i + 3];
}
}
// mmx-optimized version of the filter routine for stereo sound
uint FIRFilterMMX::evaluateFilterStereo(short *dest, const short *src, uint numSamples) const
{
// Create stack copies of the needed member variables for asm routines :
uint i, j;
__m64 *pVdest = (__m64*)dest;
if (length < 2) return 0;
for (i = 0; i < (numSamples - length) / 2; i ++)
{
__m64 accu1;
__m64 accu2;
const __m64 *pVsrc = (const __m64*)src;
const __m64 *pVfilter = (const __m64*)filterCoeffsAlign;
accu1 = accu2 = _mm_setzero_si64();
for (j = 0; j < lengthDiv8 * 2; j ++)
{
__m64 temp1, temp2;
temp1 = _mm_unpacklo_pi16(pVsrc[0], pVsrc[1]); // = l2 l0 r2 r0
temp2 = _mm_unpackhi_pi16(pVsrc[0], pVsrc[1]); // = l3 l1 r3 r1
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp1, pVfilter[0])); // += l2*f2+l0*f0 r2*f2+r0*f0
accu1 = _mm_add_pi32(accu1, _mm_madd_pi16(temp2, pVfilter[1])); // += l3*f3+l1*f1 r3*f3+r1*f1
temp1 = _mm_unpacklo_pi16(pVsrc[1], pVsrc[2]); // = l4 l2 r4 r2
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp2, pVfilter[0])); // += l3*f2+l1*f0 r3*f2+r1*f0
accu2 = _mm_add_pi32(accu2, _mm_madd_pi16(temp1, pVfilter[1])); // += l4*f3+l2*f1 r4*f3+r2*f1
// accu1 += l2*f2+l0*f0 r2*f2+r0*f0
// += l3*f3+l1*f1 r3*f3+r1*f1
// accu2 += l3*f2+l1*f0 r3*f2+r1*f0
// l4*f3+l2*f1 r4*f3+r2*f1
pVfilter += 2;
pVsrc += 2;
}
// accu >>= resultDivFactor
accu1 = _mm_srai_pi32(accu1, resultDivFactor);
accu2 = _mm_srai_pi32(accu2, resultDivFactor);
// pack 2*2*32bits => 4*16 bits
pVdest[0] = _mm_packs_pi32(accu1, accu2);
src += 4;
pVdest ++;
}
_m_empty(); // clear emms state
return (numSamples & 0xfffffffe) - length;
}
#else
// workaround to not complain about empty module
bool _dontcomplain_mmx_empty;
#endif // SOUNDTOUCH_ALLOW_MMX

365
Externals/soundtouch/sse_optimized.cpp vendored
View File

@ -1,365 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE
/// optimized functions have been gathered into this single source
/// code file, regardless to their class or original source code file, in order
/// to ease porting the library to other compiler and processor platforms.
///
/// The SSE-optimizations are programmed using SSE compiler intrinsics that
/// are supported both by Microsoft Visual C++ and GCC compilers, so this file
/// should compile with both toolsets.
///
/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++
/// 6.0 processor pack" update to support SSE instruction set. The update is
/// available for download at Microsoft Developers Network, see here:
/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx
///
/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and
/// perform a search with keywords "processor pack".
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include "cpu_detect.h"
#include "STTypes.h"
using namespace soundtouch;
#ifdef SOUNDTOUCH_ALLOW_SSE
// SSE routines available only with float sample type
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'TDStretchSSE'
//
//////////////////////////////////////////////////////////////////////////////
#include "TDStretch.h"
#include <xmmintrin.h>
#include <math.h>
// Calculates cross correlation of two buffers
double TDStretchSSE::calcCrossCorr(const float *pV1, const float *pV2, double &anorm)
{
int i;
const float *pVec1;
const __m128 *pVec2;
__m128 vSum, vNorm;
// Note. It means a major slow-down if the routine needs to tolerate
// unaligned __m128 memory accesses. It's way faster if we can skip
// unaligned slots and use _mm_load_ps instruction instead of _mm_loadu_ps.
// This can mean up to ~ 10-fold difference (incl. part of which is
// due to skipping every second round for stereo sound though).
//
// Compile-time define SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION is provided
// for choosing if this little cheating is allowed.
#ifdef ST_SIMD_AVOID_UNALIGNED
// Little cheating allowed, return valid correlation only for
// aligned locations, meaning every second round for stereo sound.
#define _MM_LOAD _mm_load_ps
if (((ulongptr)pV1) & 15) return -1e50; // skip unaligned locations
#else
// No cheating allowed, use unaligned load & take the resulting
// performance hit.
#define _MM_LOAD _mm_loadu_ps
#endif
// ensure overlapLength is divisible by 8
assert((overlapLength % 8) == 0);
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
// Note: pV2 _must_ be aligned to 16-bit boundary, pV1 need not.
pVec1 = (const float*)pV1;
pVec2 = (const __m128*)pV2;
vSum = vNorm = _mm_setzero_ps();
// Unroll the loop by factor of 4 * 4 operations. Use same routine for
// stereo & mono, for mono it just means twice the amount of unrolling.
for (i = 0; i < channels * overlapLength / 16; i ++)
{
__m128 vTemp;
// vSum += pV1[0..3] * pV2[0..3]
vTemp = _MM_LOAD(pVec1);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp ,pVec2[0]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
// vSum += pV1[4..7] * pV2[4..7]
vTemp = _MM_LOAD(pVec1 + 4);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[1]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
// vSum += pV1[8..11] * pV2[8..11]
vTemp = _MM_LOAD(pVec1 + 8);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[2]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
// vSum += pV1[12..15] * pV2[12..15]
vTemp = _MM_LOAD(pVec1 + 12);
vSum = _mm_add_ps(vSum, _mm_mul_ps(vTemp, pVec2[3]));
vNorm = _mm_add_ps(vNorm, _mm_mul_ps(vTemp ,vTemp));
pVec1 += 16;
pVec2 += 4;
}
// return value = vSum[0] + vSum[1] + vSum[2] + vSum[3]
float *pvNorm = (float*)&vNorm;
float norm = (pvNorm[0] + pvNorm[1] + pvNorm[2] + pvNorm[3]);
anorm = norm;
float *pvSum = (float*)&vSum;
return (double)(pvSum[0] + pvSum[1] + pvSum[2] + pvSum[3]) / sqrt(norm < 1e-9 ? 1.0 : norm);
/* This is approximately corresponding routine in C-language yet without normalization:
double corr, norm;
uint i;
// Calculates the cross-correlation value between 'pV1' and 'pV2' vectors
corr = norm = 0.0;
for (i = 0; i < channels * overlapLength / 16; i ++)
{
corr += pV1[0] * pV2[0] +
pV1[1] * pV2[1] +
pV1[2] * pV2[2] +
pV1[3] * pV2[3] +
pV1[4] * pV2[4] +
pV1[5] * pV2[5] +
pV1[6] * pV2[6] +
pV1[7] * pV2[7] +
pV1[8] * pV2[8] +
pV1[9] * pV2[9] +
pV1[10] * pV2[10] +
pV1[11] * pV2[11] +
pV1[12] * pV2[12] +
pV1[13] * pV2[13] +
pV1[14] * pV2[14] +
pV1[15] * pV2[15];
for (j = 0; j < 15; j ++) norm += pV1[j] * pV1[j];
pV1 += 16;
pV2 += 16;
}
return corr / sqrt(norm);
*/
}
double TDStretchSSE::calcCrossCorrAccumulate(const float *pV1, const float *pV2, double &norm)
{
// call usual calcCrossCorr function because SSE does not show big benefit of
// accumulating "norm" value, and also the "norm" rolling algorithm would get
// complicated due to SSE-specific alignment-vs-nonexact correlation rules.
return calcCrossCorr(pV1, pV2, norm);
}
//////////////////////////////////////////////////////////////////////////////
//
// implementation of SSE optimized functions of class 'FIRFilter'
//
//////////////////////////////////////////////////////////////////////////////
#include "FIRFilter.h"
FIRFilterSSE::FIRFilterSSE() : FIRFilter()
{
filterCoeffsAlign = nullptr;
filterCoeffsUnalign = nullptr;
}
FIRFilterSSE::~FIRFilterSSE()
{
delete[] filterCoeffsUnalign;
filterCoeffsAlign = nullptr;
filterCoeffsUnalign = nullptr;
}
// (overloaded) Calculates filter coefficients for SSE routine
void FIRFilterSSE::setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor)
{
uint i;
float fDivider;
FIRFilter::setCoefficients(coeffs, newLength, uResultDivFactor);
// Scale the filter coefficients so that it won't be necessary to scale the filtering result
// also rearrange coefficients suitably for SSE
// Ensure that filter coeffs array is aligned to 16-byte boundary
delete[] filterCoeffsUnalign;
filterCoeffsUnalign = new float[2 * newLength + 4];
filterCoeffsAlign = (float *)SOUNDTOUCH_ALIGN_POINTER_16(filterCoeffsUnalign);
fDivider = (float)resultDivider;
// rearrange the filter coefficients for mmx routines
for (i = 0; i < newLength; i ++)
{
filterCoeffsAlign[2 * i + 0] =
filterCoeffsAlign[2 * i + 1] = coeffs[i + 0] / fDivider;
}
}
// SSE-optimized version of the filter routine for stereo sound
uint FIRFilterSSE::evaluateFilterStereo(float *dest, const float *source, uint numSamples) const
{
int count = (int)((numSamples - length) & (uint)-2);
int j;
assert(count % 2 == 0);
if (count < 2) return 0;
assert(source != nullptr);
assert(dest != nullptr);
assert((length % 8) == 0);
assert(filterCoeffsAlign != nullptr);
assert(((ulongptr)filterCoeffsAlign) % 16 == 0);
// filter is evaluated for two stereo samples with each iteration, thus use of 'j += 2'
#pragma omp parallel for
for (j = 0; j < count; j += 2)
{
const float *pSrc;
float *pDest;
const __m128 *pFil;
__m128 sum1, sum2;
uint i;
pSrc = (const float*)source + j * 2; // source audio data
pDest = dest + j * 2; // destination audio data
pFil = (const __m128*)filterCoeffsAlign; // filter coefficients. NOTE: Assumes coefficients
// are aligned to 16-byte boundary
sum1 = sum2 = _mm_setzero_ps();
for (i = 0; i < length / 8; i ++)
{
// Unroll loop for efficiency & calculate filter for 2*2 stereo samples
// at each pass
// sum1 is accu for 2*2 filtered stereo sound data at the primary sound data offset
// sum2 is accu for 2*2 filtered stereo sound data for the next sound sample offset.
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc) , pFil[0]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 2), pFil[0]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 4), pFil[1]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 6), pFil[1]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 8) , pFil[2]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 10), pFil[2]));
sum1 = _mm_add_ps(sum1, _mm_mul_ps(_mm_loadu_ps(pSrc + 12), pFil[3]));
sum2 = _mm_add_ps(sum2, _mm_mul_ps(_mm_loadu_ps(pSrc + 14), pFil[3]));
pSrc += 16;
pFil += 4;
}
// Now sum1 and sum2 both have a filtered 2-channel sample each, but we still need
// to sum the two hi- and lo-floats of these registers together.
// post-shuffle & add the filtered values and store to dest.
_mm_storeu_ps(pDest, _mm_add_ps(
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(1,0,3,2)), // s2_1 s2_0 s1_3 s1_2
_mm_shuffle_ps(sum1, sum2, _MM_SHUFFLE(3,2,1,0)) // s2_3 s2_2 s1_1 s1_0
));
}
// Ideas for further improvement:
// 1. If it could be guaranteed that 'source' were always aligned to 16-byte
// boundary, a faster aligned '_mm_load_ps' instruction could be used.
// 2. If it could be guaranteed that 'dest' were always aligned to 16-byte
// boundary, a faster '_mm_store_ps' instruction could be used.
return (uint)count;
/* original routine in C-language. please notice the C-version has differently
organized coefficients though.
double suml1, suml2;
double sumr1, sumr2;
uint i, j;
for (j = 0; j < count; j += 2)
{
const float *ptr;
const float *pFil;
suml1 = sumr1 = 0.0;
suml2 = sumr2 = 0.0;
ptr = src;
pFil = filterCoeffs;
for (i = 0; i < lengthLocal; i ++)
{
// unroll loop for efficiency.
suml1 += ptr[0] * pFil[0] +
ptr[2] * pFil[2] +
ptr[4] * pFil[4] +
ptr[6] * pFil[6];
sumr1 += ptr[1] * pFil[1] +
ptr[3] * pFil[3] +
ptr[5] * pFil[5] +
ptr[7] * pFil[7];
suml2 += ptr[8] * pFil[0] +
ptr[10] * pFil[2] +
ptr[12] * pFil[4] +
ptr[14] * pFil[6];
sumr2 += ptr[9] * pFil[1] +
ptr[11] * pFil[3] +
ptr[13] * pFil[5] +
ptr[15] * pFil[7];
ptr += 16;
pFil += 8;
}
dest[0] = (float)suml1;
dest[1] = (float)sumr1;
dest[2] = (float)suml2;
dest[3] = (float)sumr2;
src += 4;
dest += 4;
}
*/
}
#endif // SOUNDTOUCH_ALLOW_SSE

2
Installer/dolphin-emu.spec
View File

@ -60,7 +60,6 @@ BuildRequires: openal-soft-devel
BuildRequires: mbedtls-devel
BuildRequires: SDL2-devel
BuildRequires: SFML-devel
BuildRequires: soundtouch-devel
%endif
%if 0%{?suse_version}
@ -69,7 +68,6 @@ BuildRequires: libSDL2-devel
BuildRequires: lzo-devel
BuildRequires: openal-devel
BuildRequires: sfml-devel
BuildRequires: soundtouch-devel
BuildRequires: update-desktop-files
%endif

2
Source/Android/app/src/main/java/org/dolphinemu/dolphinemu/features/settings/model/BooleanSetting.kt
View File

@ -30,7 +30,7 @@ enum class BooleanSetting(
"OverrideRegionSettings",
false
),
MAIN_AUDIO_STRETCH(Settings.FILE_DOLPHIN, Settings.SECTION_INI_CORE, "AudioStretch", false),
MAIN_AUDIO_FILL_GAPS(Settings.FILE_DOLPHIN, Settings.SECTION_INI_CORE, "AudioFillGaps", true),
MAIN_BBA_XLINK_CHAT_OSD(
Settings.FILE_DOLPHIN,
Settings.SECTION_INI_CORE,

6
Source/Android/app/src/main/java/org/dolphinemu/dolphinemu/features/settings/ui/SettingsFragmentPresenter.kt
View File

@ -533,9 +533,9 @@ class SettingsFragmentPresenter(
sl.add(
SwitchSetting(
context,
BooleanSetting.MAIN_AUDIO_STRETCH,
R.string.audio_stretch,
R.string.audio_stretch_description
BooleanSetting.MAIN_AUDIO_FILL_GAPS,
R.string.audio_fill_gaps,
R.string.audio_fill_gaps_description
)
)
sl.add(

19
Source/Android/app/src/main/res/values/strings.xml
View File

@ -132,7 +132,8 @@
<!-- SerialPort1 Subsetting Fragment -->
<string name="serialport1_submenu">Serial Port 1</string>
<string name="xlink_kai_guide_header">For setup instructions, <a href="https://www.teamxlink.co.uk/wiki/Dolphin">refer to this page.</a></string>
<string name="xlink_kai_guide_header">For setup instructions, <a href="https://www.teamxlink.co.uk/wiki/Dolphin">refer to this page.</a>
</string>
<string name="xlink_kai_bba_ip">XLink Kai IP Address/hostname</string>
<string name="xlink_kai_bba_ip_description">IP address or hostname of device running the XLink Kai client</string>
<string name="bba_tapserver_destination">Tapserver destination</string>
@ -181,8 +182,8 @@
<!-- Audio Settings -->
<string name="audio_submenu">Audio</string>
<string name="dsp_emulation_engine">DSP Emulation Engine</string>
<string name="audio_stretch">Audio Stretching</string>
<string name="audio_stretch_description">Stretches audio to reduce stuttering. Increases latency.</string>
<string name="audio_fill_gaps">Fill Audio Gaps</string>
<string name="audio_fill_gaps_description">Repeat existing audio during lag spikes to prevent stuttering. If unsure, leave this checked.</string>
<string name="audio_volume">Audio Volume</string>
<!-- Path Settings -->
@ -877,9 +878,15 @@ It can efficiently compress both junk data and encrypted Wii data.
<string name="about_branch">Branch:\n%s</string>
<string name="about_revision">Revision:\n%s</string>
<string name="about_dolphin_description">Dolphin is a free and open-source GameCube and Wii emulator.\n\nThis software should not be used to play games you do not legally own.</string>
<string name="about_website"><a href="https://dolphin-emu.org/">Website</a></string>
<string name="about_github"><a href="https://github.com/dolphin-emu/dolphin">GitHub</a></string>
<string name="about_support"><a href="https://forums.dolphin-emu.org/">Support</a></string>
<string name="about_website">
<a href="https://dolphin-emu.org/">Website</a>
</string>
<string name="about_github">
<a href="https://github.com/dolphin-emu/dolphin">GitHub</a>
</string>
<string name="about_support">
<a href="https://forums.dolphin-emu.org/">Support</a>
</string>
<string name="about_copyright_warning">\u00A9 2003–2024+ Dolphin Team. \u201cGameCube\u201d and \u201cWii\u201d are trademarks of Nintendo. Dolphin is not affiliated with Nintendo in any way.</string>
<string name="system_driver">System driver</string>
<string name="system_driver_desc">The GPU driver that is part of the OS.</string>

85
Source/Core/AudioCommon/AudioStretcher.cpp
View File

@ -1,85 +0,0 @@
// Copyright 2017 Dolphin Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#include "AudioCommon/AudioStretcher.h"
#include <algorithm>
#include <cmath>
#include <cstddef>
#include "Common/Logging/Log.h"
#include "Core/Config/MainSettings.h"
namespace AudioCommon
{
AudioStretcher::AudioStretcher(unsigned int sample_rate) : m_sample_rate(sample_rate)
{
m_sound_touch.setChannels(2);
m_sound_touch.setSampleRate(sample_rate);
m_sound_touch.setPitch(1.0);
m_sound_touch.setTempo(1.0);
}
void AudioStretcher::Clear()
{
m_sound_touch.clear();
}
void AudioStretcher::ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out)
{
const double time_delta = static_cast<double>(num_out) / m_sample_rate; // seconds
// We were given actual_samples number of samples, and num_samples were requested from us.
double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
const double max_latency = Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY);
const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio;
const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
if (backlog_fullness > 5.0)
{
// Too many samples in backlog: Don't push anymore on
num_in = 0;
}
// We ideally want the backlog to be about 50% full.
// This gives some headroom both ways to prevent underflow and overflow.
// We tweak current_ratio to encourage this.
constexpr double tweak_time_scale = 0.5; // seconds
current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
// This low-pass filter smoothes out variance in the calculated stretch ratio.
// The time-scale determines how responsive this filter is.
constexpr double lpf_time_scale = 1.0; // seconds
const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
// Place a lower limit of 10% speed. When a game boots up, there will be
// many silence samples. These do not need to be timestretched.
m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
m_sound_touch.setTempo(m_stretch_ratio);
DEBUG_LOG_FMT(AUDIO, "Audio stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in,
num_out, m_stretch_ratio, backlog_fullness, lpf_gain);
m_sound_touch.putSamples(in, num_in);
}
void AudioStretcher::GetStretchedSamples(short* out, unsigned int num_out)
{
const size_t samples_received = m_sound_touch.receiveSamples(out, num_out);
if (samples_received != 0)
{
m_last_stretched_sample[0] = out[samples_received * 2 - 2];
m_last_stretched_sample[1] = out[samples_received * 2 - 1];
}
// Perform padding if we've run out of samples.
for (size_t i = samples_received; i < num_out; i++)
{
out[i * 2 + 0] = m_last_stretched_sample[0];
out[i * 2 + 1] = m_last_stretched_sample[1];
}
}
} // namespace AudioCommon

27
Source/Core/AudioCommon/AudioStretcher.h
View File

@ -1,27 +0,0 @@
// Copyright 2017 Dolphin Emulator Project
// SPDX-License-Identifier: GPL-2.0-or-later
#pragma once
#include <array>
#include <SoundTouch.h>
namespace AudioCommon
{
class AudioStretcher
{
public:
explicit AudioStretcher(unsigned int sample_rate);
void ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out);
void GetStretchedSamples(short* out, unsigned int num_out);
void Clear();
private:
unsigned int m_sample_rate;
std::array<short, 2> m_last_stretched_sample = {};
soundtouch::SoundTouch m_sound_touch;
double m_stretch_ratio = 1.0;
};
} // namespace AudioCommon

3
Source/Core/AudioCommon/CMakeLists.txt
View File

@ -1,8 +1,6 @@
add_library(audiocommon
AudioCommon.cpp
AudioCommon.h
AudioStretcher.cpp
AudioStretcher.h
CubebStream.h
Enums.h
Mixer.cpp
@ -90,7 +88,6 @@ PUBLIC
common
PRIVATE
SoundTouch
FreeSurround)
if(ENABLE_CUBEB)

456
Source/Core/AudioCommon/Mixer.cpp
View File

@ -13,8 +13,8 @@
#include "Common/Logging/Log.h"
#include "Common/Swap.h"
#include "Core/Config/MainSettings.h"
#include "Core/ConfigManager.h"
#include "VideoCommon/PerformanceMetrics.h"
#include "Core/Core.h"
#include "Core/System.h"
static u32 DPL2QualityToFrameBlockSize(AudioCommon::DPL2Quality quality)
{
@ -31,8 +31,8 @@ static u32 DPL2QualityToFrameBlockSize(AudioCommon::DPL2Quality quality)
}
}
Mixer::Mixer(unsigned int BackendSampleRate)
: m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate),
Mixer::Mixer(u32 BackendSampleRate)
: m_output_sample_rate(BackendSampleRate),
m_surround_decoder(BackendSampleRate,
DPL2QualityToFrameBlockSize(Config::Get(Config::MAIN_DPL2_QUALITY)))
{
@ -58,177 +58,80 @@ void Mixer::DoState(PointerWrap& p)
}
// Executed from sound stream thread
unsigned int Mixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
bool consider_framelimit, float emulationspeed,
int timing_variance)
void Mixer::MixerFifo::Mix(s16* samples, std::size_t num_samples)
{
unsigned int currentSample = 0;
constexpr u32 half = 0x80000000;
// Cache access in non-volatile variable
// This is the only function changing the read value, so it's safe to
// cache it locally although it's written here.
// The writing pointer will be modified outside, but it will only increase,
// so we will just ignore new written data while interpolating.
// Without this cache, the compiler wouldn't be allowed to optimize the
// interpolation loop.
u32 indexR = m_indexR.load();
u32 indexW = m_indexW.load();
const u64 out_sample_rate = m_mixer->m_output_sample_rate;
u64 in_sample_rate = FIXED_SAMPLE_RATE_DIVIDEND / m_input_sample_rate_divisor;
// render numleft sample pairs to samples[]
// advance indexR with sample position
// remember fractional offset
const float emulation_speed = m_mixer->m_config_emulation_speed;
if (0 < emulation_speed && emulation_speed != 1.0)
in_sample_rate = static_cast<u64>(std::llround(in_sample_rate * emulation_speed));
float aid_sample_rate =
FIXED_SAMPLE_RATE_DIVIDEND / static_cast<float>(m_input_sample_rate_divisor);
if (consider_framelimit && emulationspeed > 0.0f)
const u32 index_jump = (in_sample_rate << GRANULE_BUFFER_FRAC_BITS) / (out_sample_rate);
const StereoPair volume{m_LVolume.load() / 256.0f, m_RVolume.load() / 256.0f};
while (num_samples-- > 0)
{
float numLeft = static_cast<float>(((indexW - indexR) & INDEX_MASK) / 2);
StereoPair sample = Granule::InterpStereoPair(m_front, m_back, m_current_index);
sample *= volume;
u32 low_watermark = (FIXED_SAMPLE_RATE_DIVIDEND * timing_variance) /
(static_cast<u64>(m_input_sample_rate_divisor) * 1000);
low_watermark = std::min(low_watermark, MAX_SAMPLES / 2);
sample.l += samples[0] + m_quantization_error.l;
samples[0] = ToShort(std::lround(sample.l));
m_quantization_error.l = std::clamp(sample.l - samples[0], -1.0f, 1.0f);
m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
float offset = (m_numLeftI - low_watermark) * CONTROL_FACTOR;
if (offset > MAX_FREQ_SHIFT)
offset = MAX_FREQ_SHIFT;
if (offset < -MAX_FREQ_SHIFT)
offset = -MAX_FREQ_SHIFT;
sample.r += samples[1] + m_quantization_error.r;
samples[1] = ToShort(std::lround(sample.r));
m_quantization_error.r = std::clamp(sample.r - samples[1], -1.0f, 1.0f);
aid_sample_rate = (aid_sample_rate + offset) * emulationspeed;
samples += 2;
m_current_index += index_jump;
if (m_current_index < half)
{
m_front = m_back;
Dequeue(&m_back);
m_current_index += half;
}
}
const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
s32 lvolume = m_LVolume.load();
s32 rvolume = m_RVolume.load();
const auto read_buffer = [this](auto index) {
return m_little_endian ? m_buffer[index] : Common::swap16(m_buffer[index]);
};
// TODO: consider a higher-quality resampling algorithm.
for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2)
{
u32 indexR2 = indexR + 2; // next sample
s16 l1 = read_buffer(indexR & INDEX_MASK); // current
s16 l2 = read_buffer(indexR2 & INDEX_MASK); // next
int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
sampleL = (sampleL * lvolume) >> 8;
sampleL += samples[currentSample + 1];
samples[currentSample + 1] = std::clamp(sampleL, -32767, 32767);
s16 r1 = read_buffer((indexR + 1) & INDEX_MASK); // current
s16 r2 = read_buffer((indexR2 + 1) & INDEX_MASK); // next
int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
sampleR = (sampleR * rvolume) >> 8;
sampleR += samples[currentSample];
samples[currentSample] = std::clamp(sampleR, -32767, 32767);
m_frac += ratio;
indexR += 2 * (u16)(m_frac >> 16);
m_frac &= 0xffff;
}
// Actual number of samples written to the buffer without padding.
unsigned int actual_sample_count = currentSample / 2;
// Padding
short s[2];
s[0] = read_buffer((indexR - 1) & INDEX_MASK);
s[1] = read_buffer((indexR - 2) & INDEX_MASK);
s[0] = (s[0] * rvolume) >> 8;
s[1] = (s[1] * lvolume) >> 8;
for (; currentSample < numSamples * 2; currentSample += 2)
{
int sampleR = std::clamp(s[0] + samples[currentSample + 0], -32767, 32767);
int sampleL = std::clamp(s[1] + samples[currentSample + 1], -32767, 32767);
samples[currentSample + 0] = sampleR;
samples[currentSample + 1] = sampleL;
}
// Flush cached variable
m_indexR.store(indexR);
return actual_sample_count;
}
unsigned int Mixer::Mix(short* samples, unsigned int num_samples)
std::size_t Mixer::Mix(s16* samples, std::size_t num_samples)
{
if (!samples)
return 0;
memset(samples, 0, num_samples * 2 * sizeof(short));
memset(samples, 0, num_samples * 2 * sizeof(s16));
// TODO: Determine how emulation speed will be used in audio
// const float emulation_speed = g_perf_metrics.GetSpeed();
const float emulation_speed = m_config_emulation_speed;
const int timing_variance = m_config_timing_variance;
if (m_config_audio_stretch)
{
unsigned int available_samples =
std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples());
ASSERT_MSG(AUDIO, available_samples <= MAX_SAMPLES,
"Audio stretching would overflow m_scratch_buffer: min({}, {}) -> {} > {} ({})",
m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples(),
available_samples, MAX_SAMPLES, num_samples);
m_scratch_buffer.fill(0);
m_dma_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
timing_variance);
m_streaming_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
timing_variance);
m_wiimote_speaker_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
timing_variance);
m_skylander_portal_mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
timing_variance);
for (auto& mixer : m_gba_mixers)
{
mixer.Mix(m_scratch_buffer.data(), available_samples, false, emulation_speed,
timing_variance);
}
if (!m_is_stretching)
{
m_stretcher.Clear();
m_is_stretching = true;
}
m_stretcher.ProcessSamples(m_scratch_buffer.data(), available_samples, num_samples);
m_stretcher.GetStretchedSamples(samples, num_samples);
}
else
{
m_dma_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
m_streaming_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
m_wiimote_speaker_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
m_skylander_portal_mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
for (auto& mixer : m_gba_mixers)
mixer.Mix(samples, num_samples, true, emulation_speed, timing_variance);
m_is_stretching = false;
}
m_dma_mixer.Mix(samples, num_samples);
m_streaming_mixer.Mix(samples, num_samples);
m_wiimote_speaker_mixer.Mix(samples, num_samples);
m_skylander_portal_mixer.Mix(samples, num_samples);
for (auto& mixer : m_gba_mixers)
mixer.Mix(samples, num_samples);
return num_samples;
}
unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
std::size_t Mixer::MixSurround(float* samples, std::size_t num_samples)
{
if (!num_samples)
return 0;
memset(samples, 0, num_samples * SURROUND_CHANNELS * sizeof(float));
size_t needed_frames = m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);
std::size_t needed_frames = m_surround_decoder.QueryFramesNeededForSurroundOutput(num_samples);
// Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads
// and writes.
ASSERT_MSG(AUDIO, needed_frames <= MAX_SAMPLES,
constexpr std::size_t max_samples = 0x8000;
ASSERT_MSG(AUDIO, needed_frames <= max_samples,
"needed_frames would overflow m_scratch_buffer: {} -> {} > {}", num_samples,
needed_frames, MAX_SAMPLES);
size_t available_frames = Mix(m_scratch_buffer.data(), static_cast<u32>(needed_frames));
needed_frames, max_samples);
std::array<s16, max_samples> buffer;
std::size_t available_frames = Mix(buffer.data(), static_cast<std::size_t>(needed_frames));
if (available_frames != needed_frames)
{
ERROR_LOG_FMT(AUDIO,
@ -237,71 +140,58 @@ unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
return 0;
}
m_surround_decoder.PutFrames(m_scratch_buffer.data(), needed_frames);
m_surround_decoder.PutFrames(buffer.data(), needed_frames);
m_surround_decoder.ReceiveFrames(samples, num_samples);
return num_samples;
}
void Mixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples)
void Mixer::MixerFifo::PushSamples(const s16* samples, std::size_t num_samples)
{
// Cache access in non-volatile variable
// indexR isn't allowed to cache in the audio throttling loop as it
// needs to get updates to not deadlock.
u32 indexW = m_indexW.load();
// Check if we have enough free space
// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
if (num_samples * 2 + ((indexW - m_indexR.load()) & INDEX_MASK) >= MAX_SAMPLES * 2)
return;
// AyuanX: Actual re-sampling work has been moved to sound thread
// to alleviate the workload on main thread
// and we simply store raw data here to make fast mem copy
int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
if (over_bytes > 0)
while (num_samples-- > 0)
{
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
}
else
{
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
}
const s16 l = m_little_endian ? samples[1] : Common::swap16(samples[1]);
const s16 r = m_little_endian ? samples[0] : Common::swap16(samples[0]);
m_indexW.fetch_add(num_samples * 2);
m_buffer[m_buffer_index] = StereoPair(l, r);
m_buffer_index = (m_buffer_index + 1) & GRANULE_BUFFER_MASK;
samples += 2;
if (m_buffer_index == 0 || m_buffer_index == m_buffer.size() / 2)
Enqueue(Granule(m_buffer, m_buffer_index));
}
}
void Mixer::PushSamples(const short* samples, unsigned int num_samples)
void Mixer::PushSamples(const s16* samples, std::size_t num_samples)
{
m_dma_mixer.PushSamples(samples, num_samples);
if (m_log_dsp_audio)
{
int sample_rate_divisor = m_dma_mixer.GetInputSampleRateDivisor();
const s32 sample_rate_divisor = m_dma_mixer.GetInputSampleRateDivisor();
auto volume = m_dma_mixer.GetVolume();
m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples, sample_rate_divisor, volume.first,
volume.second);
m_wave_writer_dsp.AddStereoSamplesBE(samples, static_cast<u32>(num_samples),
sample_rate_divisor, volume.first, volume.second);
}
}
void Mixer::PushStreamingSamples(const short* samples, unsigned int num_samples)
void Mixer::PushStreamingSamples(const s16* samples, std::size_t num_samples)
{
m_streaming_mixer.PushSamples(samples, num_samples);
if (m_log_dtk_audio)
{
int sample_rate_divisor = m_streaming_mixer.GetInputSampleRateDivisor();
const s32 sample_rate_divisor = m_streaming_mixer.GetInputSampleRateDivisor();
auto volume = m_streaming_mixer.GetVolume();
m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples, sample_rate_divisor, volume.first,
volume.second);
m_wave_writer_dtk.AddStereoSamplesBE(samples, static_cast<u32>(num_samples),
sample_rate_divisor, volume.first, volume.second);
}
}
void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
unsigned int sample_rate_divisor)
void Mixer::PushWiimoteSpeakerSamples(const s16* samples, std::size_t num_samples,
u32 sample_rate_divisor)
{
// Max 20 bytes/speaker report, may be 4-bit ADPCM so multiply by 2
static constexpr u32 MAX_SPEAKER_SAMPLES = 20 * 2;
std::array<short, MAX_SPEAKER_SAMPLES * 2> samples_stereo;
static constexpr std::size_t MAX_SPEAKER_SAMPLES = 20 * 2;
std::array<s16, MAX_SPEAKER_SAMPLES * 2> samples_stereo;
ASSERT_MSG(AUDIO, num_samples <= MAX_SPEAKER_SAMPLES,
"num_samples would overflow samples_stereo: {} > {}", num_samples,
@ -310,7 +200,7 @@ void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_sam
{
m_wiimote_speaker_mixer.SetInputSampleRateDivisor(sample_rate_divisor);
for (unsigned int i = 0; i < num_samples; ++i)
for (std::size_t i = 0; i < num_samples; ++i)
{
samples_stereo[i * 2] = samples[i];
samples_stereo[i * 2 + 1] = samples[i];
@ -320,12 +210,12 @@ void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_sam
}
}
void Mixer::PushSkylanderPortalSamples(const u8* samples, unsigned int num_samples)
void Mixer::PushSkylanderPortalSamples(const u8* samples, std::size_t num_samples)
{
// Skylander samples are always supplied as 64 bytes, 32 x 16 bit samples
// The portal speaker is 1 channel, so duplicate and play as stereo audio
static constexpr u32 MAX_PORTAL_SPEAKER_SAMPLES = 32;
std::array<short, MAX_PORTAL_SPEAKER_SAMPLES * 2> samples_stereo;
static constexpr std::size_t MAX_PORTAL_SPEAKER_SAMPLES = 32;
std::array<s16, MAX_PORTAL_SPEAKER_SAMPLES * 2> samples_stereo;
ASSERT_MSG(AUDIO, num_samples <= MAX_PORTAL_SPEAKER_SAMPLES,
"num_samples is not less or equal to 32: {} > {}", num_samples,
@ -333,7 +223,7 @@ void Mixer::PushSkylanderPortalSamples(const u8* samples, unsigned int num_sampl
if (num_samples <= MAX_PORTAL_SPEAKER_SAMPLES)
{
for (unsigned int i = 0; i < num_samples; ++i)
for (std::size_t i = 0; i < num_samples; ++i)
{
s16 sample = static_cast<u16>(samples[i * 2 + 1]) << 8 | static_cast<u16>(samples[i * 2]);
samples_stereo[i * 2] = sample;
@ -344,37 +234,38 @@ void Mixer::PushSkylanderPortalSamples(const u8* samples, unsigned int num_sampl
}
}
void Mixer::PushGBASamples(int device_number, const short* samples, unsigned int num_samples)
void Mixer::PushGBASamples(std::size_t device_number, const s16* samples, std::size_t num_samples)
{
m_gba_mixers[device_number].PushSamples(samples, num_samples);
}
void Mixer::SetDMAInputSampleRateDivisor(unsigned int rate_divisor)
void Mixer::SetDMAInputSampleRateDivisor(u32 rate_divisor)
{
m_dma_mixer.SetInputSampleRateDivisor(rate_divisor);
}
void Mixer::SetStreamInputSampleRateDivisor(unsigned int rate_divisor)
void Mixer::SetStreamInputSampleRateDivisor(u32 rate_divisor)
{
m_streaming_mixer.SetInputSampleRateDivisor(rate_divisor);
}
void Mixer::SetGBAInputSampleRateDivisors(int device_number, unsigned int rate_divisor)
void Mixer::SetGBAInputSampleRateDivisors(std::size_t device_number, u32 rate_divisor)
{
m_gba_mixers[device_number].SetInputSampleRateDivisor(rate_divisor);
}
void Mixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
void Mixer::SetStreamingVolume(u32 lvolume, u32 rvolume)
{
m_streaming_mixer.SetVolume(lvolume, rvolume);
m_streaming_mixer.SetVolume(std::clamp<u32>(lvolume, 0x00, 0xff),
std::clamp<u32>(rvolume, 0x00, 0xff));
}
void Mixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
void Mixer::SetWiimoteSpeakerVolume(u32 lvolume, u32 rvolume)
{
m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
}
void Mixer::SetGBAVolume(int device_number, unsigned int lvolume, unsigned int rvolume)
void Mixer::SetGBAVolume(std::size_t device_number, u32 lvolume, u32 rvolume)
{
m_gba_mixers[device_number].SetVolume(lvolume, rvolume);
}
@ -456,8 +347,7 @@ void Mixer::StopLogDSPAudio()
void Mixer::RefreshConfig()
{
m_config_emulation_speed = Config::Get(Config::MAIN_EMULATION_SPEED);
m_config_timing_variance = Config::Get(Config::MAIN_TIMING_VARIANCE);
m_config_audio_stretch = Config::Get(Config::MAIN_AUDIO_STRETCH);
m_audio_fill_gaps = Config::Get(Config::MAIN_AUDIO_FILL_GAPS);
}
void Mixer::MixerFifo::DoState(PointerWrap& p)
@ -467,17 +357,17 @@ void Mixer::MixerFifo::DoState(PointerWrap& p)
p.Do(m_RVolume);
}
void Mixer::MixerFifo::SetInputSampleRateDivisor(unsigned int rate_divisor)
void Mixer::MixerFifo::SetInputSampleRateDivisor(u32 rate_divisor)
{
m_input_sample_rate_divisor = rate_divisor;
}
unsigned int Mixer::MixerFifo::GetInputSampleRateDivisor() const
u32 Mixer::MixerFifo::GetInputSampleRateDivisor() const
{
return m_input_sample_rate_divisor;
}
void Mixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
void Mixer::MixerFifo::SetVolume(u32 lvolume, u32 rvolume)
{
m_LVolume.store(lvolume + (lvolume >> 7));
m_RVolume.store(rvolume + (rvolume >> 7));
@ -488,11 +378,161 @@ std::pair<s32, s32> Mixer::MixerFifo::GetVolume() const
return std::make_pair(m_LVolume.load(), m_RVolume.load());
}
unsigned int Mixer::MixerFifo::AvailableSamples() const
void Mixer::MixerFifo::Enqueue(const Granule& granule)
{
unsigned int samples_in_fifo = ((m_indexW.load() - m_indexR.load()) & INDEX_MASK) / 2;
if (samples_in_fifo <= 1)
return 0; // Mixer::MixerFifo::Mix always keeps one sample in the buffer.
return (samples_in_fifo - 1) * static_cast<u64>(m_mixer->m_sampleRate) *
m_input_sample_rate_divisor / FIXED_SAMPLE_RATE_DIVIDEND;
const std::size_t head = m_queue_head.load(std::memory_order_relaxed);
std::size_t next_head = (head + 1) % GRANULE_QUEUE_SIZE;
if (next_head == m_queue_tail.load(std::memory_order_acquire))
next_head = (head + GRANULE_QUEUE_SIZE / 2) % GRANULE_QUEUE_SIZE;
m_queue[head] = granule;
m_queue_head.store(next_head, std::memory_order_release);
m_queue_looping.store(false, std::memory_order_relaxed);
}
void Mixer::MixerFifo::Dequeue(Granule* granule)
{
// import numpy as np
// import scipy.signal as signal
// window = np.cumsum(signal.windows.dpss(32, 10))[::-1]
// window /= window.max()
// elements = ", ".join([f"{x:.10f}f" for x in window])
// print(f'constexpr std::array<StereoPair, {len(window)}> FADE_WINDOW = {{ {elements} }};')
constexpr std::array<float, 32> FADE_WINDOW = {
1.0000000000f, 0.9999999932f, 0.9999998472f, 0.9999982765f, 0.9999870876f, 0.9999278274f,
0.9996794215f, 0.9988227502f, 0.9963278433f, 0.9900772448f, 0.9764215513f, 0.9501402658f,
0.9052392639f, 0.8367449916f, 0.7430540364f, 0.6277889467f, 0.5000000000f, 0.3722110533f,
0.2569459636f, 0.1632550084f, 0.0947607361f, 0.0498597342f, 0.0235784487f, 0.0099227552f,
0.0036721567f, 0.0011772498f, 0.0003205785f, 0.0000721726f, 0.0000129124f, 0.0000017235f,
0.0000001528f, 0.0000000068f};
const std::size_t tail = m_queue_tail.load(std::memory_order_relaxed);
std::size_t next_tail = (tail + 1) % GRANULE_QUEUE_SIZE;
if (next_tail == m_queue_head.load(std::memory_order_acquire))
{
// Only fill gaps when running to prevent stutter on pause.
const bool is_running = Core::GetState(Core::System::GetInstance()) == Core::State::Running;
if (m_mixer->m_audio_fill_gaps && is_running)
{
next_tail = (tail + GRANULE_QUEUE_SIZE / 2) % GRANULE_QUEUE_SIZE;
m_queue_looping.store(true, std::memory_order_relaxed);
}
else
{
*granule = Granule();
return;
}
}
if (m_queue_looping.load(std::memory_order_relaxed))
m_queue_fade_index = std::min(m_queue_fade_index + 1, FADE_WINDOW.size() - 1);
else
m_queue_fade_index = 0;
*granule = m_queue[tail];
*granule *= StereoPair(FADE_WINDOW[m_queue_fade_index]);
m_queue_tail.store(next_tail, std::memory_order_release);
}
// Implementation of Granule's constructor
constexpr Mixer::MixerFifo::Granule::Granule(const GranuleBuffer& input,
const std::size_t start_index)
{
// import numpy as np
// import scipy.signal as signal
// window = np.convolve(np.ones(128), signal.windows.dpss(128 + 1, 4))
// window /= (window[:len(window) // 2] + window[len(window) // 2:]).max()
// elements = ", ".join([f"{x:.10f}f" for x in window])
// print(f'constexpr std::array<StereoPair, GRANULE_BUFFER_SIZE> GRANULE_WINDOW = {{ {elements}
// }};')
constexpr std::array<float, GRANULE_BUFFER_SIZE> GRANULE_WINDOW = {
0.0000016272f, 0.0000050749f, 0.0000113187f, 0.0000216492f, 0.0000377350f, 0.0000616906f,
0.0000961509f, 0.0001443499f, 0.0002102045f, 0.0002984010f, 0.0004144844f, 0.0005649486f,
0.0007573262f, 0.0010002765f, 0.0013036694f, 0.0016786636f, 0.0021377783f, 0.0026949534f,
0.0033656000f, 0.0041666352f, 0.0051165029f, 0.0062351752f, 0.0075441359f, 0.0090663409f,
0.0108261579f, 0.0128492811f, 0.0151626215f, 0.0177941726f, 0.0207728499f, 0.0241283062f,
0.0278907219f, 0.0320905724f, 0.0367583739f, 0.0419244083f, 0.0476184323f, 0.0538693708f,
0.0607049996f, 0.0681516192f, 0.0762337261f, 0.0849736833f, 0.0943913952f, 0.1045039915f,
0.1153255250f, 0.1268666867f, 0.1391345431f, 0.1521323012f, 0.1658591025f, 0.1803098534f,
0.1954750915f, 0.2113408944f, 0.2278888303f, 0.2450959552f, 0.2629348550f, 0.2813737361f,
0.3003765625f, 0.3199032396f, 0.3399098438f, 0.3603488941f, 0.3811696664f, 0.4023185434f,
0.4237393998f, 0.4453740162f, 0.4671625177f, 0.4890438330f, 0.5109561670f, 0.5328374823f,
0.5546259838f, 0.5762606002f, 0.5976814566f, 0.6188303336f, 0.6396511059f, 0.6600901562f,
0.6800967604f, 0.6996234375f, 0.7186262639f, 0.7370651450f, 0.7549040448f, 0.7721111697f,
0.7886591056f, 0.8045249085f, 0.8196901466f, 0.8341408975f, 0.8478676988f, 0.8608654569f,
0.8731333133f, 0.8846744750f, 0.8954960085f, 0.9056086048f, 0.9150263167f, 0.9237662739f,
0.9318483808f, 0.9392950004f, 0.9461306292f, 0.9523815677f, 0.9580755917f, 0.9632416261f,
0.9679094276f, 0.9721092781f, 0.9758716938f, 0.9792271501f, 0.9822058274f, 0.9848373785f,
0.9871507189f, 0.9891738421f, 0.9909336591f, 0.9924558641f, 0.9937648248f, 0.9948834971f,
0.9958333648f, 0.9966344000f, 0.9973050466f, 0.9978622217f, 0.9983213364f, 0.9986963306f,
0.9989997235f, 0.9992426738f, 0.9994350514f, 0.9995855156f, 0.9997015990f, 0.9997897955f,
0.9998556501f, 0.9999038491f, 0.9999383094f, 0.9999622650f, 0.9999783508f, 0.9999886813f,
0.9999949251f, 0.9999983728f, 0.9999983728f, 0.9999949251f, 0.9999886813f, 0.9999783508f,
0.9999622650f, 0.9999383094f, 0.9999038491f, 0.9998556501f, 0.9997897955f, 0.9997015990f,
0.9995855156f, 0.9994350514f, 0.9992426738f, 0.9989997235f, 0.9986963306f, 0.9983213364f,
0.9978622217f, 0.9973050466f, 0.9966344000f, 0.9958333648f, 0.9948834971f, 0.9937648248f,
0.9924558641f, 0.9909336591f, 0.9891738421f, 0.9871507189f, 0.9848373785f, 0.9822058274f,
0.9792271501f, 0.9758716938f, 0.9721092781f, 0.9679094276f, 0.9632416261f, 0.9580755917f,
0.9523815677f, 0.9461306292f, 0.9392950004f, 0.9318483808f, 0.9237662739f, 0.9150263167f,
0.9056086048f, 0.8954960085f, 0.8846744750f, 0.8731333133f, 0.8608654569f, 0.8478676988f,
0.8341408975f, 0.8196901466f, 0.8045249085f, 0.7886591056f, 0.7721111697f, 0.7549040448f,
0.7370651450f, 0.7186262639f, 0.6996234375f, 0.6800967604f, 0.6600901562f, 0.6396511059f,
0.6188303336f, 0.5976814566f, 0.5762606002f, 0.5546259838f, 0.5328374823f, 0.5109561670f,
0.4890438330f, 0.4671625177f, 0.4453740162f, 0.4237393998f, 0.4023185434f, 0.3811696664f,
0.3603488941f, 0.3399098438f, 0.3199032396f, 0.3003765625f, 0.2813737361f, 0.2629348550f,
0.2450959552f, 0.2278888303f, 0.2113408944f, 0.1954750915f, 0.1803098534f, 0.1658591025f,
0.1521323012f, 0.1391345431f, 0.1268666867f, 0.1153255250f, 0.1045039915f, 0.0943913952f,
0.0849736833f, 0.0762337261f, 0.0681516192f, 0.0607049996f, 0.0538693708f, 0.0476184323f,
0.0419244083f, 0.0367583739f, 0.0320905724f, 0.0278907219f, 0.0241283062f, 0.0207728499f,
0.0177941726f, 0.0151626215f, 0.0128492811f, 0.0108261579f, 0.0090663409f, 0.0075441359f,
0.0062351752f, 0.0051165029f, 0.0041666352f, 0.0033656000f, 0.0026949534f, 0.0021377783f,
0.0016786636f, 0.0013036694f, 0.0010002765f, 0.0007573262f, 0.0005649486f, 0.0004144844f,
0.0002984010f, 0.0002102045f, 0.0001443499f, 0.0000961509f, 0.0000616906f, 0.0000377350f,
0.0000216492f, 0.0000113187f, 0.0000050749f, 0.0000016272f};
const auto input_middle = input.end() - start_index;
std::ranges::rotate_copy(input, input_middle, m_buffer.begin());
for (std::size_t i = 0; i < m_buffer.size(); ++i)
m_buffer[i] *= StereoPair(GRANULE_WINDOW[i]);
}
Mixer::MixerFifo::StereoPair Mixer::MixerFifo::Granule::InterpStereoPair(const Granule& prev,
const Granule& next,
const u32 frac)
{
const std::size_t prev_index = frac >> Mixer::MixerFifo::GRANULE_BUFFER_FRAC_BITS;
const std::size_t next_index = prev_index - (GRANULE_BUFFER_SIZE / 2);
const u32 frac_t = frac & ((1 << GRANULE_BUFFER_FRAC_BITS) - 1);
const float t1 = frac_t / static_cast<float>(1 << GRANULE_BUFFER_FRAC_BITS);
const float t2 = t1 * t1;
const float t3 = t2 * t1;
// The Granules are pre-windowed, so we can just add them together
StereoPair s0 = prev.m_buffer[(prev_index - 2) & GRANULE_BUFFER_MASK] +
next.m_buffer[(next_index - 2) & GRANULE_BUFFER_MASK];
StereoPair s1 = prev.m_buffer[(prev_index - 1) & GRANULE_BUFFER_MASK] +
next.m_buffer[(next_index - 1) & GRANULE_BUFFER_MASK];
StereoPair s2 = prev.m_buffer[(prev_index + 0) & GRANULE_BUFFER_MASK] +
next.m_buffer[(next_index + 0) & GRANULE_BUFFER_MASK];
StereoPair s3 = prev.m_buffer[(prev_index + 1) & GRANULE_BUFFER_MASK] +
next.m_buffer[(next_index + 1) & GRANULE_BUFFER_MASK];
StereoPair s4 = prev.m_buffer[(prev_index + 2) & GRANULE_BUFFER_MASK] +
next.m_buffer[(next_index + 2) & GRANULE_BUFFER_MASK];
StereoPair s5 = prev.m_buffer[(prev_index + 3) & GRANULE_BUFFER_MASK] +
next.m_buffer[(next_index + 3) & GRANULE_BUFFER_MASK];
s0 *= StereoPair{(+0.0f + 1.0f * t1 - 2.0f * t2 + 1.0f * t3) / 12.0f};
s1 *= StereoPair{(+0.0f - 8.0f * t1 + 15.0f * t2 - 7.0f * t3) / 12.0f};
s2 *= StereoPair{(+3.0f + 0.0f * t1 - 7.0f * t2 + 4.0f * t3) / 3.0f};
s3 *= StereoPair{(+0.0f + 2.0f * t1 + 5.0f * t2 - 4.0f * t3) / 3.0f};
s4 *= StereoPair{(+0.0f - 1.0f * t1 - 6.0f * t2 + 7.0f * t3) / 12.0f};
s5 *= StereoPair{(+0.0f + 0.0f * t1 + 1.0f * t2 - 1.0f * t3) / 12.0f};
return s0 + s1 + s2 + s3 + s4 + s5;
}

147
Source/Core/AudioCommon/Mixer.h
View File

@ -3,10 +3,12 @@
#pragma once
#include <algorithm>
#include <array>
#include <atomic>
#include <bit>
#include <cmath>
#include "AudioCommon/AudioStretcher.h"
#include "AudioCommon/SurroundDecoder.h"
#include "AudioCommon/WaveFile.h"
#include "Common/CommonTypes.h"
@ -17,32 +19,32 @@ class PointerWrap;
class Mixer final
{
public:
explicit Mixer(unsigned int BackendSampleRate);
explicit Mixer(u32 BackendSampleRate);
~Mixer();
void DoState(PointerWrap& p);
// Called from audio threads
unsigned int Mix(short* samples, unsigned int numSamples);
unsigned int MixSurround(float* samples, unsigned int num_samples);
std::size_t Mix(s16* samples, std::size_t numSamples);
std::size_t MixSurround(float* samples, std::size_t num_samples);
// Called from main thread
void PushSamples(const short* samples, unsigned int num_samples);
void PushStreamingSamples(const short* samples, unsigned int num_samples);
void PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
unsigned int sample_rate_divisor);
void PushSkylanderPortalSamples(const u8* samples, unsigned int num_samples);
void PushGBASamples(int device_number, const short* samples, unsigned int num_samples);
void PushSamples(const s16* samples, std::size_t num_samples);
void PushStreamingSamples(const s16* samples, std::size_t num_samples);
void PushWiimoteSpeakerSamples(const s16* samples, std::size_t num_samples,
u32 sample_rate_divisor);
void PushSkylanderPortalSamples(const u8* samples, std::size_t num_samples);
void PushGBASamples(std::size_t device_number, const s16* samples, std::size_t num_samples);
unsigned int GetSampleRate() const { return m_sampleRate; }
u32 GetSampleRate() const { return m_output_sample_rate; }
void SetDMAInputSampleRateDivisor(unsigned int rate_divisor);
void SetStreamInputSampleRateDivisor(unsigned int rate_divisor);
void SetGBAInputSampleRateDivisors(int device_number, unsigned int rate_divisor);
void SetDMAInputSampleRateDivisor(u32 rate_divisor);
void SetStreamInputSampleRateDivisor(u32 rate_divisor);
void SetGBAInputSampleRateDivisors(std::size_t device_number, u32 rate_divisor);
void SetStreamingVolume(unsigned int lvolume, unsigned int rvolume);
void SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume);
void SetGBAVolume(int device_number, unsigned int lvolume, unsigned int rvolume);
void SetStreamingVolume(u32 lvolume, u32 rvolume);
void SetWiimoteSpeakerVolume(u32 lvolume, u32 rvolume);
void SetGBAVolume(std::size_t device_number, u32 lvolume, u32 rvolume);
void StartLogDTKAudio(const std::string& filename);
void StopLogDTKAudio();
@ -54,44 +56,105 @@ public:
static constexpr u64 FIXED_SAMPLE_RATE_DIVIDEND = 54000000 * 2;
private:
static constexpr u32 MAX_SAMPLES = 1024 * 4; // 128 ms
static constexpr u32 INDEX_MASK = MAX_SAMPLES * 2 - 1;
static constexpr int MAX_FREQ_SHIFT = 200; // Per 32000 Hz
static constexpr float CONTROL_FACTOR = 0.2f;
static constexpr u32 CONTROL_AVG = 32; // In freq_shift per FIFO size offset
const unsigned int SURROUND_CHANNELS = 6;
const std::size_t SURROUND_CHANNELS = 6;
class MixerFifo final
{
static constexpr std::size_t GRANULE_QUEUE_SIZE = 20;
template <typename T>
static s16 ToShort(const T x)
{
return static_cast<s16>(std::clamp<T>(x, static_cast<T>(std::numeric_limits<s16>::min()),
static_cast<T>(std::numeric_limits<s16>::max())));
}
struct StereoPair final
{
float l = 0.f;
float r = 0.f;
constexpr StereoPair() = default;
constexpr explicit StereoPair(float mono) : l(mono), r(mono) {}
constexpr StereoPair(float left, float right) : l(left), r(right) {}
constexpr StereoPair(s16 left, s16 right) : l(left), r(right) {}
StereoPair operator+(const StereoPair& other) const
{
return StereoPair(l + other.l, r + other.r);
}
StereoPair& operator*=(const StereoPair& other)
{
l *= other.l;
r *= other.r;
return *this;
}
};
static constexpr std::size_t GRANULE_BUFFER_SIZE = 256;
static constexpr std::size_t GRANULE_BUFFER_MASK = GRANULE_BUFFER_SIZE - 1;
static constexpr std::size_t GRANULE_BUFFER_BITS = std::countr_one(GRANULE_BUFFER_MASK);
static constexpr std::size_t GRANULE_BUFFER_FRAC_BITS = 32 - GRANULE_BUFFER_BITS;
using GranuleBuffer = std::array<StereoPair, GRANULE_BUFFER_SIZE>;
class Granule final
{
public:
constexpr Granule() = default;
constexpr Granule(const GranuleBuffer& input, std::size_t start_index);
static StereoPair InterpStereoPair(const Granule& front, const Granule& back, u32 frac);
Granule& operator*=(const StereoPair& x)
{
for (auto& sample : m_buffer)
sample *= x;
return *this;
}
private:
GranuleBuffer m_buffer{};
};
public:
MixerFifo(Mixer* mixer, unsigned sample_rate_divisor, bool little_endian)
MixerFifo(Mixer* mixer, u32 sample_rate_divisor, bool little_endian)
: m_mixer(mixer), m_input_sample_rate_divisor(sample_rate_divisor),
m_little_endian(little_endian)
{
}
void DoState(PointerWrap& p);
void PushSamples(const short* samples, unsigned int num_samples);
unsigned int Mix(short* samples, unsigned int numSamples, bool consider_framelimit,
float emulationspeed, int timing_variance);
void SetInputSampleRateDivisor(unsigned int rate_divisor);
unsigned int GetInputSampleRateDivisor() const;
void SetVolume(unsigned int lvolume, unsigned int rvolume);
void PushSamples(const s16* samples, std::size_t num_samples);
void Mix(s16* samples, std::size_t num_samples);
void SetInputSampleRateDivisor(u32 rate_divisor);
u32 GetInputSampleRateDivisor() const;
void SetVolume(u32 lvolume, u32 rvolume);
std::pair<s32, s32> GetVolume() const;
unsigned int AvailableSamples() const;
private:
Mixer* m_mixer;
unsigned m_input_sample_rate_divisor;
u32 m_input_sample_rate_divisor;
bool m_little_endian;
std::array<short, MAX_SAMPLES * 2> m_buffer{};
std::atomic<u32> m_indexW{0};
std::atomic<u32> m_indexR{0};
std::size_t m_buffer_index = 0;
GranuleBuffer m_buffer{};
u32 m_current_index = 0;
Granule m_front, m_back;
std::array<Granule, GRANULE_QUEUE_SIZE> m_queue;
std::atomic<std::size_t> m_queue_head{0};
std::atomic<std::size_t> m_queue_tail{0};
std::atomic<bool> m_queue_looping{false};
std::size_t m_queue_fade_index = 0;
void Enqueue(const Granule& granule);
void Dequeue(Granule* granule);
// Volume ranges from 0-256
std::atomic<s32> m_LVolume{256};
std::atomic<s32> m_RVolume{256};
float m_numLeftI = 0.0f;
u32 m_frac = 0;
StereoPair m_quantization_error;
};
void RefreshConfig();
@ -104,12 +167,9 @@ private:
MixerFifo{this, FIXED_SAMPLE_RATE_DIVIDEND / 48000, true},
MixerFifo{this, FIXED_SAMPLE_RATE_DIVIDEND / 48000, true},
MixerFifo{this, FIXED_SAMPLE_RATE_DIVIDEND / 48000, true}};
unsigned int m_sampleRate;
u32 m_output_sample_rate;
bool m_is_stretching = false;
AudioCommon::AudioStretcher m_stretcher;
AudioCommon::SurroundDecoder m_surround_decoder;
std::array<short, MAX_SAMPLES * 2> m_scratch_buffer{};
WaveFileWriter m_wave_writer_dtk;
WaveFileWriter m_wave_writer_dsp;
@ -118,8 +178,7 @@ private:
bool m_log_dsp_audio = false;
float m_config_emulation_speed;
int m_config_timing_variance;
bool m_config_audio_stretch;
bool m_audio_fill_gaps = true;
Config::ConfigChangedCallbackID m_config_changed_callback_id;
};

4
Source/Core/AudioCommon/OpenALStream.cpp
View File

@ -293,7 +293,7 @@ void OpenALStream::SoundLoop()
if (use_surround)
{
std::array<float, OAL_MAX_FRAMES * SURROUND_CHANNELS> dpl2;
u32 rendered_frames = m_mixer->MixSurround(dpl2.data(), min_frames);
u32 rendered_frames = static_cast<u32>(m_mixer->MixSurround(dpl2.data(), min_frames));
if (rendered_frames < min_frames)
continue;
@ -351,7 +351,7 @@ void OpenALStream::SoundLoop()
}
else
{
u32 rendered_frames = m_mixer->Mix(m_realtime_buffer.data(), min_frames);
u32 rendered_frames = static_cast<u32>(m_mixer->Mix(m_realtime_buffer.data(), min_frames));
if (!rendered_frames)
continue;

3
Source/Core/Core/Config/MainSettings.cpp
View File

@ -56,8 +56,7 @@ const Info<bool> MAIN_DPL2_DECODER{{System::Main, "Core", "DPL2Decoder"}, false}
const Info<AudioCommon::DPL2Quality> MAIN_DPL2_QUALITY{{System::Main, "Core", "DPL2Quality"},
AudioCommon::GetDefaultDPL2Quality()};
const Info<int> MAIN_AUDIO_LATENCY{{System::Main, "Core", "AudioLatency"}, 20};
const Info<bool> MAIN_AUDIO_STRETCH{{System::Main, "Core", "AudioStretch"}, false};
const Info<int> MAIN_AUDIO_STRETCH_LATENCY{{System::Main, "Core", "AudioStretchMaxLatency"}, 80};
const Info<bool> MAIN_AUDIO_FILL_GAPS{{System::Main, "Core", "AudioFillGaps"}, true};
const Info<std::string> MAIN_MEMCARD_A_PATH{{System::Main, "Core", "MemcardAPath"}, ""};
const Info<std::string> MAIN_MEMCARD_B_PATH{{System::Main, "Core", "MemcardBPath"}, ""};
const Info<std::string>& GetInfoForMemcardPath(ExpansionInterface::Slot slot)

3
Source/Core/Core/Config/MainSettings.h
View File

@ -72,8 +72,7 @@ extern const Info<bool> MAIN_OVERRIDE_REGION_SETTINGS;
extern const Info<bool> MAIN_DPL2_DECODER;
extern const Info<AudioCommon::DPL2Quality> MAIN_DPL2_QUALITY;
extern const Info<int> MAIN_AUDIO_LATENCY;
extern const Info<bool> MAIN_AUDIO_STRETCH;
extern const Info<int> MAIN_AUDIO_STRETCH_LATENCY;
extern const Info<bool> MAIN_AUDIO_FILL_GAPS;
extern const Info<std::string> MAIN_MEMCARD_A_PATH;
extern const Info<std::string> MAIN_MEMCARD_B_PATH;
const Info<std::string>& GetInfoForMemcardPath(ExpansionInterface::Slot slot);

2
Source/Core/DolphinLib.props
View File

@ -2,7 +2,6 @@
<Project>
<ItemGroup>
<ClInclude Include="AudioCommon\AudioCommon.h" />
<ClInclude Include="AudioCommon\AudioStretcher.h" />
<ClInclude Include="AudioCommon\CubebStream.h" />
<ClInclude Include="AudioCommon\CubebUtils.h" />
<ClInclude Include="AudioCommon\Enums.h" />
@ -769,7 +768,6 @@
</ItemGroup>
<ItemGroup>
<ClCompile Include="AudioCommon\AudioCommon.cpp" />
<ClCompile Include="AudioCommon\AudioStretcher.cpp" />
<ClCompile Include="AudioCommon\CubebStream.cpp" />
<ClCompile Include="AudioCommon\CubebUtils.cpp" />
<ClCompile Include="AudioCommon\Mixer.cpp" />

1
Source/Core/DolphinLib.vcxproj
View File

@ -56,7 +56,6 @@
<Import Project="$(ExternalsDir)rcheevos\exports.props" />
<Import Project="$(ExternalsDir)SDL\exports.props" />
<Import Project="$(ExternalsDir)SFML\exports.props" />
<Import Project="$(ExternalsDir)soundtouch\exports.props" />
<Import Project="$(ExternalsDir)spirv_cross\exports.props" />
<Import Project="$(ExternalsDir)tinygltf\exports.props" />
<Import Project="$(ExternalsDir)xxhash\exports.props" />

1
Source/Core/DolphinQt/DolphinQt.vcxproj
View File

@ -482,7 +482,6 @@
<Import Project="$(ExternalsDir)picojson\exports.props" />
<Import Project="$(ExternalsDir)rcheevos\exports.props" />
<Import Project="$(ExternalsDir)SFML\exports.props" />
<Import Project="$(ExternalsDir)soundtouch\exports.props" />
<Import Project="$(ExternalsDir)xxhash\exports.props" />
<Import Project="$(ExternalsDir)zstd\exports.props" />
<Import Project="$(VCTargetsPath)\Microsoft.Cpp.targets" />

59
Source/Core/DolphinQt/Settings/AudioPane.cpp
View File

@ -25,6 +25,7 @@
#include "Core/Core.h"
#include "Core/System.h"
#include "DolphinQt/Config/ConfigControls/ConfigBool.h"
#include "DolphinQt/Config/SettingsWindow.h"
#include "DolphinQt/Settings.h"
@ -137,43 +138,29 @@ void AudioPane::CreateWidgets()
backend_layout->addRow(dolby_quality_layout);
backend_layout->addRow(m_dolby_quality_latency_label);
auto* stretching_box = new QGroupBox(tr("Audio Stretching Settings"));
auto* stretching_layout = new QGridLayout;
m_stretching_enable = new QCheckBox(tr("Enable Audio Stretching"));
m_stretching_buffer_slider = new QSlider(Qt::Horizontal);
m_stretching_buffer_indicator = new QLabel();
m_stretching_buffer_label = new QLabel(tr("Buffer Size:"));
stretching_box->setLayout(stretching_layout);
m_stretching_buffer_slider->setMinimum(5);
m_stretching_buffer_slider->setMaximum(300);
m_stretching_enable->setToolTip(tr("Enables stretching of the audio to match emulation speed."));
m_stretching_buffer_slider->setToolTip(tr("Size of stretch buffer in milliseconds. "
"Values too low may cause audio crackling."));
stretching_layout->addWidget(m_stretching_enable, 0, 0, 1, -1);
stretching_layout->addWidget(m_stretching_buffer_label, 1, 0);
stretching_layout->addWidget(m_stretching_buffer_slider, 1, 1);
stretching_layout->addWidget(m_stretching_buffer_indicator, 1, 2);
dsp_box->setSizePolicy(QSizePolicy::Expanding, QSizePolicy::Fixed);
auto* misc_box = new QGroupBox(tr("Miscellaneous Settings"));
auto* misc_layout = new QGridLayout;
misc_box->setLayout(misc_layout);
m_speed_up_mute_enable = new QCheckBox(tr("Mute When Disabling Speed Limit"));
m_speed_up_mute_enable->setToolTip(
m_audio_fill_gaps = new ConfigBool(tr("Fill Audio Gaps"), Config::MAIN_AUDIO_FILL_GAPS);
m_audio_fill_gaps->SetDescription(
tr("Repeat existing audio during lag spikes to prevent stuttering."
"<br><br><dolphin_emphasis>If unsure, leave this checked.</dolphin_emphasis>"));
m_speed_up_mute_enable = new ConfigBool(tr("Mute When Disabling Speed Limit"),
Config::MAIN_AUDIO_MUTE_ON_DISABLED_SPEED_LIMIT);
m_speed_up_mute_enable->SetDescription(
tr("Mutes the audio when overriding the emulation speed limit (default hotkey: Tab)."));
misc_layout->addWidget(m_speed_up_mute_enable, 0, 0, 1, 1);
misc_layout->addWidget(m_audio_fill_gaps, 0, 0, 1, 1);
misc_layout->addWidget(m_speed_up_mute_enable, 1, 0, 1, 1);
auto* const main_vbox_layout = new QVBoxLayout;
main_vbox_layout->addWidget(dsp_box);
main_vbox_layout->addWidget(backend_box);
main_vbox_layout->addWidget(stretching_box);
main_vbox_layout->addWidget(misc_box);
m_main_layout = new QHBoxLayout;
@ -192,14 +179,11 @@ void AudioPane::ConnectWidgets()
{
connect(m_latency_spin, &QSpinBox::valueChanged, this, &AudioPane::SaveSettings);
}
connect(m_stretching_buffer_slider, &QSlider::valueChanged, this, &AudioPane::SaveSettings);
connect(m_dolby_pro_logic, &QCheckBox::toggled, this, &AudioPane::SaveSettings);
connect(m_dolby_quality_slider, &QSlider::valueChanged, this, &AudioPane::SaveSettings);
connect(m_stretching_enable, &QCheckBox::toggled, this, &AudioPane::SaveSettings);
connect(m_dsp_hle, &QRadioButton::toggled, this, &AudioPane::SaveSettings);
connect(m_dsp_lle, &QRadioButton::toggled, this, &AudioPane::SaveSettings);
connect(m_dsp_interpreter, &QRadioButton::toggled, this, &AudioPane::SaveSettings);
connect(m_speed_up_mute_enable, &QCheckBox::toggled, this, &AudioPane::SaveSettings);
#ifdef _WIN32
connect(m_wasapi_device_combo, &QComboBox::currentIndexChanged, this, &AudioPane::SaveSettings);
@ -255,17 +239,6 @@ void AudioPane::LoadSettings()
if (m_latency_control_supported)
m_latency_spin->setValue(Config::Get(Config::MAIN_AUDIO_LATENCY));
// Stretch
m_stretching_enable->setChecked(Config::Get(Config::MAIN_AUDIO_STRETCH));
m_stretching_buffer_label->setEnabled(m_stretching_enable->isChecked());
m_stretching_buffer_slider->setValue(Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY));
m_stretching_buffer_slider->setEnabled(m_stretching_enable->isChecked());
m_stretching_buffer_indicator->setEnabled(m_stretching_enable->isChecked());
m_stretching_buffer_indicator->setText(tr("%1 ms").arg(m_stretching_buffer_slider->value()));
// Misc
m_speed_up_mute_enable->setChecked(Config::Get(Config::MAIN_AUDIO_MUTE_ON_DISABLED_SPEED_LIMIT));
#ifdef _WIN32
if (Config::Get(Config::MAIN_WASAPI_DEVICE) == "default")
{
@ -326,16 +299,8 @@ void AudioPane::SaveSettings()
if (m_latency_control_supported)
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_LATENCY, m_latency_spin->value());
// Stretch
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_STRETCH, m_stretching_enable->isChecked());
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_STRETCH_LATENCY, m_stretching_buffer_slider->value());
m_stretching_buffer_label->setEnabled(m_stretching_enable->isChecked());
m_stretching_buffer_slider->setEnabled(m_stretching_enable->isChecked());
m_stretching_buffer_indicator->setEnabled(m_stretching_enable->isChecked());
m_stretching_buffer_indicator->setText(
tr("%1 ms").arg(Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY)));
// Misc
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_FILL_GAPS, m_audio_fill_gaps->isChecked());
Config::SetBaseOrCurrent(Config::MAIN_AUDIO_MUTE_ON_DISABLED_SPEED_LIMIT,
m_speed_up_mute_enable->isChecked());

10
Source/Core/DolphinQt/Settings/AudioPane.h
View File

@ -18,6 +18,7 @@ class QRadioButton;
class QSlider;
class QSpinBox;
class SettingsWindow;
class ConfigBool;
class AudioPane final : public QWidget
{
@ -71,12 +72,7 @@ private:
QComboBox* m_wasapi_device_combo;
#endif
// Audio Stretching
QCheckBox* m_stretching_enable;
QLabel* m_stretching_buffer_label;
QSlider* m_stretching_buffer_slider;
QLabel* m_stretching_buffer_indicator;
// Misc Settings
QCheckBox* m_speed_up_mute_enable;
ConfigBool* m_audio_fill_gaps;
ConfigBool* m_speed_up_mute_enable;
};

11
Source/dolphin-emu.sln
View File

@ -33,8 +33,6 @@ Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "xxhash", "..\Externals\xxha
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "zlib-ng", "..\Externals\zlib-ng\zlib-ng.vcxproj", "{F6EA7144-8D64-4EBB-A13E-76DFBD911EAE}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SoundTouch", "..\Externals\soundtouch\SoundTouch.vcxproj", "{EC082900-B4D8-42E9-9663-77F02F6936AE}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "mbedTLS", "..\Externals\mbedtls\mbedTLS.vcxproj", "{BDB6578B-0691-4E80-A46C-DF21639FD3B8}"
EndProject
Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "SCMRevGen", "Core\Common\SCMRevGen.vcxproj", "{41279555-F94F-4EBC-99DE-AF863C10C5C4}"
@ -209,14 +207,6 @@ Global
{F6EA7144-8D64-4EBB-A13E-76DFBD911EAE}.Release|ARM64.Build.0 = Release|ARM64
{F6EA7144-8D64-4EBB-A13E-76DFBD911EAE}.Release|x64.ActiveCfg = Release|x64
{F6EA7144-8D64-4EBB-A13E-76DFBD911EAE}.Release|x64.Build.0 = Release|x64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Debug|ARM64.ActiveCfg = Debug|ARM64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Debug|ARM64.Build.0 = Debug|ARM64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Debug|x64.ActiveCfg = Debug|x64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Debug|x64.Build.0 = Debug|x64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Release|ARM64.ActiveCfg = Release|ARM64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Release|ARM64.Build.0 = Release|ARM64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Release|x64.ActiveCfg = Release|x64
{EC082900-B4D8-42E9-9663-77F02F6936AE}.Release|x64.Build.0 = Release|x64
{BDB6578B-0691-4E80-A46C-DF21639FD3B8}.Debug|ARM64.ActiveCfg = Debug|ARM64
{BDB6578B-0691-4E80-A46C-DF21639FD3B8}.Debug|ARM64.Build.0 = Debug|ARM64
{BDB6578B-0691-4E80-A46C-DF21639FD3B8}.Debug|x64.ActiveCfg = Debug|x64
@ -460,7 +450,6 @@ Global
{31643FDB-1BB8-4965-9DE7-000FC88D35AE} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{677EA016-1182-440C-9345-DC88D1E98C0C} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{F6EA7144-8D64-4EBB-A13E-76DFBD911EAE} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{EC082900-B4D8-42E9-9663-77F02F6936AE} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{BDB6578B-0691-4E80-A46C-DF21639FD3B8} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{93D73454-2512-424E-9CDA-4BB357FE13DD} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}
{349EE8F9-7D25-4909-AAF5-FF3FADE72187} = {87ADDFF9-5768-4DA2-A33B-2477593D6677}