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Revert "Audio Core (#2)"

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This reverts commit a8d0c51c69.
This commit is contained in:
Dragios 2016-04-16 01:32:48 +08:00
parent 69effbcb6e
commit 28f64f98f7
65 changed files with 52 additions and 8680 deletions
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3
.gitmodules vendored
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@ -7,6 +7,3 @@
[submodule "nihstro"]
path = externals/nihstro
url = https://github.com/neobrain/nihstro.git
[submodule "rubberband"]
path = externals/rubberband/rubberband
url = https://github.com/breakfastquay/rubberband.git

7
CMakeLists.txt
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@ -148,7 +148,6 @@ if (ENABLE_SDL2)
download_bundled_external("sdl2/" ${SDL2_VER} SDL2_PREFIX)
endif()
set(SDL2_FOUND YES)
set(SDL2_INCLUDE_DIR "${SDL2_PREFIX}/include" CACHE PATH "Path to SDL2 headers")
set(SDL2_LIBRARY "${SDL2_PREFIX}/lib/x64/SDL2.lib" CACHE PATH "Path to SDL2 library")
set(SDL2_DLL_DIR "${SDL2_PREFIX}/lib/x64/" CACHE PATH "Path to SDL2.dll")
@ -241,15 +240,11 @@ if (MSVC)
add_subdirectory(externals/getopt)
endif()
# process subdirectories
if(ENABLE_QT)
include_directories(externals/qhexedit)
add_subdirectory(externals/qhexedit)
endif()
add_subdirectory(externals/soundtouch)
add_subdirectory(externals/rubberband)
# process subdirectories
add_subdirectory(src)
# Install freedesktop.org metadata files, following those specifications:

2
externals/boost vendored

@ -1 +1 @@
Subproject commit 2dcb9d979665b6aabb1635c617973e02914e60ec
Subproject commit d81b9269900ae183d0dc98403eea4c971590a807

79
externals/rubberband/CMakeLists.txt vendored
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@ -1,79 +0,0 @@
set(SRCS
rubberband/src/audiocurves/CompoundAudioCurve.cpp
rubberband/src/audiocurves/ConstantAudioCurve.cpp
rubberband/src/audiocurves/HighFrequencyAudioCurve.cpp
rubberband/src/audiocurves/PercussiveAudioCurve.cpp
rubberband/src/audiocurves/SilentAudioCurve.cpp
rubberband/src/audiocurves/SpectralDifferenceAudioCurve.cpp
rubberband/src/base/Profiler.cpp
rubberband/src/dsp/AudioCurveCalculator.cpp
rubberband/src/dsp/FFT.cpp
rubberband/src/dsp/Resampler.cpp
rubberband/src/kissfft/kiss_fft.c
rubberband/src/kissfft/kiss_fftr.c
rubberband/src/RubberBandStretcher.cpp
rubberband/src/speex/resample.c
rubberband/src/StretchCalculator.cpp
rubberband/src/StretcherChannelData.cpp
rubberband/src/StretcherImpl.cpp
rubberband/src/StretcherProcess.cpp
rubberband/src/system/Allocators.cpp
rubberband/src/system/sysutils.cpp
rubberband/src/system/Thread.cpp
rubberband/src/system/VectorOpsComplex.cpp
)
SET(HEADERS
rubberband/src/audiocurves/CompoundAudioCurve.h
rubberband/src/audiocurves/ConstantAudioCurve.h
rubberband/src/audiocurves/HighFrequencyAudioCurve.h
rubberband/src/audiocurves/PercussiveAudioCurve.h
rubberband/src/audiocurves/SilentAudioCurve.h
rubberband/src/audiocurves/SpectralDifferenceAudioCurve.h
rubberband/src/base/Profiler.h
rubberband/src/base/RingBuffer.h
rubberband/src/base/Scavenger.h
rubberband/src/dsp/AudioCurveCalculator.h
rubberband/src/dsp/FFT.h
rubberband/src/dsp/MovingMedian.h
rubberband/src/dsp/Resampler.h
rubberband/src/dsp/SampleFilter.h
rubberband/src/dsp/SincWindow.h
rubberband/src/dsp/Window.h
rubberband/src/float_cast/float_cast.h
rubberband/src/kissfft/kiss_fft.h
rubberband/src/kissfft/kiss_fftr.h
rubberband/src/kissfft/_kiss_fft_guts.h
rubberband/src/pommier/neon_mathfun.h
rubberband/src/pommier/sse_mathfun.h
rubberband/src/speex/speex_resampler.h
rubberband/src/StretchCalculator.h
rubberband/src/StretcherChannelData.h
rubberband/src/StretcherImpl.h
rubberband/src/system/Allocators.h
rubberband/src/system/sysutils.h
rubberband/src/system/Thread.h
rubberband/src/system/VectorOps.h
rubberband/src/system/VectorOpsComplex.h
rubberband/rubberband/rubberband-c.h
rubberband/rubberband/RubberBandStretcher.h
)
add_library(rubberband STATIC ${SRCS} ${HEADERS})
target_include_directories(rubberband PRIVATE rubberband/src)
target_include_directories(rubberband PRIVATE rubberband)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS USE_SPEEX)
if(APPLE)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS HAVE_VDSP)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS USE_PTHREADS)
target_link_libraries(rubberband "-framework Accelerate")
elseif(MSVC)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS USE_KISSFFT)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS __MSVC__)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS WIN32)
else()
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS USE_KISSFFT)
set_property(TARGET rubberband APPEND PROPERTY COMPILE_DEFINITIONS USE_PTHREADS)
endif()

1
externals/rubberband/rubberband vendored

@ -1 +0,0 @@
Subproject commit c93a18535ffea1ca7b18eb41c34064b77f8419e3

236
externals/soundtouch/AAFilter.cpp vendored
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@ -1,236 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-05 23:40:22 +0200 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.cpp 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 3.141592655357989
#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 == NULL) 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;
//#if SOUNDTOUCH_INTEGER_SAMPLES
// scale & round to nearest integer
temp += (temp >= 0) ? 0.5 : -0.5;
// ensure no overfloods
assert(temp >= -32768 && temp <= 32767);
//#endif
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();
}

100
externals/soundtouch/AAFilter.h vendored
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@ -1,100 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-07 21:41:23 +0200 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: AAFilter.h 187 2014-01-07 19:41:23Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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

371
externals/soundtouch/BPMDetect.cpp vendored
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@ -1,371 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-21 23:24:29 +0200 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include <assert.h>
#include <string.h>
#include <stdio.h>
#include "FIFOSampleBuffer.h"
#include "PeakFinder.h"
#include "BPMDetect.h"
using namespace soundtouch;
#define INPUT_BLOCK_SAMPLES 2048
#define DECIMATED_BLOCK_SAMPLES 256
/// decay constant for calculating RMS volume sliding average approximation
/// (time constant is about 10 sec)
const float avgdecay = 0.99986f;
/// Normalization coefficient for calculating RMS sliding average approximation.
const float avgnorm = (1 - avgdecay);
////////////////////////////////////////////////////////////////////////////////
// Enable following define to create bpm analysis file:
// #define _CREATE_BPM_DEBUG_FILE
#ifdef _CREATE_BPM_DEBUG_FILE
#define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt"
static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff)
{
FILE *fptr = fopen(DEBUGFILE_NAME, "wt");
int i;
if (fptr)
{
printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n");
for (i = minpos; i < maxpos; i ++)
{
fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]);
}
fclose(fptr);
}
}
#else
#define _SaveDebugData(a,b,c,d)
#endif
////////////////////////////////////////////////////////////////////////////////
BPMDetect::BPMDetect(int numChannels, int aSampleRate)
{
this->sampleRate = aSampleRate;
this->channels = numChannels;
decimateSum = 0;
decimateCount = 0;
envelopeAccu = 0;
// Initialize RMS volume accumulator to RMS level of 1500 (out of 32768) that's
// safe initial RMS signal level value for song data. This value is then adapted
// to the actual level during processing.
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// integer samples
RMSVolumeAccu = (1500 * 1500) / avgnorm;
#else
// float samples, scaled to range [-1..+1[
RMSVolumeAccu = (0.045f * 0.045f) / avgnorm;
#endif
// choose decimation factor so that result is approx. 1000 Hz
decimateBy = sampleRate / 1000;
assert(decimateBy > 0);
assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
// Calculate window length & starting item according to desired min & max bpms
windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
assert(windowLen > windowStart);
// allocate new working objects
xcorr = new float[windowLen];
memset(xcorr, 0, windowLen * sizeof(float));
// allocate processing buffer
buffer = new FIFOSampleBuffer();
// we do processing in mono mode
buffer->setChannels(1);
buffer->clear();
}
BPMDetect::~BPMDetect()
{
delete[] xcorr;
delete buffer;
}
/// convert to mono, low-pass filter & decimate to about 500 Hz.
/// return number of outputted samples.
///
/// Decimation is used to remove the unnecessary frequencies and thus to reduce
/// the amount of data needed to be processed as calculating autocorrelation
/// function is a very-very heavy operation.
///
/// Anti-alias filtering is done simply by averaging the samples. This is really a
/// poor-man's anti-alias filtering, but it's not so critical in this kind of application
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
/// narrow band)
int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
{
int count, outcount;
LONG_SAMPLETYPE out;
assert(channels > 0);
assert(decimateBy > 0);
outcount = 0;
for (count = 0; count < numsamples; count ++)
{
int j;
// convert to mono and accumulate
for (j = 0; j < channels; j ++)
{
decimateSum += src[j];
}
src += j;
decimateCount ++;
if (decimateCount >= decimateBy)
{
// Store every Nth sample only
out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels));
decimateSum = 0;
decimateCount = 0;
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// check ranges for sure (shouldn't actually be necessary)
if (out > 32767)
{
out = 32767;
}
else if (out < -32768)
{
out = -32768;
}
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[outcount] = (SAMPLETYPE)out;
outcount ++;
}
}
return outcount;
}
// Calculates autocorrelation function of the sample history buffer
void BPMDetect::updateXCorr(int process_samples)
{
int offs;
SAMPLETYPE *pBuffer;
assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
pBuffer = buffer->ptrBegin();
#pragma omp parallel for
for (offs = windowStart; offs < windowLen; offs ++)
{
LONG_SAMPLETYPE sum;
int i;
sum = 0;
for (i = 0; i < process_samples; i ++)
{
sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
}
// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients
// if it's desired that the system adapts automatically to
// various bpms, e.g. in processing continouos music stream.
// The 'xcorr_decay' should be a value that's smaller than but
// close to one, and should also depend on 'process_samples' value.
xcorr[offs] += (float)sum;
}
}
// Calculates envelope of the sample data
void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
{
const static double decay = 0.7f; // decay constant for smoothing the envelope
const static double norm = (1 - decay);
int i;
LONG_SAMPLETYPE out;
double val;
for (i = 0; i < numsamples; i ++)
{
// calc average RMS volume
RMSVolumeAccu *= avgdecay;
val = (float)fabs((float)samples[i]);
RMSVolumeAccu += val * val;
// cut amplitudes that are below cutoff ~2 times RMS volume
// (we're interested in peak values, not the silent moments)
if (val < 0.5 * sqrt(RMSVolumeAccu * avgnorm))
{
val = 0;
}
// smooth amplitude envelope
envelopeAccu *= decay;
envelopeAccu += val;
out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
// cut peaks (shouldn't be necessary though)
if (out > 32767) out = 32767;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
samples[i] = (SAMPLETYPE)out;
}
}
void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples)
{
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
// iterate so that max INPUT_BLOCK_SAMPLES processed per iteration
while (numSamples > 0)
{
int block;
int decSamples;
block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples;
// decimate. note that converts to mono at the same time
decSamples = decimate(decimated, samples, block);
samples += block * channels;
numSamples -= block;
// envelope new samples and add them to buffer
calcEnvelope(decimated, decSamples);
buffer->putSamples(decimated, decSamples);
}
// when the buffer has enought samples for processing...
if ((int)buffer->numSamples() > windowLen)
{
int processLength;
// how many samples are processed
processLength = (int)buffer->numSamples() - windowLen;
// ... calculate autocorrelations for oldest samples...
updateXCorr(processLength);
// ... and remove them from the buffer
buffer->receiveSamples(processLength);
}
}
void BPMDetect::removeBias()
{
int i;
float minval = 1e12f; // arbitrary large number
for (i = windowStart; i < windowLen; i ++)
{
if (xcorr[i] < minval)
{
minval = xcorr[i];
}
}
for (i = windowStart; i < windowLen; i ++)
{
xcorr[i] -= minval;
}
}
float BPMDetect::getBpm()
{
double peakPos;
double coeff;
PeakFinder peakFinder;
coeff = 60.0 * ((double)sampleRate / (double)decimateBy);
// save bpm debug analysis data if debug data enabled
_SaveDebugData(xcorr, windowStart, windowLen, coeff);
// remove bias from xcorr data
removeBias();
// find peak position
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
assert(decimateBy != 0);
if (peakPos < 1e-9) return 0.0; // detection failed.
// calculate BPM
return (float) (coeff / peakPos);
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-08-30 22:53:44 +0300 (Thu, 30 Aug 2012) $
// File revision : $Revision: 4 $
//
// $Id: BPMDetect.h 150 2012-08-30 19:53:44Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 "STTypes.h"
#include "FIFOSampleBuffer.h"
namespace soundtouch
{
/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit.
#define MIN_BPM 29
/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit.
#define MAX_BPM 200
/// Class for calculating BPM rate for audio data.
class BPMDetect
{
protected:
/// Auto-correlation accumulator bins.
float *xcorr;
/// Amplitude envelope sliding average approximation level accumulator
double envelopeAccu;
/// RMS volume sliding average approximation level accumulator
double RMSVolumeAccu;
/// 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;
/// FIFO-buffer for decimated processing samples.
soundtouch::FIFOSampleBuffer *buffer;
/// 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();
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();
};
}
#endif // _BPMDetect_H_

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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})

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-11-08 20:53:01 +0200 (Thu, 08 Nov 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.cpp 160 2012-11-08 18:53:01Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 = NULL;
bufferUnaligned = NULL;
samplesInBuffer = 0;
bufferPos = 0;
channels = (uint)numChannels;
ensureCapacity(32); // allocate initial capacity
}
// destructor
FIFOSampleBuffer::~FIFOSampleBuffer()
{
delete[] bufferUnaligned;
bufferUnaligned = NULL;
buffer = NULL;
}
// Sets number of channels, 1 = mono, 2 = stereo
void FIFOSampleBuffer::setChannels(int numChannels)
{
uint usedBytes;
assert(numChannels > 0);
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
// succesfully 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 enought 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 == NULL)
{
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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-05 23:40:22 +0200 (Sun, 05 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSampleBuffer.h 177 2014-01-05 21:40:22Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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();
/// 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();
/// 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 succesfully 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.
);
/// 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.
);
/// 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.
);
/// Returns number of samples currently available.
virtual uint numSamples() const;
/// 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;
/// Clears all the samples.
virtual void clear();
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
uint adjustAmountOfSamples(uint numSamples);
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// '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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2012-06-13 22:29:53 +0300 (Wed, 13 Jun 2012) $
// File revision : $Revision: 4 $
//
// $Id: FIFOSamplePipe.h 143 2012-06-13 19:29:53Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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
{
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 == NULL);
assert(pOutput != NULL);
output = pOutput;
}
/// Constructor. Doesn't define output pipe; it has to be set be
/// 'setOutPipe' function.
FIFOProcessor()
{
output = NULL;
}
/// Constructor. Configures output pipe.
FIFOProcessor(FIFOSamplePipe *pOutput ///< Output pipe.
)
{
output = pOutput;
}
/// Destructor.
virtual ~FIFOProcessor()
{
}
/// 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()
{
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.
)
{
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.
)
{
return output->receiveSamples(maxSamples);
}
/// Returns number of samples currently available.
virtual uint numSamples() const
{
return output->numSamples();
}
/// Returns nonzero if there aren't any samples available for outputting.
virtual int isEmpty() const
{
return output->isEmpty();
}
/// allow trimming (downwards) amount of samples in pipeline.
/// Returns adjusted amount of samples
virtual uint adjustAmountOfSamples(uint numSamples)
{
return output->adjustAmountOfSamples(numSamples);
}
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-21 23:24:29 +0200 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.cpp 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 = NULL;
}
FIRFilter::~FIRFilter()
{
delete[] filterCoeffs;
}
// Usual C-version of the filter routine for stereo sound
uint FIRFilter::evaluateFilterStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
end = 2 * (numSamples - length);
#pragma omp parallel for
for (j = 0; j < end; j += 2)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE suml, sumr;
uint i;
suml = sumr = 0;
ptr = src + j;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
suml += ptr[2 * i + 0] * filterCoeffs[i + 0] +
ptr[2 * i + 2] * filterCoeffs[i + 1] +
ptr[2 * i + 4] * filterCoeffs[i + 2] +
ptr[2 * i + 6] * filterCoeffs[i + 3];
sumr += ptr[2 * i + 1] * filterCoeffs[i + 0] +
ptr[2 * i + 3] * filterCoeffs[i + 1] +
ptr[2 * i + 5] * filterCoeffs[i + 2] +
ptr[2 * i + 7] * filterCoeffs[i + 3];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
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;
#else
suml *= dScaler;
sumr *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j] = (SAMPLETYPE)suml;
dest[j + 1] = (SAMPLETYPE)sumr;
}
return numSamples - length;
}
// Usual C-version of the filter routine for mono sound
uint FIRFilter::evaluateFilterMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples) const
{
int j, end;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
end = numSamples - length;
#pragma omp parallel for
for (j = 0; j < end; j ++)
{
const SAMPLETYPE *pSrc = src + j;
LONG_SAMPLETYPE sum;
uint i;
sum = 0;
for (i = 0; i < length; i += 4)
{
// loop is unrolled by factor of 4 here for efficiency
sum += pSrc[i + 0] * filterCoeffs[i + 0] +
pSrc[i + 1] * filterCoeffs[i + 1] +
pSrc[i + 2] * filterCoeffs[i + 2] +
pSrc[i + 3] * filterCoeffs[i + 3];
}
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
sum >>= resultDivFactor;
// saturate to 16 bit integer limits
sum = (sum < -32768) ? -32768 : (sum > 32767) ? 32767 : sum;
#else
sum *= dScaler;
#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;
#ifdef SOUNDTOUCH_FLOAT_SAMPLES
// when using floating point samples, use a scaler instead of a divider
// because division is much slower operation than multiplying.
double dScaler = 1.0 / (double)resultDivider;
#endif
assert(length != 0);
assert(src != NULL);
assert(dest != NULL);
assert(filterCoeffs != NULL);
assert(numChannels < 16);
end = numChannels * (numSamples - length);
#pragma omp parallel for
for (j = 0; j < end; j += numChannels)
{
const SAMPLETYPE *ptr;
LONG_SAMPLETYPE sums[16];
uint c, i;
for (c = 0; c < numChannels; c ++)
{
sums[c] = 0;
}
ptr = src + j;
for (i = 0; i < length; 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;
#else
sums[c] *= dScaler;
#endif // SOUNDTOUCH_INTEGER_SAMPLES
dest[j+c] = (SAMPLETYPE)sums[c];
}
}
return numSamples - length;
}
// 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");
lengthDiv8 = newLength / 8;
length = lengthDiv8 * 8;
assert(length == newLength);
resultDivFactor = uResultDivFactor;
resultDivider = (SAMPLETYPE)::pow(2.0, (int)resultDivFactor);
delete[] filterCoeffs;
filterCoeffs = new SAMPLETYPE[length];
memcpy(filterCoeffs, coeffs, length * sizeof(SAMPLETYPE));
}
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 s)
{
// 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;
}
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-02-21 23:24:29 +0200 (Sat, 21 Feb 2015) $
// File revision : $Revision: 4 $
//
// $Id: FIRFilter.h 202 2015-02-21 21:24:29Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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;
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;
public:
FIRFilterMMX();
~FIRFilterMMX();
virtual void setCoefficients(const short *coeffs, uint newLength, uint uResultDivFactor);
};
#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;
public:
FIRFilterSSE();
~FIRFilterSSE();
virtual void setCoefficients(const float *coeffs, uint newLength, uint uResultDivFactor);
};
#endif // SOUNDTOUCH_ALLOW_SSE
}
#endif // FIRFilter_H

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////////////////////////////////////////////////////////////////////////////////
///
/// Cubic interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateCubic.cpp 179 2014-01-06 18:41:42Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// Cubic interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateCubic.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateCubic_H_
#define _InterpolateCubic_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
class InterpolateCubic : public TransposerBase
{
protected:
virtual void resetRegisters();
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);
double fract;
public:
InterpolateCubic();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// Linear interpolation algorithm.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateLinear.cpp 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <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 = (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;
}

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@ -1,92 +0,0 @@
////////////////////////////////////////////////////////////////////////////////
///
/// Linear interpolation routine.
///
/// Author : Copyright (c) Olli Parviainen
/// Author e-mail : oparviai 'at' iki.fi
/// SoundTouch WWW: http://www.surina.net/soundtouch
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateLinear.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateLinear_H_
#define _InterpolateLinear_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
/// Linear transposer class that uses integer arithmetics
class InterpolateLinearInteger : public TransposerBase
{
protected:
int iFract;
int iRate;
virtual void resetRegisters();
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:
InterpolateLinearInteger();
/// Sets new target rate. Normal rate = 1.0, smaller values represent slower
/// rate, larger faster rates.
virtual void setRate(double newRate);
};
/// Linear transposer class that uses floating point arithmetics
class InterpolateLinearFloat : public TransposerBase
{
protected:
double fract;
virtual void resetRegisters();
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();
};
}
#endif

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@ -1,185 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateShannon.cpp 195 2014-04-06 15:57:21Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#include <math.h>
#include "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 *pdest,
const SAMPLETYPE *psrc,
int &srcSamples)
{
// not implemented
assert(false);
return 0;
}

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@ -1,72 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// $Id: InterpolateShannon.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// SoundTouch audio processing library
// Copyright (c) Olli Parviainen
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2.1 of the License, or (at your option) any later version.
//
// This library is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
////////////////////////////////////////////////////////////////////////////////
#ifndef _InterpolateShannon_H_
#define _InterpolateShannon_H_
#include "RateTransposer.h"
#include "STTypes.h"
namespace soundtouch
{
class InterpolateShannon : public TransposerBase
{
protected:
void resetRegisters();
int transposeMono(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
int transposeStereo(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
int transposeMulti(SAMPLETYPE *dest,
const SAMPLETYPE *src,
int &srcSamples);
double fract;
public:
InterpolateShannon();
};
}
#endif

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-05-18 18:22:02 +0300 (Mon, 18 May 2015) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.cpp 213 2015-05-18 15:22:02Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 < 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 = 3; i < 10; i ++)
{
double peaktmp, harmonic;
int i1,i2;
harmonic = (double)i * 0.5;
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;
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2011-12-30 22:33:46 +0200 (Fri, 30 Dec 2011) $
// File revision : $Revision: 4 $
//
// $Id: PeakFinder.h 132 2011-12-30 20:33:46Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 beloging to the peak.
int lastPos ///< Index of last vector item beloging 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_

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-07-26 17:45:48 +0300 (Sun, 26 Jul 2015) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.cpp 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 = true;
// Instantiates the anti-alias filter
pAAFilter = new AAFilter(64);
pTransposer = TransposerBase::newInstance();
}
RateTransposer::~RateTransposer()
{
delete pAAFilter;
delete pTransposer;
}
/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
void RateTransposer::enableAAFilter(bool newMode)
{
bUseAAFilter = newMode;
}
/// 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)
{
uint count;
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)
{
count = 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)
{
assert(nChannels > 0);
if (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();
}
// 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();
}
//////////////////////////////////////////////////////////////////////////////
//
// 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 arithmetics 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 NULL;
}
#endif
}

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@ -1,179 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-07-26 17:45:48 +0300 (Sun, 26 Jul 2015) $
// File revision : $Revision: 4 $
//
// $Id: RateTransposer.h 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 void resetRegisters() = 0;
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);
// 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();
/// Operator 'new' is overloaded so that it automatically creates a suitable instance
/// depending on if we're to use integer or floating point arithmetics.
// static void *operator new(size_t s);
/// Use this function instead of "new" operator to create a new instance of this class.
/// This function automatically chooses a correct implementation, depending on if
/// integer ot floating point arithmetics are to be used.
// static RateTransposer *newInstance();
/// Returns the output buffer object
FIFOSamplePipe *getOutput() { return &outputBuffer; };
/// Returns the store buffer object
// FIFOSamplePipe *getStore() { return &storeBuffer; };
/// 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);
/// Clears all the samples in the object
void clear();
/// Returns nonzero if there aren't any samples available for outputting.
int isEmpty() const;
};
}
#endif

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@ -1,185 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-05-18 18:25:07 +0300 (Mon, 18 May 2015) $
// File revision : $Revision: 3 $
//
// $Id: STTypes.h 215 2015-05-18 15:25:07Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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
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 )
#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
{
/// 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__))
// 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 take minor shortcuts
// for improved performance. Undefine to require faithfully similar SIMD
// calculations as in normal C implementation.
#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
#define SOUNDTOUCH_ALLOW_MMX 1
#endif
#else
// floating point samples
typedef float SAMPLETYPE;
// data type for sample accumulation: Use double to utilize full precision.
typedef double LONG_SAMPLETYPE;
#ifdef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS
// Allow SSE optimizations
#define SOUNDTOUCH_ALLOW_SSE 1
#endif
#endif // SOUNDTOUCH_INTEGER_SAMPLES
};
// 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

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@ -1,526 +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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-07-26 17:45:48 +0300 (Sun, 26 Jul 2015) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.cpp 225 2015-07-26 14:45:48Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 (numChannels != 1 && numChannels != 2)
{
//ST_THROW_RT_ERROR("Illegal number of channels");
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)
{
bSrateSet = true;
// set sample rate, leave other tempo changer parameters as they are.
pTDStretch->setParameters((int)srate);
}
// 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");
}
// Transpose the rate of the new samples if necessary
/* Bypass the nominal setting - can introduce a click in sound when tempo/pitch control crosses the nominal value...
if (rate == 1.0f)
{
// The rate value is same as the original, simply evaluate the tempo changer.
assert(output == pTDStretch);
if (pRateTransposer->isEmpty() == 0)
{
// yet flush the last samples in the pitch transposer buffer
// (may happen if 'rate' changes from a non-zero value to zero)
pTDStretch->moveSamples(*pRateTransposer);
}
pTDStretch->putSamples(samples, nSamples);
}
*/
// accumulate how many samples are expected out from processing, given the current
// 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);
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
// pRateTransposer->clearInput();
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(NULL, &temp, NULL, NULL);
return temp;
case SETTING_SEEKWINDOW_MS:
pTDStretch->getParameters(NULL, NULL, &temp, NULL);
return temp;
case SETTING_OVERLAP_MS:
pTDStretch->getParameters(NULL, NULL, NULL, &temp);
return temp;
case SETTING_NOMINAL_INPUT_SEQUENCE :
return pTDStretch->getInputSampleReq();
case SETTING_NOMINAL_OUTPUT_SEQUENCE :
return pTDStretch->getOutputBatchSize();
default :
return 0;
}
}
// Clears all the samples in the object's output and internal processing
// buffers.
void SoundTouch::clear()
{
samplesExpectedOut = 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;
}

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//////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-09-20 10:38:32 +0300 (Sun, 20 Sep 2015) $
// File revision : $Revision: 4 $
//
// $Id: SoundTouch.h 230 2015-09-20 07:38:32Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 "1.9.2"
/// SoundTouch library version id
#define SOUNDTOUCH_VERSION_ID (10902)
//
// 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 nominal average processing sequence
/// size in samples. This value tells approcimate value how many input samples
/// SoundTouch needs to gather before it does DSP processing run for the sample batch.
///
/// Notices:
/// - This is read-only parameter, i.e. setSetting ignores this parameter
/// - Returned value is approximate average value, exact processing batch
/// size may wary from time to time
/// - This parameter value is not constant but may 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
/// - Returned value is approximate average value, exact processing batch
/// size may wary from time to time
/// - This parameter value is not constant but may change depending on
/// tempo/pitch/rate/samplerate settings.
#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7
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();
/// 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);
/// 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.
);
/// 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.
);
/// 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.
);
/// Clears all the samples in the object's output and internal processing
/// buffers.
virtual void clear();
/// Changes a setting controlling the processing system behaviour. See the
/// 'SETTING_...' defines for available setting ID's.
///
/// \return 'true' if the setting was succesfully 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;
/// 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

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<PropertyGroup Condition="'$(Configuration)'=='Debug'" Label="Configuration">
<UseDebugLibraries>true</UseDebugLibraries>
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<Import Project="..\..\Source\VSProps\ClDisableAllWarnings.props" />
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<ItemGroup>
<ClCompile Include="AAFilter.cpp" />
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<ClCompile Include="cpu_detect_x86.cpp" />
<ClCompile Include="FIFOSampleBuffer.cpp" />
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<ClCompile Include="InterpolateLinear.cpp" />
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<ClCompile Include="mmx_optimized.cpp" />
<ClCompile Include="PeakFinder.cpp" />
<ClCompile Include="RateTransposer.cpp" />
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</Project>

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 4 $
//
// $Id: TDStretch.h 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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;
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();
/// 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();
/// 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 NULL, 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
);
/// 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;
}
};
// 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);
double calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm);
virtual void overlapStereo(short *output, const short *input) const;
virtual void clearCrossCorrState();
};
#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);
double calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm);
};
#endif /// SOUNDTOUCH_ALLOW_SSE
}
#endif /// TDStretch_H

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect.h 11 2008-02-10 16:26:55Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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
/// 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_

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2014-01-07 20:24:28 +0200 (Tue, 07 Jan 2014) $
// File revision : $Revision: 4 $
//
// $Id: cpu_detect_x86.cpp 183 2014-01-07 18:24:28Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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
}

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 4 $
//
// $Id: mmx_optimized.cpp 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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)
{
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 = NULL;
filterCoeffsUnalign = NULL;
}
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;
}
#endif // SOUNDTOUCH_ALLOW_MMX

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////////////////////////////////////////////////////////////////////////////////
///
/// 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
///
////////////////////////////////////////////////////////////////////////////////
//
// Last changed : $Date: 2015-08-09 00:00:15 +0300 (Sun, 09 Aug 2015) $
// File revision : $Revision: 4 $
//
// $Id: sse_optimized.cpp 226 2015-08-08 21:00:15Z oparviai $
//
////////////////////////////////////////////////////////////////////////////////
//
// License :
//
// 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 SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION
// 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 = NULL;
filterCoeffsUnalign = NULL;
}
FIRFilterSSE::~FIRFilterSSE()
{
delete[] filterCoeffsUnalign;
filterCoeffsAlign = NULL;
filterCoeffsUnalign = NULL;
}
// (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 != NULL);
assert(dest != NULL);
assert((length % 8) == 0);
assert(filterCoeffsAlign != NULL);
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

28
src/audio_core/CMakeLists.txt
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@ -2,14 +2,8 @@ set(SRCS
audio_core.cpp
codec.cpp
hle/dsp.cpp
hle/effects.cpp
hle/filter.cpp
hle/final.cpp
hle/pipe.cpp
hle/source.cpp
interpolate.cpp
null_sink.cpp
time_stretch.cpp
)
set(HEADERS
@ -17,31 +11,11 @@ set(HEADERS
codec.h
hle/common.h
hle/dsp.h
hle/effects.h
hle/filter.h
hle/final.h
hle/pipe.h
hle/source.h
interpolate.h
null_sink.h
sink.h
time_stretch.h
)
if(SDL2_FOUND)
set(SRCS ${SRCS} sdl2_sink.cpp)
set(HEADERS ${HEADERS} sdl2_sink.h)
include_directories(${SDL2_INCLUDE_DIR})
endif()
include_directories(../../externals/soundtouch)
include_directories(../../externals/rubberband/rubberband/rubberband)
create_directory_groups(${SRCS} ${HEADERS})
add_library(audio_core STATIC ${SRCS} ${HEADERS})
target_link_libraries(audio_core SoundTouch rubberband)
if(SDL2_FOUND)
target_link_libraries(audio_core ${SDL2_LIBRARY})
endif()
add_library(audio_core STATIC ${SRCS} ${HEADERS})

1
src/audio_core/audio_core.cpp
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@ -4,7 +4,6 @@
#include "audio_core/audio_core.h"
#include "audio_core/hle/dsp.h"
#include "audio_core/sdl2_sink.h"
#include "core/core_timing.h"
#include "core/hle/kernel/vm_manager.h"

8
src/audio_core/audio_core.h
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@ -4,17 +4,15 @@
#pragma once
#include <cstddef>
namespace Kernel {
class VMManager;
}
namespace AudioCore {
constexpr size_t num_sources = 24;
constexpr size_t samples_per_frame = 160; ///< Samples per audio frame at native sample rate
constexpr unsigned native_sample_rate = 32728; ///< 32kHz
constexpr int num_sources = 24;
constexpr int samples_per_frame = 160; ///< Samples per audio frame at native sample rate
constexpr int native_sample_rate = 32728; ///< 32kHz
/// Initialise Audio Core
void Init();

4
src/audio_core/codec.cpp
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@ -77,7 +77,9 @@ StereoBuffer16 DecodeADPCM(const u8* const data, const size_t sample_count, cons
}
static s16 SignExtendS8(u8 x) {
return s16(u16(x) << 8);
// The data is actually signed PCM8.
// We sign extend this to signed PCM16.
return static_cast<s16>(static_cast<s8>(x));
}
StereoBuffer16 DecodePCM8(const unsigned num_channels, const u8* const data, const size_t sample_count) {

0
src/audio_core/hle/common.cpp
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107
src/audio_core/hle/dsp.cpp
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@ -2,23 +2,8 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <atomic>
#include <thread>
#include "audio_core/audio_core.h"
#include "audio_core/hle/effects.h"
#include "audio_core/hle/dsp.h"
#include "audio_core/hle/final.h"
#include "audio_core/hle/pipe.h"
#include "audio_core/hle/source.h"
#include "audio_core/sink.h"
#include "audio_core/time_stretch.h"
#include "common/microprofile.h"
#include "common/profiler.h"
#include "common/thread.h"
#include "core/hle/service/dsp_dsp.h"
namespace DSP {
namespace HLE {
@ -26,104 +11,14 @@ namespace HLE {
SharedMemory g_region0;
SharedMemory g_region1;
static void ThreadFunc();
static Common::Barrier ThreadFunc_barrier(2);
static std::atomic<bool> ThreadFunc_quit = true;
void Init() {
ResetPipes();
SourceInit();
EffectsInit();
FinalInit();
TimeStretch::Init();
ThreadFunc_quit = false;
std::thread thread(ThreadFunc);
thread.detach();
// Lioncash won't like this. Don't do it.
std::memset(&g_region0, 0, sizeof(SharedMemory));
std::memset(&g_region1, 0, sizeof(SharedMemory));
DSP::HLE::ResetPipes();
}
void Shutdown() {
TimeStretch::Shutdown();
if (!ThreadFunc_quit) {
ThreadFunc_quit = true;
ThreadFunc_barrier.Sync();
}
}
static bool next_region_is_ready = true;
unsigned num_frames = 500;
double time_for_a_frame = 0.005;
static Common::Profiling::TimingCategory profile_tick("DSP::Tick");
static Common::Profiling::TimingCategory profile_work("DSP::Work");
MICROPROFILE_DEFINE(DSP_Tick, "DSP", "Tick", MP_RGB(204, 204, 0));
MICROPROFILE_DEFINE(DSP_Work, "DSP", "Work", MP_RGB(153, 153, 0));
static void DoWork() {
Common::Profiling::ScopeTimer timer_work(profile_work);
MICROPROFILE_SCOPE(DSP_Work);
auto& region = CurrentRegion();
for (int i = 0; i < AudioCore::num_sources; i++) {
auto& config = region.source_configurations.config[i];
auto& coeffs = region.adpcm_coefficients.coeff[i];
auto& status = region.source_statuses.status[i];
SourceUpdate(i, config, coeffs, status);
}
EffectsUpdate(region.dsp_configuration, region.intermediate_mix_samples);
FinalUpdate(region.dsp_configuration, region.dsp_status, region.final_samples);
StereoFrame16 samples = FinalFrame();
#if 0
std::vector<s16> output;
output.reserve(AudioCore::samples_per_frame * 2);
for (int i = 0; i < AudioCore::samples_per_frame; i++) {
output.push_back(samples[0][i]);
output.push_back(samples[1][i]);
}
AudioCore::sink->EnqueueSamples(output);
#else
TimeStretch::Tick(AudioCore::sink->SamplesInQueue());
TimeStretch::AddSamples(samples);
TimeStretch::OutputSamples([&](const std::vector<s16>& output) {
if (AudioCore::sink->SamplesInQueue() < 16000) {
AudioCore::sink->EnqueueSamples(output);
}
});
#endif
}
static void ThreadFunc() {
while (true) {
ThreadFunc_barrier.Sync();
if (ThreadFunc_quit) {
break;
}
DoWork();
}
}
bool Tick() {
Common::Profiling::ScopeTimer timer_tick(profile_tick);
MICROPROFILE_SCOPE(DSP_Tick);
if (GetDspState() != DspState::On || !DSP_DSP::SemaphoreSignalled())
return false;
ThreadFunc_barrier.Sync();
return true;
}

37
src/audio_core/hle/dsp.h
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@ -313,10 +313,10 @@ ASSERT_DSP_STRUCT(SourceConfiguration::Configuration::Buffer, 20);
struct SourceStatus {
struct Status {
u8 is_enabled; ///< Is this channel enabled? (Doesn't have to be playing anything.)
u8 current_buffer_id_dirty; ///< Non-zero when current_buffer_id changes
u8 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes
u16_le sync; ///< Is set by the DSP to the value of SourceConfiguration::sync
u32_dsp buffer_position; ///< Number of samples into the current buffer
u16_le current_buffer_id; ///< Updated when a buffer finishes playing
u16_le previous_buffer_id; ///< Updated when a buffer finishes playing
INSERT_PADDING_DSPWORDS(1);
};
@ -330,13 +330,6 @@ struct DspConfiguration {
union {
u32_le dirty_raw;
BitField<0, 1, u32_le> unknown10_dirty;
BitField<1, 1, u32_le> unknown11_dirty;
BitField<2, 1, u32_le> unknown12_dirty;
BitField<3, 1, u32_le> unknown13_dirty;
BitField<4, 1, u32_le> unknown14_dirty;
BitField<5, 1, u32_le> unknown15_dirty;
BitField<8, 1, u32_le> mixer1_enabled_dirty;
BitField<9, 1, u32_le> mixer2_enabled_dirty;
BitField<10, 1, u32_le> delay_effect_0_dirty;
@ -344,7 +337,6 @@ struct DspConfiguration {
BitField<12, 1, u32_le> reverb_effect_0_dirty;
BitField<13, 1, u32_le> reverb_effect_1_dirty;
BitField<15, 1, u32_le> unknown17_dirty;
BitField<16, 1, u32_le> volume_0_dirty;
BitField<24, 1, u32_le> volume_1_dirty;
@ -352,16 +344,12 @@ struct DspConfiguration {
BitField<26, 1, u32_le> output_format_dirty;
BitField<27, 1, u32_le> limiter_enabled_dirty;
BitField<28, 1, u32_le> headphones_connected_dirty;
BitField<30, 1, u32_le> unknown16_dirty;
BitField<31, 1, u32_le> unknown18_dirty;
};
/// The DSP has three intermediate audio mixers. This controls the volume level (0.0-1.0) for each at the final mixer
float_le volume[3];
u16 unknown17;
INSERT_PADDING_DSPWORDS(2);
INSERT_PADDING_DSPWORDS(3);
enum class OutputFormat : u16_le {
Mono = 0,
@ -373,10 +361,7 @@ struct DspConfiguration {
u16_le limiter_enabled; ///< Not sure of the exact gain equation for the limiter.
u16_le headphones_connected; ///< Application updates the DSP on headphone status.
INSERT_PADDING_DSPWORDS(1); ///< TODO: Surround sound related
u16 unknown16;
u16 unknown15;
u16 unknown18;
INSERT_PADDING_DSPWORDS(4); ///< TODO: Surround sound related
INSERT_PADDING_DSPWORDS(2); ///< TODO: Intermediate mixer 1/2 related
u16_le mixer1_enabled;
u16_le mixer2_enabled;
@ -494,29 +479,25 @@ struct SharedMemory {
AdpcmCoefficients adpcm_coefficients;
struct {
u16 unknown[256];
INSERT_PADDING_DSPWORDS(0x100);
} unknown10;
struct {
u16 unknown[192];
INSERT_PADDING_DSPWORDS(0xC0);
} unknown11;
struct {
u16 unknown[384];
INSERT_PADDING_DSPWORDS(0x180);
} unknown12;
struct {
// biq
u32_dsp unknown[5];
INSERT_PADDING_DSPWORDS(0xA);
} unknown13;
struct {
// biq
u32_dsp unknown[5];
INSERT_PADDING_DSPWORDS(0x13A3);
} unknown14;
INSERT_PADDING_DSPWORDS(0x1399);
u16_le frame_counter;
};
ASSERT_DSP_STRUCT(SharedMemory, 0x8000);

44
src/audio_core/hle/effects.cpp
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@ -1,44 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/audio_core.h"
#include "audio_core/hle/common.h"
#include "audio_core/hle/effects.h"
#include "audio_core/hle/source.h"
namespace DSP {
namespace HLE {
struct State {
QuadFrame32 current_frame;
};
static std::array<State, 3> state;
void EffectsInit() {
state = {};
}
void EffectsUpdate(const DspConfiguration& config, IntermediateMixSamples& samples) {
state[0].current_frame.fill({});
state[1].current_frame.fill({});
state[2].current_frame.fill({});
for (size_t source_id = 0; source_id < AudioCore::num_sources; source_id++) {
SourceFrameMixInto(state[0].current_frame, source_id, 0);
SourceFrameMixInto(state[1].current_frame, source_id, 1);
SourceFrameMixInto(state[2].current_frame, source_id, 2);
}
// TODO: Delay
// TODO: Reverb
}
const QuadFrame32& IntermediateMixFrame(int mix_id) {
return state[mix_id].current_frame;
}
}
}

20
src/audio_core/hle/effects.h
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@ -1,20 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include "audio_core/hle/common.h"
#include "audio_core/hle/dsp.h"
namespace DSP {
namespace HLE {
void EffectsInit();
void EffectsUpdate(const DspConfiguration& config, IntermediateMixSamples& samples);
const QuadFrame32& IntermediateMixFrame(int mix_id);
}
}

62
src/audio_core/hle/final.cpp
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@ -1,62 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include "audio_core/hle/common.h"
#include "audio_core/hle/effects.h"
#include "audio_core/hle/final.h"
#include "common/logging/log.h"
#include "common/math_util.h"
namespace DSP {
namespace HLE {
struct State {
StereoFrame16 current_frame;
std::array<float, 3> volumes = {1.0, 0.0, 0.0};
};
static State state;
void FinalInit() {
state = {};
}
void FinalUpdate(const DspConfiguration& config, DspStatus& status, FinalMixSamples& samples) {
// TODO: Final processing
bool clipping = false;
std::array<QuadFrame32, 3> mix;
for (int k = 0; k < 3; k++) {
mix[k] = IntermediateMixFrame(k);
}
for (int i = 0; i < AudioCore::samples_per_frame; i++) {
for (int j = 0; j < 2; j++) {
s32 value = 0;
for (int k = 0; k < 3; k++) {
value += 0.2 * state.volumes[0] * mix[k][i][j + 0];
value += 0.2 * state.volumes[0] * mix[k][i][j + 2];
}
if (value > 0x8000 || value < -0x7FFF) {
clipping = true;
}
state.current_frame[i][j] = static_cast<s16>(MathUtil::Clamp(value, -0x7FFF, 0x8000));
}
}
if (clipping) {
LOG_ERROR(Audio_DSP, "AUDIO IS CLIPPING");
}
}
const StereoFrame16& FinalFrame() {
return state.current_frame;
}
}
}

19
src/audio_core/hle/final.h
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@ -1,19 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include "audio_core/hle/dsp.h"
namespace DSP {
namespace HLE {
void FinalInit();
void FinalUpdate(const DspConfiguration& config, DspStatus& status, FinalMixSamples& samples);
const StereoFrame16& FinalFrame();
}
}

2
src/audio_core/hle/pipe.cpp
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@ -17,7 +17,7 @@ namespace HLE {
static DspState dsp_state = DspState::Off;
static std::array<std::vector<u8>, DspPipe_MAX> pipe_data;
static std::array<std::vector<u8>, static_cast<size_t>(DspPipe::DspPipe_MAX)> pipe_data;
void ResetPipes() {
for (auto& data : pipe_data) {

2
src/audio_core/hle/pipe.h
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@ -23,8 +23,6 @@ enum class DspPipe {
DspPipe_MAX
};
constexpr size_t DspPipe_MAX = static_cast<size_t>(DspPipe::DspPipe_MAX);
/**
* Read a DSP pipe.
* @param pipe_number The Pipe ID

334
src/audio_core/hle/source.cpp
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@ -1,334 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <algorithm>
#include <array>
#include <cmath>
#include <queue>
#include <vector>
#include "audio_core/codec.h"
#include "audio_core/hle/common.h"
#include "audio_core/hle/source.h"
#include "audio_core/interpolate.h"
#include "common/assert.h"
#include "common/logging/log.h"
#include "core/memory.h"
namespace DSP {
namespace HLE {
using MonoOrStereo = SourceConfiguration::Configuration::MonoOrStereo;
using Format = SourceConfiguration::Configuration::Format;
using DspBuffer = SourceConfiguration::Configuration::Buffer;
struct Buffer {
PAddr physical_address;
u32 length;
u8 adpcm_ps;
u16 adpcm_yn[2];
bool adpcm_dirty;
bool is_looping;
u16 buffer_id;
MonoOrStereo mono_or_stereo;
Format format;
bool from_queue;
bool operator < (const Buffer& other) const {
// We want things with lower id to appear first, unless we have wraparound.
// priority_queue puts a before b when b < a.
if (this->buffer_id < 10 && other.buffer_id > 65520) return true;
if (other.buffer_id < 10 && this->buffer_id > 65520) return false;
return this->buffer_id > other.buffer_id;
}
};
struct State {
size_t source_id;
bool enabled = false;
float rate_multiplier = 1.0;
u16 sync = 0;
std::array<std::array<float, 4>, 3> gains = {};
MonoOrStereo mono_or_stereo = MonoOrStereo::Mono;
Format format = Format::PCM16;
std::array<s16, 16> adpcm_coeffs = {};
Codec::ADPCMState adpcm_state = {0, 0};
AudioInterp::State interp_state = {};
bool do_not_trigger_update = true;
bool buffer_update = false;
u32 current_buffer_id = 0;
u32 previous_buffer_id = 0;
std::priority_queue<Buffer> queue = {};
u32 current_sample_number = 0;
u32 next_sample_number = 0;
Codec::StereoBuffer16 current_buffer = {};
QuadFrame32 current_frame = {};
};
static void ParseConfig(State& s, SourceConfiguration::Configuration& config, const s16_le adpcm_coeffs[16]) {
if (!config.dirty_raw) {
return;
}
if (config.reset_flag) {
config.reset_flag.Assign(0);
size_t id = s.source_id;
s = {};
s.source_id = id;
LOG_DEBUG(Audio_DSP, "source_id=%zu reset", s.source_id);
}
if (config.enable_dirty) {
config.enable_dirty.Assign(0);
s.enabled = config.enable != 0;
LOG_TRACE(Audio_DSP, "source_id=%zu enable=%d", s.source_id, s.enabled);
}
if (config.sync_dirty) {
config.sync_dirty.Assign(0);
s.sync = config.sync;
LOG_DEBUG(Audio_DSP, "source_id=%zu sync=%u", s.source_id, s.sync);
}
if (config.rate_multiplier_dirty) {
config.rate_multiplier_dirty.Assign(0);
s.rate_multiplier = config.rate_multiplier;
LOG_TRACE(Audio_DSP, "source_id=%zu rate=%f", s.source_id, s.rate_multiplier);
}
if (config.adpcm_coefficients_dirty) {
config.adpcm_coefficients_dirty.Assign(0);
std::copy(adpcm_coeffs, adpcm_coeffs + s.adpcm_coeffs.size(), s.adpcm_coeffs.begin());
LOG_TRACE(Audio_DSP, "source_id=%zu adpcm update", s.source_id);
}
if (config.gain_0_dirty) {
config.gain_0_dirty.Assign(0);
for (int i = 0; i < 4; i++) {
s.gains[0][i] = config.gain[0][i];
LOG_TRACE(Audio_DSP, "source_id=%zu gains[0][%i] = %f", s.source_id, i, s.gains[0][i]);
}
}
if (config.gain_1_dirty) {
config.gain_1_dirty.Assign(0);
for (int i = 0; i < 4; i++) {
s.gains[1][i] = config.gain[1][i];
LOG_TRACE(Audio_DSP, "source_id=%zu gains[1][%i] = %f", s.source_id, i, s.gains[1][i]);
}
}
if (config.gain_2_dirty) {
config.gain_2_dirty.Assign(0);
for (int i = 0; i < 4; i++) {
s.gains[2][i] = config.gain[2][i];
LOG_TRACE(Audio_DSP, "source_id=%zu gains[2][%i] = %f", s.source_id, i, s.gains[2][i]);
}
}
// if (config.unknown_flag) {
//config.unknown_flag = 0;
// LOG_WARNING(Audio_DSP, "(STUB) unknown_flag is set!!!");
// }
if (config.format_dirty || config.embedded_buffer_dirty) {
config.format_dirty.Assign(0);
s.format = config.format;
LOG_DEBUG(Audio_DSP, "source_id=%zu format=%u", s.source_id, s.format);
}
if (config.mono_or_stereo_dirty || config.embedded_buffer_dirty) {
config.mono_or_stereo_dirty.Assign(0);
s.mono_or_stereo = config.mono_or_stereo;
LOG_DEBUG(Audio_DSP, "source_id=%zu mono_or_stereo=%u", s.source_id, s.mono_or_stereo);
}
if (config.buffer_queue_dirty) {
config.buffer_queue_dirty.Assign(0);
for (int i = 0; i < 4; i++) {
if (config.buffers_dirty & (1 << i)) {
const auto& b = config.buffers[i];
s.queue.emplace(Buffer{
b.physical_address,
b.length,
(u8)b.adpcm_ps,
{ b.adpcm_yn[0], b.adpcm_yn[1] },
b.adpcm_dirty != 0,
b.is_looping != 0,
b.buffer_id,
s.mono_or_stereo,
s.format,
true
});
LOG_TRACE(Audio_DSP, "enqueueing queued %i addr=0x%08x len=%u id=%u", i, b.physical_address, b.length, b.buffer_id);
}
}
config.buffers_dirty = 0;
}
if (config.embedded_buffer_dirty) {
config.embedded_buffer_dirty.Assign(0);
s.queue.emplace(Buffer {
config.physical_address,
config.length,
(u8)config.adpcm_ps,
{ config.adpcm_yn[0], config.adpcm_yn[1] },
config.adpcm_dirty.ToBool(),
config.is_looping.ToBool(),
config.buffer_id,
s.mono_or_stereo,
s.format,
false
});
LOG_TRACE(Audio_DSP, "enqueueing embedded addr=0x%08x len=%u id=%u", config.physical_address, config.length, config.buffer_id);
}
if (config.interpolation_dirty) {
config.interpolation_dirty.Assign(0);
//config.interpolation_mode
LOG_DEBUG(Audio_DSP, "source_id=%zu interpolation_mode=%u ", s.source_id, config.interpolation_mode);
}
if (config.dirty_raw) {
LOG_WARNING(Audio_DSP, "source_id=%zu remaining_dirty=%x", s.source_id, config.dirty_raw);
}
config.dirty_raw = 0;
}
static bool DequeueBuffer(State& s) {
if (!s.current_buffer.empty())
return true;
if (s.queue.empty())
return false;
const Buffer buf = s.queue.top();
s.queue.pop();
const u8* const memory = Memory::GetPhysicalPointer(buf.physical_address);
ASSERT(memory);
if (buf.adpcm_dirty) {
s.adpcm_state.yn1 = buf.adpcm_yn[0];
s.adpcm_state.yn2 = buf.adpcm_yn[1];
}
if (buf.is_looping) {
LOG_ERROR(Audio_DSP, "Looped buffers are unimplemented at the moment");
}
const unsigned num_channels = buf.mono_or_stereo == MonoOrStereo::Stereo ? 2 : 1;
switch (buf.format) {
case Format::PCM8:
s.current_buffer = Codec::DecodePCM8(num_channels, memory, buf.length);
break;
case Format::PCM16:
s.current_buffer = Codec::DecodePCM16(num_channels, memory, buf.length);
break;
case Format::ADPCM:
ASSERT(num_channels == 1);
s.current_buffer = Codec::DecodeADPCM(memory, buf.length, s.adpcm_coeffs, s.adpcm_state);
break;
default:
UNIMPLEMENTED();
break;
}
s.current_sample_number = s.next_sample_number = 0;
s.current_buffer_id = buf.buffer_id;
s.buffer_update = buf.from_queue;
LOG_TRACE(Audio_DSP, "source_id=%u buffer_id=%u from_queue=%d", s.source_id, buf.buffer_id, buf.from_queue);
return true;
}
static void ResampleBuffer(State& s) {
s.current_frame.fill({});
s.current_sample_number = s.next_sample_number;
while (true) {
if (!DequeueBuffer(s))
break;
auto result = AudioInterp::None(s.interp_state, s.current_frame, s.current_buffer, s.rate_multiplier);
s.next_sample_number += std::get<0>(result);
if (!std::get<1>(result))
break;
}
}
static void AdvanceFrame(State& s) {
ResampleBuffer(s);
if (s.current_sample_number == s.next_sample_number) {
s.enabled = false;
}
// TODO: Filters
}
static void UpdateStatus(State& s, SourceStatus::Status& status) {
// Applications depend on the correct emulation of
// previous_buffer_id_dirty and previous_buffer_id to synchronise
// audio with video.
status.is_enabled = s.enabled;
status.current_buffer_id_dirty = s.buffer_update ? 1 : 0;
s.buffer_update = false;
status.current_buffer_id = s.current_buffer_id;
status.buffer_position = s.current_sample_number;
status.sync = s.sync;
}
std::array<State, AudioCore::num_sources> state = {};
void SourceInit() {
state = {};
for (size_t i = 0; i < state.size(); i++) {
state[i] = {};
state[i].source_id = i;
}
}
void SourceUpdate(int source_id, SourceConfiguration::Configuration& config, const s16_le adpcm_coeffs[16], SourceStatus::Status& status) {
ASSERT(source_id >= 0 && source_id < AudioCore::num_sources);
ParseConfig(state[source_id], config, adpcm_coeffs);
AdvanceFrame(state[source_id]);
UpdateStatus(state[source_id], status);
}
const QuadFrame32& SourceFrame(int source_id) {
ASSERT(source_id >= 0 && source_id < AudioCore::num_sources);
ASSERT(state[source_id].enabled);
return state[source_id].current_frame;
}
void SourceFrameMixInto(QuadFrame32& dest, int source_id, int intermediate_mix_id) {
ASSERT(source_id >= 0 && source_id < AudioCore::num_sources);
ASSERT(intermediate_mix_id >= 0 && intermediate_mix_id < 3);
const State& s = state[source_id];
if (!s.enabled)
return;
for (int i = 0; i < dest.size(); i++) {
for (int channel = 0; channel < 4; channel++) {
dest[i][channel] += s.gains[intermediate_mix_id][channel] * s.current_frame[i][channel];
}
}
}
}
}

35
src/audio_core/hle/source.h
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@ -1,35 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include "audio_core/audio_core.h"
#include "audio_core/hle/common.h"
#include "audio_core/hle/dsp.h"
namespace DSP {
namespace HLE {
/// Initialise this DSP module
void SourceInit();
/**
* Perform processing for this DSP module.
* This module performs:
* - Buffer management
* - Decoding of buffers
* - Buffer resampling and interpolation
* - Per-source filtering (SimpleFilter, BiquadFilter)
* - Per-source gain
*/
void SourceUpdate(int source_id, SourceConfiguration::Configuration& config, const s16_le adpcm_coeffs[16], SourceStatus::Status& status);
/// Output of this DSP module.
const QuadFrame32& SourceFrame(int source_id);
/// Mix current frame from source_id into buffer based on gain coefficients for intermediate_mix_id.
void SourceFrameMixInto(QuadFrame32& dest, int source_id, int intermediate_mix_id);
}
}

237
src/audio_core/interpolate.cpp
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@ -1,237 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#define _USE_MATH_DEFINES
#include <cmath>
#include "audio_core/interpolate.h"
#include "common/assert.h"
#include "common/math_util.h"
namespace AudioInterp {
/// Kaiser window with alpha=2.4, N=11
static const std::array<double, required_history/2> kaiser_window {{
0.327949,
0.521302,
0.707379,
0.861996,
0.964250
}};
struct BiquadLpf {
/// Calculate coefficients required for a biquad filter to behave as a low-pass filter.
void Init(double freq) {
const double w0 = 2 * M_PI * freq;
const double Q = 0.707;
const double a = sin(w0) / (2.0*Q);
double a0;
b0 = 0.5 * (1.0 - cos(w0));
b1 = (1.0 - cos(w0));
b2 = 0.5 * (1.0 - cos(w0));
a0 = 1.0 + a;
a1 = -2.0 * cos(w0);
a2 = 1.0 - a;
// Normalize;
b0 /= a0;
b1 /= a0;
b2 /= a0;
a1 /= a0;
a2 /= a0;
}
inline s32 Process(s32 x) {
double xn0 = x;
double yn0 = b0 * xn0 + b1 * xn1 + b2 * xn2 - a1 * yn1 - a2 * yn2;
// Advance state
xn2 = xn1;
xn1 = xn0;
yn2 = yn1;
yn1 = yn0;
return (s32)yn0;
}
private:
double a1, a2;
double b0, b1, b2;
double xn1 = 0.0, xn2 = 0.0, yn1 = 0.0, yn2 = 0.0;
};
double sinc(double x) {
DEBUG_ASSERT(x != 0);
return sin(x) / x;
}
std::tuple<size_t, bool> KaiserSinc(State& state, DSP::HLE::QuadFrame32& output, std::array<std::vector<s16>, 2>& input, const float rate_change) {
ASSERT(input[0].size() == input[1].size());
ASSERT(input[0].size() > required_history);
size_t position = 0;
double& position_fractional = state.position_fractional;
for (int j = 0; j < 2; j++) {
input[j].insert(input[j].begin(), state.history[j].begin(), state.history[j].end());
}
std::array<BiquadLpf, 2> lpf;
const double lpf_cutoff = std::min(0.5 * rate_change, 0.5 / rate_change);
lpf[0].Init(lpf_cutoff);
lpf[1].Init(lpf_cutoff);
auto step = [&](size_t i) -> s32 {
auto& in = input[i];
s32 sample = 0;
sample += kaiser_window[0] * sinc(-5.0 - position_fractional) * in[position + 0];
sample += kaiser_window[1] * sinc(-4.0 - position_fractional) * in[position + 1];
sample += kaiser_window[2] * sinc(-3.0 - position_fractional) * in[position + 2];
sample += kaiser_window[3] * sinc(-2.0 - position_fractional) * in[position + 3];
sample += kaiser_window[4] * sinc(-1.0 - position_fractional) * in[position + 4];
sample += in[position + 5];
sample += kaiser_window[4] * sinc(+1.0 - position_fractional) * in[position + 6];
sample += kaiser_window[3] * sinc(+2.0 - position_fractional) * in[position + 7];
sample += kaiser_window[2] * sinc(+3.0 - position_fractional) * in[position + 8];
sample += kaiser_window[1] * sinc(+4.0 - position_fractional) * in[position + 9];
sample += kaiser_window[0] * sinc(+5.0 - position_fractional) * in[position + 10];
//sample = lpf[i].Process(sample);
return sample;
};
const size_t position_stop = input[0].size() - required_history;
while (state.output_position < output[0].size() && position < position_stop) {
s32 sample0 = step(0);
s32 sample1 = step(1);
output[0][state.output_position] = sample0;
output[1][state.output_position] = sample0;
output[2][state.output_position] = sample1;
output[3][state.output_position] = sample1;
position_fractional += rate_change;
position += (size_t)position_fractional;
position_fractional -= (size_t)position_fractional;
state.output_position++;
}
bool continue_feeding_me = true;
if (state.output_position >= output[0].size()) {
state.output_position = 0;
continue_feeding_me = false;
}
for (int j = 0; j < 2; j++) {
std::copy(input[j].begin() + position,
input[j].begin() + position + required_history,
state.history[j].begin());
if (position + required_history >= input[j].size()) {
input[j].clear();
} else {
input[j].erase(input[j].begin(),
input[j].begin() + position + required_history);
}
}
ASSERT(input[0].size() == input[1].size());
return std::make_tuple(position, continue_feeding_me);
}
std::tuple<size_t, bool> Linear(State& state, DSP::HLE::QuadFrame32& output, std::array<std::vector<s16>, 2>& input, const float rate_change) {
ASSERT(input[0].size() == input[1].size());
while (input[0].size() < 2) {
input[0].emplace_back(0);
input[1].emplace_back(0);
}
size_t position = 0;
double& position_fractional = state.position_fractional;
auto step = [&](size_t i) -> s32 {
auto& in = input[i];
s32 sample = 0;
sample = position_fractional * in[position + 0] + (1.0 - position_fractional) * in[position + 1];
return sample;
};
const size_t position_stop = input.size() - 1;
while (state.output_position < output.size() && position < position_stop) {
s32 sample0 = step(0);
s32 sample1 = step(1);
output[state.output_position][0] = sample0;
output[state.output_position][1] = sample0;
output[state.output_position][2] = sample1;
output[state.output_position][3] = sample1;
position_fractional += rate_change;
position += (size_t)position_fractional;
position_fractional -= (size_t)position_fractional;
state.output_position++;
}
bool continue_feeding_me = true;
if (state.output_position >= output[0].size()) {
state.output_position = 0;
continue_feeding_me = false;
}
for (int j = 0; j < 2; j++) {
if (position >= input[j].size()) {
input[j].clear();
} else {
input[j].erase(input[j].begin(),
input[j].begin() + position);
}
}
ASSERT(input[0].size() == input[1].size());
return std::make_tuple(position, continue_feeding_me);
}
std::tuple<size_t, bool> None(State& state, DSP::HLE::QuadFrame32& output, std::vector<std::array<s16, 2>>& input, const float rate_change) {
size_t position = 0;
double& position_fractional = state.position_fractional;
auto step = [&](size_t i) -> s32 {
return input[position][i];
};
const size_t position_stop = input.size();
while (state.output_position < output.size() && position < position_stop) {
s32 sample0 = step(0);
s32 sample1 = step(1);
output[state.output_position][0] = sample0;
output[state.output_position][1] = sample0;
output[state.output_position][2] = sample1;
output[state.output_position][3] = sample1;
position_fractional += rate_change;
position += (size_t)position_fractional;
position_fractional -= (size_t)position_fractional;
state.output_position++;
}
bool continue_feeding_me = true;
if (state.output_position >= output.size()) {
state.output_position = 0;
continue_feeding_me = false;
}
if (position >= input.size()) {
input.clear();
} else {
input.erase(input.begin(), input.begin() + position);
}
return std::make_tuple(position, continue_feeding_me);
}
}

28
src/audio_core/interpolate.h
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@ -1,28 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <array>
#include <vector>
#include "audio_core/hle/common.h"
#include "common/common_types.h"
namespace AudioInterp {
constexpr size_t required_history = 11;
struct State {
double position_fractional = 0;
size_t output_position = 0;
std::array<std::array<s16, required_history>, 2> history = {};
};
std::tuple<size_t, bool> KaiserSinc(State& state, DSP::HLE::QuadFrame32& output, std::array<std::vector<s16>, 2>& input, const float rate_change);
std::tuple<size_t, bool> Linear(State& state, DSP::HLE::QuadFrame32& output, std::array<std::vector<s16>, 2>& input, const float rate_change);
std::tuple<size_t, bool> None(State& state, DSP::HLE::QuadFrame32& output, std::vector<std::array<s16, 2>>& input, const float rate_change);
}

3
src/audio_core/null_sink.cpp
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@ -1,3 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.

9
src/audio_core/null_sink.h
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@ -1,9 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
namespace AudioCore {
}

118
src/audio_core/sdl2_sink.cpp
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@ -1,118 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <SDL.h>
#include "audio_core/audio_core.h"
#include "audio_core/sdl2_sink.h"
#include "common/assert.h"
#include "common/logging/log.h"
namespace AudioCore {
std::unique_ptr<Sink> sink(new SDL2Sink());
SDL2Sink::SDL2Sink() {
if (SDL_Init(SDL_INIT_AUDIO) < 0) {
LOG_CRITICAL(Audio_SDL2, "SDL_Init(SDL_INIT_AUDIO) failed");
exit(-2);
}
SDL_AudioSpec desired_audiospec;
SDL_zero(desired_audiospec);
desired_audiospec.format = AUDIO_S16;
desired_audiospec.channels = 2;
desired_audiospec.freq = AudioCore::native_sample_rate;
desired_audiospec.samples = 4096;
desired_audiospec.userdata = this;
desired_audiospec.callback = &SDL2Sink::Callback; // We're going to use SDL_QueueAudio
SDL_AudioSpec obtained_audiospec;
SDL_zero(obtained_audiospec);
audio_device_id = SDL_OpenAudioDevice(nullptr, /*iscapture=*/false, &desired_audiospec, &obtained_audiospec, 0);
if (audio_device_id < 0) {
LOG_CRITICAL(Audio_SDL2, "SDL_OpenAudioDevice failed");
exit(-2);
}
sample_rate = obtained_audiospec.freq;
SDL_PauseAudioDevice(audio_device_id, 0);
}
SDL2Sink::~SDL2Sink() {
}
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units: samples/sec)
unsigned SDL2Sink::GetNativeSampleRate() const {
return sample_rate;
}
/**
* Feed stereo samples to sink.
* @param samples Samples in interleaved stereo PCM16 format. Size of vector must be multiple of two.
*/
void SDL2Sink::EnqueueSamples(const std::vector<s16>& samples) {
ASSERT(samples.size() % 2 == 0);
SDL_LockAudioDevice(audio_device_id);
queue.emplace_back(samples);
SDL_UnlockAudioDevice(audio_device_id);
}
/// Samples enqueued that have not been played yet.
size_t SDL2Sink::SamplesInQueue() const {
const size_t queue_size = RealQueueSize() + dequeue_consumed;
if (dequeue_consumed == 0)
return queue_size;
const std::chrono::duration<double> duration = std::chrono::steady_clock::now() - dequeue_time;
const size_t estimated_samples_consumed = sample_rate * duration.count();
if (estimated_samples_consumed > queue_size)
return 0;
return queue_size - estimated_samples_consumed;
}
size_t SDL2Sink::RealQueueSize() const {
size_t total_size = 0;
SDL_LockAudioDevice(audio_device_id);
for (const auto& buf : queue) {
total_size += buf.size() / 2;
}
SDL_UnlockAudioDevice(audio_device_id);
return total_size;
}
void SDL2Sink::Callback(void* sink_, u8* buffer, int buffer_size) {
SDL2Sink* sink = reinterpret_cast<SDL2Sink*>(sink_);
buffer_size /= sizeof(s16); // Convert to number of half-samples.
sink->dequeue_time = std::chrono::steady_clock::now();
sink->dequeue_consumed = buffer_size / 2;
while (buffer_size > 0 && !sink->queue.empty()) {
if (sink->queue.front().size() <= buffer_size) {
memcpy(buffer, sink->queue.front().data(), sink->queue.front().size() * sizeof(s16));
buffer += sink->queue.front().size() * sizeof(s16);
buffer_size -= sink->queue.front().size();
sink->queue.pop_front();
} else {
memcpy(buffer, sink->queue.front().data(), buffer_size * sizeof(s16));
buffer += buffer_size * sizeof(s16);
sink->queue.front().erase(sink->queue.front().begin(), sink->queue.front().begin() + buffer_size);
buffer_size = 0;
}
}
if (buffer_size > 0) {
sink->dequeue_consumed -= buffer_size / 2;
memset(buffer, 0, buffer_size * sizeof(s16));
}
}
}

46
src/audio_core/sdl2_sink.h
View File

@ -1,46 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <chrono>
#include <cstddef>
#include <list>
#include <vector>
#include "audio_core/sink.h"
namespace AudioCore {
class SDL2Sink final : public Sink {
public:
SDL2Sink();
~SDL2Sink() override;
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units: samples/sec)
unsigned GetNativeSampleRate() const override;
/**
* Feed stereo samples to sink.
* @param samples Samples in interleaved stereo PCM16 format. Size of vector must be multiple of two.
*/
void EnqueueSamples(const std::vector<s16>& samples) override;
/// Samples enqueued that have not been played yet.
size_t SamplesInQueue() const override;
private:
using SDL_AudioDeviceID = u32;
unsigned sample_rate;
SDL_AudioDeviceID audio_device_id;
std::list<std::vector<s16>> queue;
size_t RealQueueSize() const;
static void Callback(void* sink, u8* buffer, int buffer_size);
std::chrono::steady_clock::time_point dequeue_time;
size_t dequeue_consumed = 0;
};
}

3
src/audio_core/sink.h
View File

@ -4,7 +4,6 @@
#pragma once
#include <memory>
#include <vector>
#include "common/common_types.h"
@ -32,6 +31,4 @@ public:
virtual std::size_t SamplesInQueue() const = 0;
};
extern std::unique_ptr<Sink> sink;
} // namespace

83
src/audio_core/time_stretch.cpp
View File

@ -1,83 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <array>
#include <chrono>
#include <cmath>
#include <functional>
#include <list>
#include <numeric>
#include <vector>
#include <SoundTouch.h>
#include "audio_core/audio_core.h"
#include "audio_core/sink.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/math_util.h"
#include "common/logging/log.h"
namespace TimeStretch {
static soundtouch::SoundTouch soundtouch;
using steady_clock = std::chrono::steady_clock;
steady_clock::time_point frame_timer = steady_clock::now();
double smooth_ratio = 1.0;
void Tick(unsigned samples_in_queue) {
const steady_clock::time_point now = steady_clock::now();
const std::chrono::duration<double> duration = now - frame_timer;
frame_timer = now;
constexpr double native_frame_time = (double)AudioCore::samples_per_frame / (double)AudioCore::native_sample_rate;
const double actual_frame_time = duration.count();
double ratio = actual_frame_time / native_frame_time;
ratio = MathUtil::Clamp<double>(ratio, 0.01, 100.0);
// TODO: Uhh was just reading this and this seems super wonky double-check your logic wtf are you thinking.
if (samples_in_queue < 4096) {
ratio = ratio > 1.0 ? ratio * ratio : 1.0;
ratio = MathUtil::Clamp<double>(ratio, 0.01, 100.0);
} else if (AudioCore::sink->SamplesInQueue() > 16000) {
ratio = ratio > 1.0 ? sqrt(ratio) : 0.01;
ratio = MathUtil::Clamp<double>(ratio, 0.01, 100.0);
}
smooth_ratio = 0.993 * smooth_ratio + 0.007 * ratio;
smooth_ratio = MathUtil::Clamp<double>(smooth_ratio, 0.01, 100.0);
//printf("%f, %f\n", ratio, smooth_ratio);
soundtouch.setTempo(1.0 / smooth_ratio);
}
void Init() {
soundtouch.setTempo(1.0);
soundtouch.setChannels(2);
}
void Shutdown() {
soundtouch.setTempo(1.0);
}
void AddSamples(const std::array<std::array<s16, 2>, AudioCore::samples_per_frame>& samples) {
// FIXME: lol don't do this c-style cast
soundtouch.putSamples((s16*)samples.data(), AudioCore::samples_per_frame);
}
void OutputSamples(std::function<void(const std::vector<s16>&)> fn) {
size_t available = soundtouch.numSamples();
std::vector<s16> output(available * 2);
soundtouch.receiveSamples(output.data(), available);
fn(output);
}
}

24
src/audio_core/time_stretch.h
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@ -1,24 +0,0 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <array>
#include <functional>
#include <vector>
#include "common/common_types.h"
namespace TimeStretch {
void Init();
void Shutdown();
void Tick(unsigned samples_in_queue);
void AddSamples(const std::array<std::array<s16, 2>, AudioCore::samples_per_frame>& samples);
void OutputSamples(std::function<void(const std::vector<s16>&)> fn);
}

2
src/citra_qt/CMakeLists.txt
View File

@ -117,8 +117,6 @@ if (Qt5_FOUND AND MSVC)
)
windows_copy_files(citra-qt ${Qt5_PLATFORMS_DIR} ${PLATFORMS} qwindows$<$<CONFIG:Debug>:d>.*)
windows_copy_files(citra-qt ${SDL2_DLL_DIR} ${DLL_DEST} SDL2.dll)
unset(Qt5_DLL_DIR)
unset(Qt5_PLATFORMS_DIR)
unset(DLL_DEST)

1
src/common/logging/backend.cpp
View File

@ -65,7 +65,6 @@ namespace Log {
SUB(Render, OpenGL) \
CLS(Audio) \
SUB(Audio, DSP) \
SUB(Audio, SDL2) \
CLS(Loader)
// GetClassName is a macro defined by Windows.h, grrr...

1
src/common/logging/log.h
View File

@ -80,7 +80,6 @@ enum class Class : ClassType {
Render_OpenGL, ///< OpenGL backend
Audio, ///< Emulator audio output
Audio_DSP, ///< The HLE implementation of the DSP
Audio_SDL2, ///< SDL2 frontend for audio output
Loader, ///< ROM loader
Count ///< Total number of logging classes

128
src/core/hle/service/dsp_dsp.cpp
View File

@ -20,55 +20,29 @@ namespace DSP_DSP {
static u32 read_pipe_count;
static Kernel::SharedPtr<Kernel::Event> semaphore_event;
enum class InterruptType {
Zero = 0, // Unknown purpose. Channel is always zero.
One = 1, // Unknown purpose. Channel is always zero.
Pipe = 2, // Related to a pipe
MAX
};
constexpr size_t InterruptType_MAX = static_cast<size_t>(InterruptType::MAX);
/// Map of (interrupt number, channel number) to Kernel::Events. See: RegisterInterruptEvents
static std::array<std::unordered_map<u32, Kernel::SharedPtr<Kernel::Event>>, InterruptType_MAX> interrupt_events;
constexpr size_t max_number_of_interrupt_events = 6;
size_t GetNumberOfRegisteredEvents() {
size_t number = 0;
for (const auto& events : interrupt_events) {
number += events.size();
struct PairHash {
template <typename T, typename U>
std::size_t operator()(const std::pair<T, U> &x) const {
// TODO(yuriks): Replace with better hash combining function.
return std::hash<T>()(x.first) ^ std::hash<U>()(x.second);
}
return number;
}
};
/// Map of (audio interrupt number, channel number) to Kernel::Events. See: RegisterInterruptEvents
static std::unordered_map<std::pair<u32, u32>, Kernel::SharedPtr<Kernel::Event>, PairHash> interrupt_events;
// DSP Interrupts:
// Interrupt (2, 2) occurs every frame tick. Userland programs normally have a thread that's waiting
// Interrupt #2 occurs every frame tick. Userland programs normally have a thread that's waiting
// for an interrupt event. Immediately after this interrupt event, userland normally updates the
// state in the next region and increments the relevant frame counter by two.
void SignalAllInterrupts() {
// HACK: The other interrupts have currently unknown purpose, we trigger them each tick in any case.
for (auto& events : interrupt_events)
for (auto& event : events)
event.second->Signal();
for (auto& interrupt_event : interrupt_events)
interrupt_event.second->Signal();
}
void SignalInterrupt(u32 interrupt, u32 channel) {
ASSERT(interrupt < interrupt_events.size());
if (interrupt == 0 || interrupt == 1)
ASSERT(channel == 0);
auto& events = interrupt_events[interrupt];
if (events.find(channel) != events.end()) {
events[channel]->Signal();
}
}
bool SemaphoreSignalled() {
if (semaphore_event->signaled) {
semaphore_event->Clear();
return true;
} else {
return false;
}
interrupt_events[std::make_pair(interrupt, channel)]->Signal();
}
/**
@ -84,7 +58,6 @@ static void ConvertProcessAddressFromDspDram(Service::Interface* self) {
u32 addr = cmd_buff[1];
cmd_buff[0] = 0xC0080;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
cmd_buff[2] = (addr << 1) + (Memory::DSP_RAM_VADDR + 0x40000);
@ -140,11 +113,9 @@ static void LoadComponent(Service::Interface* self) {
static void GetSemaphoreEventHandle(Service::Interface* self) {
u32* cmd_buff = Kernel::GetCommandBuffer();
cmd_buff[0] = 0x160042;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
cmd_buff[3] = Kernel::g_handle_table.Create(semaphore_event).MoveFrom(); // Event handle
LOG_WARNING(Service_DSP, "(STUBBED) called");
}
@ -186,47 +157,23 @@ static void FlushDataCache(Service::Interface* self) {
static void RegisterInterruptEvents(Service::Interface* self) {
u32* cmd_buff = Kernel::GetCommandBuffer();
u32 type_num = cmd_buff[1];
u32 interrupt = cmd_buff[1];
u32 channel = cmd_buff[2];
u32 event_handle = cmd_buff[4];
if (cmd_buff[3] != 0) {
cmd_buff[0] = 0x40;
cmd_buff[1] = 0xD9001830;
return;
}
InterruptType type = static_cast<InterruptType>(type_num);
if (type == InterruptType::Zero || type == InterruptType::One) {
channel = 0;
} else if (type == InterruptType::Pipe) {
if (channel >= DSP::HLE::DspPipe_MAX) {
LOG_ERROR(Service_DSP, "Invalid (type, channel) combination (%u, %u)", type, channel);
}
} else {
// I suspect that interrupt values greater than two are invalid.
LOG_ERROR(Service_DSP, "Unimplemented (type, channel) combination (%u, %u)", type, channel);
UNIMPLEMENTED();
}
cmd_buff[0] = 0x150040;
if (event_handle) {
auto evt = Kernel::g_handle_table.Get<Kernel::Event>(cmd_buff[4]);
if (evt) {
if (GetNumberOfRegisteredEvents() < max_number_of_interrupt_events) {
interrupt_events[type_num][channel] = evt;
LOG_INFO(Service_DSP, "Registered type=%u, channel=%u, event_handle=0x%08X", type, channel, event_handle);
} else {
cmd_buff[1] = 0xC860A7FF;
LOG_ERROR(Service_DSP, "Ran out of space to register interrupts");
}
interrupt_events[std::make_pair(interrupt, channel)] = evt;
cmd_buff[1] = RESULT_SUCCESS.raw;
LOG_INFO(Service_DSP, "Registered interrupt=%u, channel=%u, event_handle=0x%08X", interrupt, channel, event_handle);
} else {
LOG_CRITICAL(Service_DSP, "Invalid event handle! type=%u, channel=%u, event_handle=0x%08X", type, channel, event_handle);
LOG_CRITICAL(Service_DSP, "Invalid event handle! interrupt=%u, channel=%u, event_handle=0x%08X", interrupt, channel, event_handle);
ASSERT(false); // This should really be handled at a IPC translation layer.
}
} else {
interrupt_events[type_num].erase(channel);
LOG_INFO(Service_DSP, "Unregistered type=%u, channel=%u, event_handle=0x%08X", type, channel, event_handle);
interrupt_events.erase(std::make_pair(interrupt, channel));
LOG_INFO(Service_DSP, "Unregistered interrupt=%u, channel=%u, event_handle=0x%08X", interrupt, channel, event_handle);
}
}
@ -240,13 +187,8 @@ static void RegisterInterruptEvents(Service::Interface* self) {
static void SetSemaphore(Service::Interface* self) {
u32* cmd_buff = Kernel::GetCommandBuffer();
cmd_buff[0] = 0x70040;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
// Observed Behaviour: Waits for DSP_PSEM to be clear then sets DSP_PSEM.
SignalAllInterrupts(); // This is a HACK
LOG_WARNING(Service_DSP, "(STUBBED) called");
}
@ -268,13 +210,7 @@ static void WriteProcessPipe(Service::Interface* self) {
u32 size = cmd_buff[2];
u32 buffer = cmd_buff[4];
if (IPC::StaticBufferDesc(size, 1) != cmd_buff[3]) {
LOG_ERROR(Service_DSP, "IPC static buffer descriptor failed validation (0x%X). pipe=%u, size=0x%X, buffer=0x%08X", cmd_buff[3], pipe, size, buffer);
cmd_buff[0] = 0x40;
cmd_buff[1] = 0xD9001830;
return;
}
ASSERT_MSG(IPC::StaticBufferDesc(size, 1) == cmd_buff[3], "IPC static buffer descriptor failed validation (0x%X). pipe=%u, size=0x%X, buffer=0x%08X", cmd_buff[3], pipe, size, buffer);
ASSERT_MSG(Memory::GetPointer(buffer) != nullptr, "Invalid Buffer: pipe=%u, size=0x%X, buffer=0x%08X", pipe, size, buffer);
std::vector<u8> message(size);
@ -285,7 +221,6 @@ static void WriteProcessPipe(Service::Interface* self) {
DSP::HLE::PipeWrite(pipe, message);
cmd_buff[0] = 0xD0040;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
LOG_DEBUG(Service_DSP, "pipe=%u, size=0x%X, buffer=0x%08X", pipe, size, buffer);
@ -315,14 +250,16 @@ static void ReadPipeIfPossible(Service::Interface* self) {
ASSERT_MSG(Memory::GetPointer(addr) != nullptr, "Invalid addr: pipe=0x%08X, unknown=0x%08X, size=0x%X, buffer=0x%08X", pipe, unknown, size, addr);
cmd_buff[0] = 0x100082;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
if (DSP::HLE::GetPipeReadableSize(pipe) >= size) {
std::vector<u8> response = DSP::HLE::PipeRead(pipe, size);
std::vector<u8> response = DSP::HLE::PipeRead(pipe, size);
Memory::WriteBlock(addr, response.data(), response.size());
Memory::WriteBlock(addr, response.data(), response.size());
cmd_buff[2] = static_cast<u32>(response.size());
cmd_buff[2] = static_cast<u32>(response.size());
} else {
cmd_buff[2] = 0; // Return no data
}
LOG_DEBUG(Service_DSP, "pipe=0x%08X, unknown=0x%08X, size=0x%X, buffer=0x%08X, return cmd_buff[2]=0x%08X", pipe, unknown, size, addr, cmd_buff[2]);
}
@ -396,7 +333,6 @@ static void SetSemaphoreMask(Service::Interface* self) {
u32 mask = cmd_buff[1];
cmd_buff[0] = 0x170040;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
LOG_WARNING(Service_DSP, "(STUBBED) called mask=0x%08X", mask);
@ -414,11 +350,10 @@ static void SetSemaphoreMask(Service::Interface* self) {
static void GetHeadphoneStatus(Service::Interface* self) {
u32* cmd_buff = Kernel::GetCommandBuffer();
cmd_buff[0] = 0x1F0080;
cmd_buff[1] = RESULT_SUCCESS.raw; // No error
cmd_buff[2] = 0; // Not using headphones?
LOG_TRACE(Service_DSP, "called");
LOG_WARNING(Service_DSP, "(STUBBED) called");
}
/**
@ -441,7 +376,6 @@ static void RecvData(Service::Interface* self) {
// Application reads this after requesting DSP shutdown, to verify the DSP has indeed shutdown or slept.
cmd_buff[0] = 0x10080;
cmd_buff[1] = RESULT_SUCCESS.raw;
switch (DSP::HLE::GetDspState()) {
case DSP::HLE::DspState::On:
@ -477,7 +411,6 @@ static void RecvDataIsReady(Service::Interface* self) {
ASSERT_MSG(register_number == 0, "Unknown register_number %u", register_number);
cmd_buff[0] = 0x20080;
cmd_buff[1] = RESULT_SUCCESS.raw;
cmd_buff[2] = 1; // Ready to read
@ -532,8 +465,7 @@ Interface::Interface() {
Interface::~Interface() {
semaphore_event = nullptr;
for (auto& events : interrupt_events)
events.clear();
interrupt_events.clear();
}
} // namespace

3
src/core/hle/service/dsp_dsp.h
View File

@ -34,7 +34,4 @@ void SignalAllInterrupts();
*/
void SignalInterrupt(u32 interrupt_id, u32 channel_id);
/// Returns true it the application signalled the semaphore, then clears the semaphore.
bool SemaphoreSignalled();
} // namespace

3
src/video_core/renderer_opengl/renderer_opengl.cpp
View File

@ -440,7 +440,8 @@ static void DebugHandler(GLenum source, GLenum type, GLuint id, GLenum severity,
level = Log::Level::Debug;
break;
}
//LOG_GENERIC(Log::Class::Render_OpenGL, level, "%s %s %d: %s", GetSource(source), GetType(type), id, message);
LOG_GENERIC(Log::Class::Render_OpenGL, level, "%s %s %d: %s",
GetSource(source), GetType(type), id, message);
}
/// Initialize the renderer