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Audio Core (#2)

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* DSP: Implement Pipe 2

Pipe 2 is a DSP pipe that is used to initialize both the DSP hardware (the
application signals to the DSP to initialize) and the application (the DSP
provides the memory location of structures in the shared memory region).

* AudioCore: Implement codecs (DecodeADPCM, DecodePCM8, DecodePCM16)

* DSP Pipes: Implement as FIFO

* AudioCore: File structure

* AudioCore: More structure

* AudioCore: Buffer management

* DSP/Source: Reorganise Source's AdvanceFrame.

* Audio Output

* lolidk

* huh?

* interp

* More interp stuff

* oops

* Zero State

* Don't mix Source frame if it's not enabled

* DSP: Forgot to zero a buffer, adjusted thread synchronisation, adjusted format spec for buffers

* asdf

* Get it to compile and tweak stretching a bit.

* revert stretch test

* deleted accidental partial catch submodule commit

* new audio stretching algorithm

* update .gitmodule

* fix OS X build

* remove getopt from rubberband

* #include <stddef> to audio_core.h

* typo

* -framework Accelerate

* OptionTransientsSmooth -> OptionTransientsCrisp

* tweak stretch tempo smoothing coefficient. also switch back to smooth.

* tweak mroe

* remove printf

* sola

* #include <cmath>

* VERY QUICK MERGE TO GET IT WORKING DOESN'T ACTIVATE AUDIO FILTERS

* Reminder to self

* fix comparison

* common/thread: Correct code style

* Thread: Make Barrier reusable

* fix threading synchonisation code

* add profiling code

* print error to console when audio clips

* fix metallic sound

* reduce logspam
This commit is contained in:
Dragios 2016-04-16 01:10:29 +08:00
parent 7805f66784
commit a8d0c51c69
65 changed files with 8679 additions and 51 deletions
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3
.gitmodules vendored
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@ -7,3 +7,6 @@
[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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@ -148,6 +148,7 @@ 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")
@ -240,11 +241,15 @@ 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 d81b9269900ae183d0dc98403eea4c971590a807
Subproject commit 2dcb9d979665b6aabb1635c617973e02914e60ec

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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 Submodule

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

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////////////////////////////////////////////////////////////////////////////////
///
/// 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();
}

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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.
///
/// 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

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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.
/// - 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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////////////////////////////////////////////////////////////////////////////////
///
/// 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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////////////////////////////////////////////////////////////////////////////////
///
/// 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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////////////////////////////////////////////////////////////////////////////////
///
/// 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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////////////////////////////////////////////////////////////////////////////////
///
/// 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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////////////////////////////////////////////////////////////////////////////////
///
/// 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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//////////////////////////////////////////////////////////////////////////////
///
/// 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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////////////////////////////////////////////////////////////////////////////////
///
/// 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

372
externals/soundtouch/sse_optimized.cpp vendored Normal file
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@ -0,0 +1,372 @@
////////////////////////////////////////////////////////////////////////////////
///
/// 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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@ -2,8 +2,14 @@ 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
@ -11,11 +17,31 @@ 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})
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()

1
src/audio_core/audio_core.cpp
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@ -4,6 +4,7 @@
#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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@ -4,15 +4,17 @@
#pragma once
#include <cstddef>
namespace Kernel {
class VMManager;
}
namespace AudioCore {
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
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
/// Initialise Audio Core
void Init();

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

0
src/audio_core/hle/common.cpp Normal file
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107
src/audio_core/hle/dsp.cpp
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@ -2,8 +2,23 @@
// 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 {
@ -11,14 +26,104 @@ 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() {
DSP::HLE::ResetPipes();
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));
}
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 previous_buffer_id_dirty; ///< Non-zero when previous_buffer_id changes
u8 current_buffer_id_dirty; ///< Non-zero when current_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 previous_buffer_id; ///< Updated when a buffer finishes playing
u16_le current_buffer_id; ///< Updated when a buffer finishes playing
INSERT_PADDING_DSPWORDS(1);
};
@ -330,6 +330,13 @@ 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;
@ -337,6 +344,7 @@ 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;
@ -344,12 +352,16 @@ 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];
INSERT_PADDING_DSPWORDS(3);
u16 unknown17;
INSERT_PADDING_DSPWORDS(2);
enum class OutputFormat : u16_le {
Mono = 0,
@ -361,7 +373,10 @@ 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(4); ///< TODO: Surround sound related
INSERT_PADDING_DSPWORDS(1); ///< TODO: Surround sound related
u16 unknown16;
u16 unknown15;
u16 unknown18;
INSERT_PADDING_DSPWORDS(2); ///< TODO: Intermediate mixer 1/2 related
u16_le mixer1_enabled;
u16_le mixer2_enabled;
@ -479,25 +494,29 @@ struct SharedMemory {
AdpcmCoefficients adpcm_coefficients;
struct {
INSERT_PADDING_DSPWORDS(0x100);
u16 unknown[256];
} unknown10;
struct {
INSERT_PADDING_DSPWORDS(0xC0);
u16 unknown[192];
} unknown11;
struct {
INSERT_PADDING_DSPWORDS(0x180);
u16 unknown[384];
} unknown12;
struct {
INSERT_PADDING_DSPWORDS(0xA);
// biq
u32_dsp unknown[5];
} unknown13;
struct {
INSERT_PADDING_DSPWORDS(0x13A3);
// biq
u32_dsp unknown[5];
} unknown14;
INSERT_PADDING_DSPWORDS(0x1399);
u16_le frame_counter;
};
ASSERT_DSP_STRUCT(SharedMemory, 0x8000);

44
src/audio_core/hle/effects.cpp Normal file
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@ -0,0 +1,44 @@
// 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
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@ -0,0 +1,20 @@
// 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);
}
}

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// 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;
}
}
}

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// 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>, static_cast<size_t>(DspPipe::DspPipe_MAX)> pipe_data;
static std::array<std::vector<u8>, DspPipe_MAX> pipe_data;
void ResetPipes() {
for (auto& data : pipe_data) {

2
src/audio_core/hle/pipe.h
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@ -23,6 +23,8 @@ 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

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// 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];
}
}
}
}
}

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// 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);
}
}

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// 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);
}
}

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// 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 Normal file
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@ -0,0 +1,3 @@
// 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 Normal file
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@ -0,0 +1,9 @@
// 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 Normal file
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@ -0,0 +1,118 @@
// 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 Normal file
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@ -0,0 +1,46 @@
// 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
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@ -4,6 +4,7 @@
#pragma once
#include <memory>
#include <vector>
#include "common/common_types.h"
@ -31,4 +32,6 @@ public:
virtual std::size_t SamplesInQueue() const = 0;
};
extern std::unique_ptr<Sink> sink;
} // namespace

83
src/audio_core/time_stretch.cpp Normal file
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@ -0,0 +1,83 @@
// 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 Normal file
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@ -0,0 +1,24 @@
// 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
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@ -117,6 +117,8 @@ 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
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@ -65,6 +65,7 @@ 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
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@ -80,6 +80,7 @@ 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

126
src/core/hle/service/dsp_dsp.cpp
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@ -20,29 +20,55 @@ namespace DSP_DSP {
static u32 read_pipe_count;
static Kernel::SharedPtr<Kernel::Event> semaphore_event;
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);
}
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 (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;
/// 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();
}
return number;
}
// DSP Interrupts:
// Interrupt #2 occurs every frame tick. Userland programs normally have a thread that's waiting
// Interrupt (2, 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& interrupt_event : interrupt_events)
interrupt_event.second->Signal();
for (auto& events : interrupt_events)
for (auto& event : events)
event.second->Signal();
}
void SignalInterrupt(u32 interrupt, u32 channel) {
interrupt_events[std::make_pair(interrupt, channel)]->Signal();
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;
}
}
/**
@ -58,6 +84,7 @@ 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);
@ -113,9 +140,11 @@ 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");
}
@ -157,23 +186,47 @@ static void FlushDataCache(Service::Interface* self) {
static void RegisterInterruptEvents(Service::Interface* self) {
u32* cmd_buff = Kernel::GetCommandBuffer();
u32 interrupt = cmd_buff[1];
u32 type_num = 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) {
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);
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");
}
} else {
LOG_CRITICAL(Service_DSP, "Invalid event handle! interrupt=%u, channel=%u, event_handle=0x%08X", interrupt, channel, event_handle);
LOG_CRITICAL(Service_DSP, "Invalid event handle! type=%u, channel=%u, event_handle=0x%08X", type, channel, event_handle);
ASSERT(false); // This should really be handled at a IPC translation layer.
}
} else {
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);
interrupt_events[type_num].erase(channel);
LOG_INFO(Service_DSP, "Unregistered type=%u, channel=%u, event_handle=0x%08X", type, channel, event_handle);
}
}
@ -187,8 +240,13 @@ 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");
}
@ -210,7 +268,13 @@ static void WriteProcessPipe(Service::Interface* self) {
u32 size = cmd_buff[2];
u32 buffer = cmd_buff[4];
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);
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(Memory::GetPointer(buffer) != nullptr, "Invalid Buffer: pipe=%u, size=0x%X, buffer=0x%08X", pipe, size, buffer);
std::vector<u8> message(size);
@ -221,6 +285,7 @@ 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);
@ -250,16 +315,14 @@ 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);
Memory::WriteBlock(addr, response.data(), response.size());
std::vector<u8> response = DSP::HLE::PipeRead(pipe, size);
cmd_buff[2] = static_cast<u32>(response.size());
} else {
cmd_buff[2] = 0; // Return no data
}
Memory::WriteBlock(addr, response.data(), response.size());
cmd_buff[2] = static_cast<u32>(response.size());
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]);
}
@ -333,6 +396,7 @@ 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);
@ -350,10 +414,11 @@ 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_WARNING(Service_DSP, "(STUBBED) called");
LOG_TRACE(Service_DSP, "called");
}
/**
@ -376,6 +441,7 @@ 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:
@ -411,6 +477,7 @@ 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
@ -465,7 +532,8 @@ Interface::Interface() {
Interface::~Interface() {
semaphore_event = nullptr;
interrupt_events.clear();
for (auto& events : interrupt_events)
events.clear();
}
} // namespace

3
src/core/hle/service/dsp_dsp.h
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@ -34,4 +34,7 @@ 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
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@ -440,8 +440,7 @@ 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