diff --git a/.travis.yml b/.travis.yml index d161c8d2059..f7a62235a60 100644 --- a/.travis.yml +++ b/.travis.yml @@ -34,6 +34,7 @@ addons: - librubberband-dev - libshout3-dev - libsndfile1-dev + - libsoundtouch-dev - libsqlite3-dev - libtag1-dev - libupower-glib-dev @@ -53,7 +54,7 @@ before_install: # Virtual X, needed for analyzer waveform tests - if [[ "$TRAVIS_OS_NAME" == "linux" ]]; then export DISPLAY=:99.0 ; fi - if [[ "$TRAVIS_OS_NAME" == "linux" ]]; then sh -e /etc/init.d/xvfb start ; fi - - if [[ "$TRAVIS_OS_NAME" == "osx" ]]; then brew install scons portaudio libsndfile libogg libvorbis portmidi taglib libshout protobuf flac ffmpeg qt chromaprint rubberband libmodplug libid3tag libmad mp4v2 faad2 wavpack opusfile lilv lame; fi + - if [[ "$TRAVIS_OS_NAME" == "osx" ]]; then brew install scons portaudio libsndfile libogg libvorbis portmidi taglib libshout protobuf flac ffmpeg qt chromaprint rubberband libmodplug libid3tag libmad mp4v2 faad2 wavpack opusfile lilv lame sound-touch; fi install: #### diff --git a/appveyor.yml b/appveyor.yml index 154be6a2e51..8b688cec82b 100644 --- a/appveyor.yml +++ b/appveyor.yml @@ -50,7 +50,7 @@ for: install: - sudo apt-get update - - sudo apt-get -y install gdb libavformat-dev libchromaprint-dev libfaad-dev libfftw3-dev libflac-dev libid3tag0-dev libmad0-dev libmodplug-dev libmp3lame-dev libmp4v2-dev libopusfile-dev libportmidi-dev libprotobuf-dev libqt5opengl5-dev libqt5sql5-sqlite libqt5svg5-dev librubberband-dev libshout3-dev libsndfile1-dev libsqlite3-dev libtag1-dev libupower-glib-dev libusb-1.0-0-dev libwavpack-dev portaudio19-dev protobuf-compiler qt5-default qtscript5-dev scons vamp-plugin-sdk qtkeychain-dev liblilv-dev + - sudo apt-get -y install gdb libavformat-dev libchromaprint-dev libfaad-dev libfftw3-dev libflac-dev libid3tag0-dev libmad0-dev libmodplug-dev libmp3lame-dev libmp4v2-dev libopusfile-dev libportmidi-dev libprotobuf-dev libqt5opengl5-dev libqt5sql5-sqlite libqt5svg5-dev librubberband-dev libshout3-dev libsndfile1-dev libsqlite3-dev libtag1-dev libupower-glib-dev libusb-1.0-0-dev libwavpack-dev portaudio19-dev protobuf-compiler qt5-default qtscript5-dev scons vamp-plugin-sdk qtkeychain-dev liblilv-dev libsoundtouch-dev build_script: - scons -j4 test=1 mad=1 faad=1 ffmpeg=1 opus=1 modplug=1 wv=1 hss1394=0 virtualize=0 debug_assertions_fatal=1 verbose=0 localecompare=1 diff --git a/build/depends.py b/build/depends.py index 5b6f702074b..ad78d0b6ba1 100644 --- a/build/depends.py +++ b/build/depends.py @@ -487,43 +487,17 @@ def configure(self, build, conf, env=None): class SoundTouch(Dependence): - SOUNDTOUCH_INTERNAL_PATH = 'lib/soundtouch' - INTERNAL_LINK = True - def sources(self, build): - if self.INTERNAL_LINK: - env = build.env.Clone() - soundtouch_dir = env.Dir(self.SOUNDTOUCH_INTERNAL_PATH) - SCons.Export('env') - SCons.Export('build') - env.SConscript(env.File('SConscript', soundtouch_dir)) - - build.env.Append(LIBPATH=self.SOUNDTOUCH_INTERNAL_PATH) - build.env.Append(LIBS=['soundtouch']) return ['src/engine/enginebufferscalest.cpp'] def configure(self, build, conf, env=None): if env is None: env = build.env + if not conf.CheckForPKG('soundtouch', '2.0.0'): + raise Exception( + "Could not find libsoundtouch >= 2.0.0 or its development headers.") + build.env.ParseConfig('pkg-config soundtouch --silence-errors --cflags --libs') - if build.platform_is_linux: - # Try using system lib - if conf.CheckForPKG('soundtouch', '2.0.0'): - # System Lib found - build.env.ParseConfig('pkg-config soundtouch --silence-errors --cflags --libs') - self.INTERNAL_LINK = False - - if self.INTERNAL_LINK: - env.Append(CPPPATH=['#' + self.SOUNDTOUCH_INTERNAL_PATH]) - - # Prevents circular import. - from .features import Optimize - - # If we do not want optimizations then disable them. - optimize = (build.flags['optimize'] if 'optimize' in build.flags - else Optimize.get_optimization_level(build)) - if optimize == Optimize.LEVEL_OFF: - env.Append(CPPDEFINES='SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS') class RubberBand(Dependence): def sources(self, build): diff --git a/lib/soundtouch-2.1.0.tar.bz2 b/lib/soundtouch-2.1.0.tar.bz2 deleted file mode 100644 index cfca78a0be5..00000000000 Binary files a/lib/soundtouch-2.1.0.tar.bz2 and /dev/null differ diff --git a/lib/soundtouch/AAFilter.cpp b/lib/soundtouch/AAFilter.cpp deleted file mode 100644 index 76a3da65d40..00000000000 --- a/lib/soundtouch/AAFilter.cpp +++ /dev/null @@ -1,222 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// FIR low-pass (anti-alias) filter with filter coefficient design routine and -/// MMX optimization. -/// -/// Anti-alias filter is used to prevent folding of high frequencies when -/// transposing the sample rate with interpolation. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#include "AAFilter.h" -#include "FIRFilter.h" - -using namespace soundtouch; - -#define PI 3.14159265358979323846 -#define TWOPI (2 * PI) - -// define this to save AA filter coefficients to a file -// #define _DEBUG_SAVE_AAFILTER_COEFFICIENTS 1 - -#ifdef _DEBUG_SAVE_AAFILTER_COEFFICIENTS - #include - - 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; - // scale & round to nearest integer - temp += (temp >= 0) ? 0.5 : -0.5; - // ensure no overfloods - assert(temp >= -32768 && temp <= 32767); - coeffs[i] = (SAMPLETYPE)temp; - } - - // Set coefficients. Use divide factor 14 => divide result by 2^14 = 16384 - pFIR->setCoefficients(coeffs, length, 14); - - _DEBUG_SAVE_AAFIR_COEFFS(coeffs, length); - - delete[] work; - delete[] coeffs; -} - - -// Applies the filter to the given sequence of samples. -// Note : The amount of outputted samples is by value of 'filter length' -// smaller than the amount of input samples. -uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples, uint numChannels) const -{ - return pFIR->evaluate(dest, src, numSamples, numChannels); -} - - -/// Applies the filter to the given src & dest pipes, so that processed amount of -/// samples get removed from src, and produced amount added to dest -/// Note : The amount of outputted samples is by value of 'filter length' -/// smaller than the amount of input samples. -uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const -{ - SAMPLETYPE *pdest; - const SAMPLETYPE *psrc; - uint numSrcSamples; - uint result; - int numChannels = src.getChannels(); - - assert(numChannels == dest.getChannels()); - - numSrcSamples = src.numSamples(); - psrc = src.ptrBegin(); - pdest = dest.ptrEnd(numSrcSamples); - result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels); - src.receiveSamples(result); - dest.putSamples(result); - - return result; -} - - -uint AAFilter::getLength() const -{ - return pFIR->getLength(); -} diff --git a/lib/soundtouch/AAFilter.h b/lib/soundtouch/AAFilter.h deleted file mode 100644 index 8e5697f7966..00000000000 --- a/lib/soundtouch/AAFilter.h +++ /dev/null @@ -1,93 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo -/// while maintaining the original pitch by using a time domain WSOLA-like method -/// with several performance-increasing tweaks. -/// -/// Anti-alias filter is used to prevent folding of high frequencies when -/// transposing the sample rate with interpolation. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef AAFilter_H -#define AAFilter_H - -#include "STTypes.h" -#include "FIFOSampleBuffer.h" - -namespace soundtouch -{ - -class AAFilter -{ -protected: - class FIRFilter *pFIR; - - /// Low-pass filter cut-off frequency, negative = invalid - double cutoffFreq; - - /// num of filter taps - uint length; - - /// Calculate the FIR coefficients realizing the given cutoff-frequency - void calculateCoeffs(); -public: - AAFilter(uint length); - - ~AAFilter(); - - /// Sets new anti-alias filter cut-off edge frequency, scaled to sampling - /// frequency (nyquist frequency = 0.5). The filter will cut off the - /// frequencies than that. - void setCutoffFreq(double newCutoffFreq); - - /// Sets number of FIR filter taps, i.e. ~filter complexity - void setLength(uint newLength); - - uint getLength() const; - - /// Applies the filter to the given sequence of samples. - /// Note : The amount of outputted samples is by value of 'filter length' - /// smaller than the amount of input samples. - uint evaluate(SAMPLETYPE *dest, - const SAMPLETYPE *src, - uint numSamples, - uint numChannels) const; - - /// Applies the filter to the given src & dest pipes, so that processed amount of - /// samples get removed from src, and produced amount added to dest - /// Note : The amount of outputted samples is by value of 'filter length' - /// smaller than the amount of input samples. - uint evaluate(FIFOSampleBuffer &dest, - FIFOSampleBuffer &src) const; - -}; - -} - -#endif diff --git a/lib/soundtouch/BPMDetect.cpp b/lib/soundtouch/BPMDetect.cpp deleted file mode 100644 index 01202f0e807..00000000000 --- a/lib/soundtouch/BPMDetect.cpp +++ /dev/null @@ -1,572 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Beats-per-minute (BPM) detection routine. -/// -/// The beat detection algorithm works as follows: -/// - Use function 'inputSamples' to input a chunks of samples to the class for -/// analysis. It's a good idea to enter a large sound file or stream in smallish -/// chunks of around few kilosamples in order not to extinguish too much RAM memory. -/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden, -/// which is basically ok as low (bass) frequencies mostly determine the beat rate. -/// Simple averaging is used for anti-alias filtering because the resulting signal -/// quality isn't of that high importance. -/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by -/// taking absolute value that's smoothed by sliding average. Signal levels that -/// are below a couple of times the general RMS amplitude level are cut away to -/// leave only notable peaks there. -/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term -/// autocorrelation function of the enveloped signal. -/// - After whole sound data file has been analyzed as above, the bpm level is -/// detected by function 'getBpm' that finds the highest peak of the autocorrelation -/// function, calculates it's precise location and converts this reading to bpm's. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#define _USE_MATH_DEFINES - -#include -#include -#include -#include -#include -#include "FIFOSampleBuffer.h" -#include "PeakFinder.h" -#include "BPMDetect.h" - -using namespace soundtouch; - -// algorithm input sample block size -static const int INPUT_BLOCK_SIZE = 2048; - -// decimated sample block size -static const int DECIMATED_BLOCK_SIZE = 256; - -/// Target sample rate after decimation -static const int TARGET_SRATE = 1000; - -/// XCorr update sequence size, update in about 200msec chunks -static const int XCORR_UPDATE_SEQUENCE = (int)(TARGET_SRATE / 5); - -/// Moving average N size -static const int MOVING_AVERAGE_N = 15; - -/// XCorr decay time constant, decay to half in 30 seconds -/// If it's desired to have the system adapt quicker to beat rate -/// changes within a continuing music stream, then the -/// 'xcorr_decay_time_constant' value can be reduced, yet that -/// can increase possibility of glitches in bpm detection. -static const double XCORR_DECAY_TIME_CONSTANT = 30.0; - -/// Data overlap factor for beat detection algorithm -static const int OVERLAP_FACTOR = 4; - -static const double TWOPI = (2 * M_PI); - -//////////////////////////////////////////////////////////////////////////////// - -// Enable following define to create bpm analysis file: - -//#define _CREATE_BPM_DEBUG_FILE - -#ifdef _CREATE_BPM_DEBUG_FILE - - static void _SaveDebugData(const char *name, const float *data, int minpos, int maxpos, double coeff) - { - FILE *fptr = fopen(name, "wt"); - int i; - - if (fptr) - { - printf("\nWriting BPM debug data into file %s\n", name); - for (i = minpos; i < maxpos; i ++) - { - fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]); - } - fclose(fptr); - } - } - - void _SaveDebugBeatPos(const char *name, const std::vector &beats) - { - printf("\nWriting beat detections data into file %s\n", name); - - FILE *fptr = fopen(name, "wt"); - if (fptr) - { - for (uint i = 0; i < beats.size(); i++) - { - BEAT b = beats[i]; - fprintf(fptr, "%lf\t%lf\n", b.pos, b.strength); - } - fclose(fptr); - } - } -#else - #define _SaveDebugData(name, a,b,c,d) - #define _SaveDebugBeatPos(name, b) -#endif - -// Hamming window -void hamming(float *w, int N) -{ - for (int i = 0; i < N; i++) - { - w[i] = (float)(0.54 - 0.46 * cos(TWOPI * i / (N - 1))); - } - -} - -//////////////////////////////////////////////////////////////////////////////// -// -// IIR2_filter - 2nd order IIR filter - -IIR2_filter::IIR2_filter(const double *lpf_coeffs) -{ - memcpy(coeffs, lpf_coeffs, 5 * sizeof(double)); - memset(prev, 0, sizeof(prev)); -} - - -float IIR2_filter::update(float x) -{ - prev[0] = x; - double y = x * coeffs[0]; - - for (int i = 4; i >= 1; i--) - { - y += coeffs[i] * prev[i]; - prev[i] = prev[i - 1]; - } - - prev[3] = y; - return (float)y; -} - - -// IIR low-pass filter coefficients, calculated with matlab/octave cheby2(2,40,0.05) -const double _LPF_coeffs[5] = { 0.00996655391939, -0.01944529148401, 0.00996655391939, 1.96867605796247, -0.96916387431724 }; - -//////////////////////////////////////////////////////////////////////////////// - -BPMDetect::BPMDetect(int numChannels, int aSampleRate) : - beat_lpf(_LPF_coeffs) -{ - beats.reserve(250); // initial reservation to prevent frequent reallocation - - this->sampleRate = aSampleRate; - this->channels = numChannels; - - decimateSum = 0; - decimateCount = 0; - - // choose decimation factor so that result is approx. 1000 Hz - decimateBy = sampleRate / TARGET_SRATE; - assert(decimateBy > 0); - assert(INPUT_BLOCK_SIZE < decimateBy * DECIMATED_BLOCK_SIZE); - - // Calculate window length & starting item according to desired min & max bpms - windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM); - windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE); - - assert(windowLen > windowStart); - - // allocate new working objects - xcorr = new float[windowLen]; - memset(xcorr, 0, windowLen * sizeof(float)); - - pos = 0; - peakPos = 0; - peakVal = 0; - init_scaler = 1; - beatcorr_ringbuffpos = 0; - beatcorr_ringbuff = new float[windowLen]; - memset(beatcorr_ringbuff, 0, windowLen * sizeof(float)); - - // allocate processing buffer - buffer = new FIFOSampleBuffer(); - // we do processing in mono mode - buffer->setChannels(1); - buffer->clear(); - - // calculate hamming windows - hamw = new float[XCORR_UPDATE_SEQUENCE]; - hamming(hamw, XCORR_UPDATE_SEQUENCE); - hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2]; - hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2); -} - - -BPMDetect::~BPMDetect() -{ - delete[] xcorr; - delete[] beatcorr_ringbuff; - delete[] hamw; - delete[] hamw2; - delete buffer; -} - - -/// convert to mono, low-pass filter & decimate to about 500 Hz. -/// return number of outputted samples. -/// -/// Decimation is used to remove the unnecessary frequencies and thus to reduce -/// the amount of data needed to be processed as calculating autocorrelation -/// function is a very-very heavy operation. -/// -/// Anti-alias filtering is done simply by averaging the samples. This is really a -/// poor-man's anti-alias filtering, but it's not so critical in this kind of application -/// (it'd also be difficult to design a high-quality filter with steep cut-off at very -/// narrow band) -int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples) -{ - int count, outcount; - LONG_SAMPLETYPE out; - - assert(channels > 0); - assert(decimateBy > 0); - outcount = 0; - for (count = 0; count < numsamples; count ++) - { - int j; - - // convert to mono and accumulate - for (j = 0; j < channels; j ++) - { - decimateSum += src[j]; - } - src += j; - - decimateCount ++; - if (decimateCount >= decimateBy) - { - // Store every Nth sample only - out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels)); - decimateSum = 0; - decimateCount = 0; -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - // check ranges for sure (shouldn't actually be necessary) - if (out > 32767) - { - out = 32767; - } - else if (out < -32768) - { - out = -32768; - } -#endif // SOUNDTOUCH_INTEGER_SAMPLES - dest[outcount] = (SAMPLETYPE)out; - outcount ++; - } - } - return outcount; -} - - -// Calculates autocorrelation function of the sample history buffer -void BPMDetect::updateXCorr(int process_samples) -{ - int offs; - SAMPLETYPE *pBuffer; - - assert(buffer->numSamples() >= (uint)(process_samples + windowLen)); - assert(process_samples == XCORR_UPDATE_SEQUENCE); - - pBuffer = buffer->ptrBegin(); - - // calculate decay factor for xcorr filtering - float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples)); - - // prescale pbuffer - float tmp[XCORR_UPDATE_SEQUENCE]; - for (int i = 0; i < process_samples; i++) - { - tmp[i] = hamw[i] * hamw[i] * pBuffer[i]; - } - - #pragma omp parallel for - for (offs = windowStart; offs < windowLen; offs ++) - { - double sum; - int i; - - sum = 0; - for (i = 0; i < process_samples; i ++) - { - sum += tmp[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary - } - xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable time constant. - - xcorr[offs] += (float)fabs(sum); - } -} - - -// Detect individual beat positions -void BPMDetect::updateBeatPos(int process_samples) -{ - SAMPLETYPE *pBuffer; - - assert(buffer->numSamples() >= (uint)(process_samples + windowLen)); - - pBuffer = buffer->ptrBegin(); - assert(process_samples == XCORR_UPDATE_SEQUENCE / 2); - - // static double thr = 0.0003; - double posScale = (double)this->decimateBy / (double)this->sampleRate; - int resetDur = (int)(0.12 / posScale + 0.5); - double corrScale = 1.0 / (double)(windowLen - windowStart); - - // prescale pbuffer - float tmp[XCORR_UPDATE_SEQUENCE / 2]; - for (int i = 0; i < process_samples; i++) - { - tmp[i] = hamw2[i] * hamw2[i] * pBuffer[i]; - } - - #pragma omp parallel for - for (int offs = windowStart; offs < windowLen; offs++) - { - double sum = 0; - for (int i = 0; i < process_samples; i++) - { - sum += tmp[i] * pBuffer[offs + i]; - } - beatcorr_ringbuff[(beatcorr_ringbuffpos + offs) % windowLen] += (float)((sum > 0) ? sum : 0); // accumulate only positive correlations - } - - int skipstep = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR; - - // compensate empty buffer at beginning by scaling coefficient - float scale = (float)windowLen / (float)(skipstep * init_scaler); - if (scale > 1.0f) - { - init_scaler++; - } - else - { - scale = 1.0f; - } - - // detect beats - for (int i = 0; i < skipstep; i++) - { - LONG_SAMPLETYPE max = 0; - - float sum = beatcorr_ringbuff[beatcorr_ringbuffpos]; - sum -= beat_lpf.update(sum); - - if (sum > peakVal) - { - // found new local largest value - peakVal = sum; - peakPos = pos; - } - if (pos > peakPos + resetDur) - { - // largest value not updated for 200msec => accept as beat - peakPos += skipstep; - if (peakVal > 0) - { - // add detected beat to end of "beats" vector - BEAT temp = { (float)(peakPos * posScale), (float)(peakVal * scale) }; - beats.push_back(temp); - } - - peakVal = 0; - peakPos = pos; - } - - beatcorr_ringbuff[beatcorr_ringbuffpos] = 0; - pos++; - beatcorr_ringbuffpos = (beatcorr_ringbuffpos + 1) % windowLen; - } -} - - -#define max(x,y) ((x) > (y) ? (x) : (y)) - -void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples) -{ - SAMPLETYPE decimated[DECIMATED_BLOCK_SIZE]; - - // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration - while (numSamples > 0) - { - int block; - int decSamples; - - block = (numSamples > INPUT_BLOCK_SIZE) ? INPUT_BLOCK_SIZE : numSamples; - - // decimate. note that converts to mono at the same time - decSamples = decimate(decimated, samples, block); - samples += block * channels; - numSamples -= block; - - buffer->putSamples(decimated, decSamples); - } - - // when the buffer has enough samples for processing... - int req = max(windowLen + XCORR_UPDATE_SEQUENCE, 2 * XCORR_UPDATE_SEQUENCE); - while ((int)buffer->numSamples() >= req) - { - // ... update autocorrelations... - updateXCorr(XCORR_UPDATE_SEQUENCE); - // ...update beat position calculation... - updateBeatPos(XCORR_UPDATE_SEQUENCE / 2); - // ... and remove proceessed samples from the buffer - int n = XCORR_UPDATE_SEQUENCE / OVERLAP_FACTOR; - buffer->receiveSamples(n); - } -} - - -void BPMDetect::removeBias() -{ - int i; - - // Remove linear bias: calculate linear regression coefficient - // 1. calc mean of 'xcorr' and 'i' - double mean_i = 0; - double mean_x = 0; - for (i = windowStart; i < windowLen; i++) - { - mean_x += xcorr[i]; - } - mean_x /= (windowLen - windowStart); - mean_i = 0.5 * (windowLen - 1 + windowStart); - - // 2. calculate linear regression coefficient - double b = 0; - double div = 0; - for (i = windowStart; i < windowLen; i++) - { - double xt = xcorr[i] - mean_x; - double xi = i - mean_i; - b += xt * xi; - div += xi * xi; - } - b /= div; - - // subtract linear regression and resolve min. value bias - float minval = FLT_MAX; // arbitrary large number - for (i = windowStart; i < windowLen; i ++) - { - xcorr[i] -= (float)(b * i); - if (xcorr[i] < minval) - { - minval = xcorr[i]; - } - } - - // subtract min.value - for (i = windowStart; i < windowLen; i ++) - { - xcorr[i] -= minval; - } -} - - -// Calculate N-point moving average for "source" values -void MAFilter(float *dest, const float *source, int start, int end, int N) -{ - for (int i = start; i < end; i++) - { - int i1 = i - N / 2; - int i2 = i + N / 2 + 1; - if (i1 < start) i1 = start; - if (i2 > end) i2 = end; - - double sum = 0; - for (int j = i1; j < i2; j ++) - { - sum += source[j]; - } - dest[i] = (float)(sum / (i2 - i1)); - } -} - - -float BPMDetect::getBpm() -{ - double peakPos; - double coeff; - PeakFinder peakFinder; - - // remove bias from xcorr data - removeBias(); - - coeff = 60.0 * ((double)sampleRate / (double)decimateBy); - - // save bpm debug data if debug data writing enabled - _SaveDebugData("soundtouch-bpm-xcorr.txt", xcorr, windowStart, windowLen, coeff); - - // Smoothen by N-point moving-average - float *data = new float[windowLen]; - memset(data, 0, sizeof(float) * windowLen); - MAFilter(data, xcorr, windowStart, windowLen, MOVING_AVERAGE_N); - - // find peak position - peakPos = peakFinder.detectPeak(data, windowStart, windowLen); - - // save bpm debug data if debug data writing enabled - _SaveDebugData("soundtouch-bpm-smoothed.txt", data, windowStart, windowLen, coeff); - - delete[] data; - - assert(decimateBy != 0); - if (peakPos < 1e-9) return 0.0; // detection failed. - - _SaveDebugBeatPos("soundtouch-detected-beats.txt", beats); - - // calculate BPM - float bpm = (float)(coeff / peakPos); - return (bpm >= MIN_BPM && bpm <= MAX_BPM_VALID) ? bpm : 0; -} - - -/// Get beat position arrays. Note: The array includes also really low beat detection values -/// in absence of clear strong beats. Consumer may wish to filter low values away. -/// - "pos" receive array of beat positions -/// - "values" receive array of beat detection strengths -/// - max_num indicates max.size of "pos" and "values" array. -/// -/// You can query a suitable array sized by calling this with NULL in "pos" & "values". -/// -/// \return number of beats in the arrays. -int BPMDetect::getBeats(float *pos, float *values, int max_num) -{ - int num = beats.size(); - if ((!pos) || (!values)) return num; // pos or values NULL, return just size - - for (int i = 0; (i < num) && (i < max_num); i++) - { - pos[i] = beats[i].pos; - values[i] = beats[i].strength; - } - return num; -} diff --git a/lib/soundtouch/BPMDetect.h b/lib/soundtouch/BPMDetect.h deleted file mode 100644 index 8ece78448db..00000000000 --- a/lib/soundtouch/BPMDetect.h +++ /dev/null @@ -1,205 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Beats-per-minute (BPM) detection routine. -/// -/// The beat detection algorithm works as follows: -/// - Use function 'inputSamples' to input a chunks of samples to the class for -/// analysis. It's a good idea to enter a large sound file or stream in smallish -/// chunks of around few kilosamples in order not to extinguish too much RAM memory. -/// - Input sound data is decimated to approx 500 Hz to reduce calculation burden, -/// which is basically ok as low (bass) frequencies mostly determine the beat rate. -/// Simple averaging is used for anti-alias filtering because the resulting signal -/// quality isn't of that high importance. -/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by -/// taking absolute value that's smoothed by sliding average. Signal levels that -/// are below a couple of times the general RMS amplitude level are cut away to -/// leave only notable peaks there. -/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term -/// autocorrelation function of the enveloped signal. -/// - After whole sound data file has been analyzed as above, the bpm level is -/// detected by function 'getBpm' that finds the highest peak of the autocorrelation -/// function, calculates it's precise location and converts this reading to bpm's. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _BPMDetect_H_ -#define _BPMDetect_H_ - -#include -#include "STTypes.h" -#include "FIFOSampleBuffer.h" - -namespace soundtouch -{ - - /// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit. - #define MIN_BPM 45 - - /// Maximum allowed BPM rate range. Used for calculating algorithm parametrs - #define MAX_BPM_RANGE 200 - - /// Maximum allowed BPM rate range. Used to restrict accepted result below a reasonable limit. - #define MAX_BPM_VALID 190 - -//////////////////////////////////////////////////////////////////////////////// - - typedef struct - { - float pos; - float strength; - } BEAT; - - - class IIR2_filter - { - double coeffs[5]; - double prev[5]; - - public: - IIR2_filter(const double *lpf_coeffs); - float update(float x); - }; - - - /// Class for calculating BPM rate for audio data. - class BPMDetect - { - protected: - /// Auto-correlation accumulator bins. - float *xcorr; - - /// Sample average counter. - int decimateCount; - - /// Sample average accumulator for FIFO-like decimation. - soundtouch::LONG_SAMPLETYPE decimateSum; - - /// Decimate sound by this coefficient to reach approx. 500 Hz. - int decimateBy; - - /// Auto-correlation window length - int windowLen; - - /// Number of channels (1 = mono, 2 = stereo) - int channels; - - /// sample rate - int sampleRate; - - /// Beginning of auto-correlation window: Autocorrelation isn't being updated for - /// the first these many correlation bins. - int windowStart; - - /// window functions for data preconditioning - float *hamw; - float *hamw2; - - // beat detection variables - int pos; - int peakPos; - int beatcorr_ringbuffpos; - int init_scaler; - float peakVal; - float *beatcorr_ringbuff; - - /// FIFO-buffer for decimated processing samples. - soundtouch::FIFOSampleBuffer *buffer; - - /// Collection of detected beat positions - //BeatCollection beats; - std::vector beats; - - // 2nd order low-pass-filter - IIR2_filter beat_lpf; - - /// Updates auto-correlation function for given number of decimated samples that - /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe - /// though). - void updateXCorr(int process_samples /// How many samples are processed. - ); - - /// Decimates samples to approx. 500 Hz. - /// - /// \return Number of output samples. - int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer - const soundtouch::SAMPLETYPE *src, ///< Source sample buffer - int numsamples ///< Number of source samples. - ); - - /// Calculates amplitude envelope for the buffer of samples. - /// Result is output to 'samples'. - void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer - int numsamples ///< Number of samples in buffer - ); - - /// remove constant bias from xcorr data - void removeBias(); - - // Detect individual beat positions - void updateBeatPos(int process_samples); - - - public: - /// Constructor. - BPMDetect(int numChannels, ///< Number of channels in sample data. - int sampleRate ///< Sample rate in Hz. - ); - - /// Destructor. - virtual ~BPMDetect(); - - /// Inputs a block of samples for analyzing: Envelopes the samples and then - /// updates the autocorrelation estimation. When whole song data has been input - /// in smaller blocks using this function, read the resulting bpm with 'getBpm' - /// function. - /// - /// Notice that data in 'samples' array can be disrupted in processing. - void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer - int numSamples ///< Number of samples in buffer - ); - - /// Analyzes the results and returns the BPM rate. Use this function to read result - /// after whole song data has been input to the class by consecutive calls of - /// 'inputSamples' function. - /// - /// \return Beats-per-minute rate, or zero if detection failed. - float getBpm(); - - /// Get beat position arrays. Note: The array includes also really low beat detection values - /// in absence of clear strong beats. Consumer may wish to filter low values away. - /// - "pos" receive array of beat positions - /// - "values" receive array of beat detection strengths - /// - max_num indicates max.size of "pos" and "values" array. - /// - /// You can query a suitable array sized by calling this with NULL in "pos" & "values". - /// - /// \return number of beats in the arrays. - int getBeats(float *pos, float *strength, int max_num); - }; -} -#endif // _BPMDetect_H_ diff --git a/lib/soundtouch/COPYING.TXT b/lib/soundtouch/COPYING.TXT deleted file mode 100644 index bbd24e6430f..00000000000 --- a/lib/soundtouch/COPYING.TXT +++ /dev/null @@ -1,458 +0,0 @@ - GNU LESSER GENERAL PUBLIC LICENSE - Version 2.1, February 1999 - - Copyright (C) 1991, 1999 Free Software Foundation, Inc. - 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - Everyone is permitted to copy and distribute verbatim copies - of this license document, but changing it is not allowed. - -[This is the first released version of the Lesser GPL. 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The class automatically removes the -/// outputted samples from the buffer, as well as grows the buffer size -/// whenever necessary. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include - -#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; - - if (!verifyNumberOfChannels(numChannels)) return; - - usedBytes = channels * samplesInBuffer; - channels = (uint)numChannels; - samplesInBuffer = usedBytes / channels; -} - - -// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and -// zeroes this pointer by copying samples from the 'bufferPos' pointer -// location on to the beginning of the buffer. -void FIFOSampleBuffer::rewind() -{ - if (buffer && bufferPos) - { - memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer); - bufferPos = 0; - } -} - - -// Adds 'numSamples' pcs of samples from the 'samples' memory position to -// the sample buffer. -void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples) -{ - memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels); - samplesInBuffer += nSamples; -} - - -// Increases the number of samples in the buffer without copying any actual -// samples. -// -// This function is used to update the number of samples in the sample buffer -// when accessing the buffer directly with 'ptrEnd' function. Please be -// careful though! -void FIFOSampleBuffer::putSamples(uint nSamples) -{ - uint req; - - req = samplesInBuffer + nSamples; - ensureCapacity(req); - samplesInBuffer += nSamples; -} - - -// Returns a pointer to the end of the used part of the sample buffer (i.e. -// where the new samples are to be inserted). This function may be used for -// inserting new samples into the sample buffer directly. Please be careful! -// -// Parameter 'slackCapacity' tells the function how much free capacity (in -// terms of samples) there _at least_ should be, in order to the caller to -// successfully insert all the required samples to the buffer. When necessary, -// the function grows the buffer size to comply with this requirement. -// -// When using this function as means for inserting new samples, also remember -// to increase the sample count afterwards, by calling the -// 'putSamples(numSamples)' function. -SAMPLETYPE *FIFOSampleBuffer::ptrEnd(uint slackCapacity) -{ - ensureCapacity(samplesInBuffer + slackCapacity); - return buffer + samplesInBuffer * channels; -} - - -// Returns a pointer to the beginning of the currently non-outputted samples. -// This function is provided for accessing the output samples directly. -// Please be careful! -// -// When using this function to output samples, also remember to 'remove' the -// outputted samples from the buffer by calling the -// 'receiveSamples(numSamples)' function -SAMPLETYPE *FIFOSampleBuffer::ptrBegin() -{ - assert(buffer); - return buffer + bufferPos * channels; -} - - -// Ensures that the buffer has enough capacity, i.e. space for _at least_ -// 'capacityRequirement' number of samples. The buffer is grown in steps of -// 4 kilobytes to eliminate the need for frequently growing up the buffer, -// as well as to round the buffer size up to the virtual memory page size. -void FIFOSampleBuffer::ensureCapacity(uint capacityRequirement) -{ - SAMPLETYPE *tempUnaligned, *temp; - - if (capacityRequirement > getCapacity()) - { - // enlarge the buffer in 4kbyte steps (round up to next 4k boundary) - sizeInBytes = (capacityRequirement * channels * sizeof(SAMPLETYPE) + 4095) & (uint)-4096; - assert(sizeInBytes % 2 == 0); - tempUnaligned = new SAMPLETYPE[sizeInBytes / sizeof(SAMPLETYPE) + 16 / sizeof(SAMPLETYPE)]; - if (tempUnaligned == 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; -} diff --git a/lib/soundtouch/FIFOSampleBuffer.h b/lib/soundtouch/FIFOSampleBuffer.h deleted file mode 100644 index de298dd82ba..00000000000 --- a/lib/soundtouch/FIFOSampleBuffer.h +++ /dev/null @@ -1,177 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// A buffer class for temporarily storaging sound samples, operates as a -/// first-in-first-out pipe. -/// -/// Samples are added to the end of the sample buffer with the 'putSamples' -/// function, and are received from the beginning of the buffer by calling -/// the 'receiveSamples' function. The class automatically removes the -/// output samples from the buffer as well as grows the storage size -/// whenever necessary. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef FIFOSampleBuffer_H -#define FIFOSampleBuffer_H - -#include "FIFOSamplePipe.h" - -namespace soundtouch -{ - -/// Sample buffer working in FIFO (first-in-first-out) principle. The class takes -/// care of storage size adjustment and data moving during input/output operations. -/// -/// Notice that in case of stereo audio, one sample is considered to consist of -/// both channel data. -class FIFOSampleBuffer : public FIFOSamplePipe -{ -private: - /// Sample buffer. - SAMPLETYPE *buffer; - - // Raw unaligned buffer memory. 'buffer' is made aligned by pointing it to first - // 16-byte aligned location of this buffer - SAMPLETYPE *bufferUnaligned; - - /// Sample buffer size in bytes - uint sizeInBytes; - - /// How many samples are currently in buffer. - uint samplesInBuffer; - - /// Channels, 1=mono, 2=stereo. - uint channels; - - /// Current position pointer to the buffer. This pointer is increased when samples are - /// removed from the pipe so that it's necessary to actually rewind buffer (move data) - /// only new data when is put to the pipe. - uint bufferPos; - - /// Rewind the buffer by moving data from position pointed by 'bufferPos' to real - /// beginning of the buffer. - void rewind(); - - /// Ensures that the buffer has capacity for at least this many samples. - void ensureCapacity(uint capacityRequirement); - - /// Returns current capacity. - uint getCapacity() const; - -public: - - /// Constructor - FIFOSampleBuffer(int numChannels = 2 ///< Number of channels, 1=mono, 2=stereo. - ///< Default is stereo. - ); - - /// destructor - ~FIFOSampleBuffer(); - - /// 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 successfully insert the - ///< desired samples to the buffer. If necessary, the function - ///< grows the buffer size to comply with this requirement. - ); - - /// Adds 'numSamples' pcs of samples from the 'samples' memory position to - /// the sample buffer. - virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples. - uint numSamples ///< Number of samples to insert. - ); - - /// 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 diff --git a/lib/soundtouch/FIFOSamplePipe.h b/lib/soundtouch/FIFOSamplePipe.h deleted file mode 100644 index 38ef31a5c24..00000000000 --- a/lib/soundtouch/FIFOSamplePipe.h +++ /dev/null @@ -1,230 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// 'FIFOSamplePipe' : An abstract base class for classes that manipulate sound -/// samples by operating like a first-in-first-out pipe: New samples are fed -/// into one end of the pipe with the 'putSamples' function, and the processed -/// samples are received from the other end with the 'receiveSamples' function. -/// -/// 'FIFOProcessor' : A base class for classes the do signal processing with -/// the samples while operating like a first-in-first-out pipe. When samples -/// are input with the 'putSamples' function, the class processes them -/// and moves the processed samples to the given 'output' pipe object, which -/// may be either another processing stage, or a fifo sample buffer object. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef FIFOSamplePipe_H -#define FIFOSamplePipe_H - -#include -#include -#include "STTypes.h" - -namespace soundtouch -{ - -/// Abstract base class for FIFO (first-in-first-out) sample processing classes. -class FIFOSamplePipe -{ -protected: - - bool verifyNumberOfChannels(int nChannels) const - { - if ((nChannels > 0) && (nChannels <= SOUNDTOUCH_MAX_CHANNELS)) - { - return true; - } - ST_THROW_RT_ERROR("Error: Illegal number of channels"); - return false; - } - -public: - // virtual default destructor - virtual ~FIFOSamplePipe() {} - - - /// Returns a pointer to the beginning of the output samples. - /// This function is provided for accessing the output samples directly. - /// Please be careful for not to corrupt the book-keeping! - /// - /// When using this function to output samples, also remember to 'remove' the - /// output samples from the buffer by calling the - /// 'receiveSamples(numSamples)' function - virtual SAMPLETYPE *ptrBegin() = 0; - - /// Adds 'numSamples' pcs of samples from the 'samples' memory position to - /// the sample buffer. - virtual void putSamples(const SAMPLETYPE *samples, ///< Pointer to samples. - uint numSamples ///< Number of samples to insert. - ) = 0; - - - // Moves samples from the 'other' pipe instance to this instance. - void moveSamples(FIFOSamplePipe &other ///< Other pipe instance where from the receive the data. - ) - { - int oNumSamples = other.numSamples(); - - putSamples(other.ptrBegin(), oNumSamples); - other.receiveSamples(oNumSamples); - }; - - /// Output samples from beginning of the sample buffer. Copies requested samples to - /// output buffer and removes them from the sample buffer. If there are less than - /// 'numsample' samples in the buffer, returns all that available. - /// - /// \return Number of samples returned. - virtual uint receiveSamples(SAMPLETYPE *output, ///< Buffer where to copy output samples. - uint maxSamples ///< How many samples to receive at max. - ) = 0; - - /// Adjusts book-keeping so that given number of samples are removed from beginning of the - /// sample buffer without copying them anywhere. - /// - /// Used to reduce the number of samples in the buffer when accessing the sample buffer directly - /// with 'ptrBegin' function. - virtual uint receiveSamples(uint maxSamples ///< Remove this many samples from the beginning of pipe. - ) = 0; - - /// Returns number of samples currently available. - virtual uint numSamples() const = 0; - - // Returns nonzero if there aren't any samples available for outputting. - virtual int isEmpty() const = 0; - - /// Clears all the samples. - virtual void clear() = 0; - - /// allow trimming (downwards) amount of samples in pipeline. - /// Returns adjusted amount of samples - virtual uint adjustAmountOfSamples(uint numSamples) = 0; - -}; - - -/// Base-class for sound processing routines working in FIFO principle. With this base -/// class it's easy to implement sound processing stages that can be chained together, -/// so that samples that are fed into beginning of the pipe automatically go through -/// all the processing stages. -/// -/// When samples are input to this class, they're first processed and then put to -/// the FIFO pipe that's defined as output of this class. This output pipe can be -/// either other processing stage or a FIFO sample buffer. -class FIFOProcessor :public FIFOSamplePipe -{ -protected: - /// Internal pipe where processed samples are put. - FIFOSamplePipe *output; - - /// Sets output pipe. - void setOutPipe(FIFOSamplePipe *pOutput) - { - assert(output == 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 diff --git a/lib/soundtouch/FIRFilter.cpp b/lib/soundtouch/FIRFilter.cpp deleted file mode 100644 index 218e50ef5a7..00000000000 --- a/lib/soundtouch/FIRFilter.cpp +++ /dev/null @@ -1,324 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// General FIR digital filter routines with MMX optimization. -/// -/// Notes : MMX optimized functions reside in a separate, platform-specific file, -/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp' -/// -/// This source file contains OpenMP optimizations that allow speeding up the -/// corss-correlation algorithm by executing it in several threads / CPU cores -/// in parallel. See the following article link for more detailed discussion -/// about SoundTouch OpenMP optimizations: -/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#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; - } -} diff --git a/lib/soundtouch/FIRFilter.h b/lib/soundtouch/FIRFilter.h deleted file mode 100644 index 297b0f81ecc..00000000000 --- a/lib/soundtouch/FIRFilter.h +++ /dev/null @@ -1,139 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// General FIR digital filter routines with MMX optimization. -/// -/// Note : MMX optimized functions reside in a separate, platform-specific file, -/// e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp' -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef FIRFilter_H -#define FIRFilter_H - -#include -#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 diff --git a/lib/soundtouch/InterpolateCubic.cpp b/lib/soundtouch/InterpolateCubic.cpp deleted file mode 100644 index fe49684817f..00000000000 --- a/lib/soundtouch/InterpolateCubic.cpp +++ /dev/null @@ -1,196 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Cubic interpolation routine. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#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; -} diff --git a/lib/soundtouch/InterpolateCubic.h b/lib/soundtouch/InterpolateCubic.h deleted file mode 100644 index 457821033f9..00000000000 --- a/lib/soundtouch/InterpolateCubic.h +++ /dev/null @@ -1,63 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Cubic interpolation routine. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _InterpolateCubic_H_ -#define _InterpolateCubic_H_ - -#include "RateTransposer.h" -#include "STTypes.h" - -namespace soundtouch -{ - -class InterpolateCubic : public TransposerBase -{ -protected: - virtual 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 diff --git a/lib/soundtouch/InterpolateLinear.cpp b/lib/soundtouch/InterpolateLinear.cpp deleted file mode 100644 index c3aa1994cfa..00000000000 --- a/lib/soundtouch/InterpolateLinear.cpp +++ /dev/null @@ -1,296 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Linear interpolation algorithm. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#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; -} diff --git a/lib/soundtouch/InterpolateLinear.h b/lib/soundtouch/InterpolateLinear.h deleted file mode 100644 index faa2e2c5a58..00000000000 --- a/lib/soundtouch/InterpolateLinear.h +++ /dev/null @@ -1,88 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Linear interpolation routine. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _InterpolateLinear_H_ -#define _InterpolateLinear_H_ - -#include "RateTransposer.h" -#include "STTypes.h" - -namespace soundtouch -{ - -/// Linear transposer class that uses integer arithmetic -class InterpolateLinearInteger : public TransposerBase -{ -protected: - int iFract; - int iRate; - - virtual 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 arithmetic -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 diff --git a/lib/soundtouch/InterpolateShannon.cpp b/lib/soundtouch/InterpolateShannon.cpp deleted file mode 100644 index 1d69a2e8840..00000000000 --- a/lib/soundtouch/InterpolateShannon.cpp +++ /dev/null @@ -1,181 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sample interpolation routine using 8-tap band-limited Shannon interpolation -/// with kaiser window. -/// -/// Notice. This algorithm is remarkably much heavier than linear or cubic -/// interpolation, and not remarkably better than cubic algorithm. Thus mostly -/// for experimental purposes -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#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; -} diff --git a/lib/soundtouch/InterpolateShannon.h b/lib/soundtouch/InterpolateShannon.h deleted file mode 100644 index c621cf1e1d7..00000000000 --- a/lib/soundtouch/InterpolateShannon.h +++ /dev/null @@ -1,68 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sample interpolation routine using 8-tap band-limited Shannon interpolation -/// with kaiser window. -/// -/// Notice. This algorithm is remarkably much heavier than linear or cubic -/// interpolation, and not remarkably better than cubic algorithm. Thus mostly -/// for experimental purposes -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _InterpolateShannon_H_ -#define _InterpolateShannon_H_ - -#include "RateTransposer.h" -#include "STTypes.h" - -namespace soundtouch -{ - -class InterpolateShannon : public TransposerBase -{ -protected: - 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 diff --git a/lib/soundtouch/PeakFinder.cpp b/lib/soundtouch/PeakFinder.cpp deleted file mode 100644 index 258571ab668..00000000000 --- a/lib/soundtouch/PeakFinder.cpp +++ /dev/null @@ -1,277 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Peak detection routine. -/// -/// The routine detects highest value on an array of values and calculates the -/// precise peak location as a mass-center of the 'hump' around the peak value. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include - -#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 = 1; i < 3; i ++) - { - double peaktmp, harmonic; - int i1,i2; - - harmonic = (double)pow(2.0, i); - peakpos = (int)(highPeak / harmonic + 0.5f); - if (peakpos < minPos) break; - peakpos = findTop(data, peakpos); // seek true local maximum index - if (peakpos == 0) continue; // no local max here - - // calculate mass-center of possible harmonic peak - peaktmp = getPeakCenter(data, peakpos); - - // accept harmonic peak if - // (a) it is found - // (b) is within �4% of the expected harmonic interval - // (c) has at least half x-corr value of the max. peak - - double diff = harmonic * peaktmp / highPeak; - if ((diff < 0.96) || (diff > 1.04)) continue; // peak too afar from expected - - // now compare to highest detected peak - i1 = (int)(highPeak + 0.5); - i2 = (int)(peaktmp + 0.5); - if (data[i2] >= 0.4*data[i1]) - { - // The harmonic is at least half as high primary peak, - // thus use the harmonic peak instead - peak = peaktmp; - } - } - - return peak; -} diff --git a/lib/soundtouch/PeakFinder.h b/lib/soundtouch/PeakFinder.h deleted file mode 100644 index 9fe66adac5f..00000000000 --- a/lib/soundtouch/PeakFinder.h +++ /dev/null @@ -1,90 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// The routine detects highest value on an array of values and calculates the -/// precise peak location as a mass-center of the 'hump' around the peak value. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _PeakFinder_H_ -#define _PeakFinder_H_ - -namespace soundtouch -{ - -class PeakFinder -{ -protected: - /// Min, max allowed peak positions within the data vector - int minPos, maxPos; - - /// Calculates the mass center between given vector items. - double calcMassCenter(const float *data, ///< Data vector. - int firstPos, ///< Index of first vector item belonging to the peak. - int lastPos ///< Index of last vector item belonging to the peak. - ) const; - - /// Finds the data vector index where the monotoniously decreasing signal crosses the - /// given level. - int findCrossingLevel(const float *data, ///< Data vector. - float level, ///< Goal crossing level. - int peakpos, ///< Peak position index within the data vector. - int direction /// Direction where to proceed from the peak: 1 = right, -1 = left. - ) const; - - // Finds real 'top' of a peak hump from neighnourhood of the given 'peakpos'. - int findTop(const float *data, int peakpos) const; - - - /// Finds the 'ground' level, i.e. smallest level between two neighbouring peaks, to right- - /// or left-hand side of the given peak position. - int findGround(const float *data, /// Data vector. - int peakpos, /// Peak position index within the data vector. - int direction /// Direction where to proceed from the peak: 1 = right, -1 = left. - ) const; - - /// get exact center of peak near given position by calculating local mass of center - double getPeakCenter(const float *data, int peakpos) const; - -public: - /// Constructor. - PeakFinder(); - - /// Detect exact peak position of the data vector by finding the largest peak 'hump' - /// and calculating the mass-center location of the peak hump. - /// - /// \return The location of the largest base harmonic peak hump. - double detectPeak(const float *data, /// Data vector to be analyzed. The data vector has - /// to be at least 'maxPos' items long. - int minPos, ///< Min allowed peak location within the vector data. - int maxPos ///< Max allowed peak location within the vector data. - ); -}; - -} - -#endif // _PeakFinder_H_ diff --git a/lib/soundtouch/README.html b/lib/soundtouch/README.html deleted file mode 100644 index 93734321e0f..00000000000 --- a/lib/soundtouch/README.html +++ /dev/null @@ -1,914 +0,0 @@ - - - - SoundTouch library README - - - - - - - -
-

SoundTouch audio processing library v2.1

-

SoundTouch library Copyright © Olli Parviainen 2001-2018

-
-

1. Introduction

-

SoundTouch is an open-source audio processing library that allows -changing the sound tempo, pitch and playback rate parameters -independently from each other, i.e.:

-
    -
  • Sound tempo can be increased or decreased while maintaining the -original pitch
    • -
  • Sound pitch can be increased or decreased while maintaining the -original tempo
    • -
  • Change playback rate that affects both tempo and pitch at the -same time
    • -
  • Choose any combination of tempo/pitch/rate
    • -
-

1.1 Contact information

-

Author email: oparviai 'at' iki.fi

-

SoundTouch WWW page: http://soundtouch.surina.net

-

SoundTouch git repository: https://gitlab.com/soundtouch/soundtouch.git

-
-

2. Compiling SoundTouch

-

Before compiling, notice that you can choose the sample data format if it's -desirable to use floating point sample data instead of 16bit integers. See -section "sample data format" for more information.

-

Also notice that SoundTouch can use OpenMP instructions for parallel -computation to accelerate the runtime processing speed in multi-core systems, -however, these improvements need to be separately enabled before compiling. See -OpenMP notes in Chapter 3 below.

-

2.1. Building in Microsoft Windows

-

Project files for Microsoft Visual C++ are supplied with the source -code package. Go to Microsoft WWW page to download - -Microsoft Visual Studio Express version for free. -

-

To build the binaries with Visual C++ compiler, either run -"make-win.bat" script, or open the appropriate project files in source -code directories with Visual Studio. The final executable will appear -under the "SoundTouch\bin" directory. If using the Visual Studio IDE -instead of the make-win.bat script, directories bin and lib may need to -be created manually to the SoundTouch package root for the final -executables. The make-win.bat script creates these directories -automatically.

-

C# example: The source code package includes also a C# example - application for Windows that shows how to invoke SoundTouch.dll - dynamic-load library for processing mp3 audio. -

OpenMP NOTE: If activating the OpenMP parallel computing in -the compilation, the target program will require additional vcomp dll library to -properly run. In Visual C++ 9.0 these libraries can be found in the following -folders.

-
    -
  • x86 32bit: C:\Program Files (x86)\Microsoft Visual Studio - 9.0\VC\redist\x86\Microsoft.VC90.OPENMP\vcomp90.dll
    • -
  • x64 64bit: C:\Program Files (x86)\Microsoft Visual Studio - 9.0\VC\redist\amd64\Microsoft.VC90.OPENMP\vcomp90.dll
    • -
-

In Visual Studio 2008, a SP1 version may be required for these libraries. In -other VC++ versions the required library will be expectedly found in similar -"redist" location.

-

Notice that as minor demonstration of a "dll hell" phenomenon both the 32-bit -and 64-bit version of vcomp90.dll have the same filename but different contents, -thus choose the proper version to allow the program start.

-

2.2. Building in Gnu platforms

-

The SoundTouch library compiles in practically any platform -supporting GNU compiler (GCC) tools. SoundTouch requires GCC version 4.3 or later.

-

To build and install the binaries, run the following commands in -/soundtouch directory:

- - - - - - - - - - - - - - - - - - - -
- 
./bootstrap  -
- 		Creates "configure" file with -local autoconf/automake toolset.
-
- 
./configure  -
- 		-

Configures the SoundTouch package for the local environment. -Notice that "configure" file is not available before running the -"./bootstrap" command as above.
-

-
- 
make         -
- 		-

Builds the SoundTouch library & SoundStretch utility. You can - optionally add "-j" switch after "make" to speed up the compilation in - multi-core systems.

-
- 
make install -
- 		-

Installs the SoundTouch & BPM libraries to /usr/local/lib -and SoundStretch utility to /usr/local/bin. Please notice that -'root' privileges may be required to install the binaries to the -destination locations.

-
-

2.2.1 Required GNU tools

-

Bash shell, GNU C++ compiler, libtool, autoconf and automake tools -are required for compiling the SoundTouch library. These are usually -included with the GNU/Linux distribution, but if not, install these -packages first. For example, Ubuntu Linux can acquire and install -these with the following command:

-
sudo apt-get install automake autoconf libtool build-essential
-

2.2.2 Problems with GCC compiler compatibility

-

At the release time the SoundTouch package has been tested to -compile in GNU/Linux platform. However, If you have problems getting the -SoundTouch library compiled, try disabling optimizations that are specific for -x86 processors by running ./configure script with switch -

-
--enable-x86-optimizations=no
-
- -Alternatively, if you don't use GNU Configure system, edit file "include/STTypes.h" -directly and remove the following definition:
- 
#define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1
-
- -

2.2.3 Compiling Shared Library / DLL version in Cygwin

-

- The GNU compilation does not automatically create a shared-library version of - SoundTouch (.so or .dll). If such is desired, then you can create it as follows - after running the usual compilation:

-
- 
g++ -shared -static -DDLL_EXPORTS -I../../include -o SoundTouch.dll \
-     SoundTouchDLL.cpp ../SoundTouch/.libs/libSoundTouch.a
-sstrip SoundTouch.dll
-
- -

2.3. Building in Android

-

Android compilation instructions are within the - source code package, see file "source/Android-lib/README-SoundTouch-Android.html" - in the source code package.

-

The Android compilation automatically builds separate .so library binaries -for ARM, X86 and MIPS processor architectures. For optimal device support, -include all these .so library binaries into the Android .apk application -package, so the target Android device can automatically choose the proper -library binary version to use.

-

The source/Android-lib folder includes also an Android -example application that processes WAV audio files using SoundTouch library in -Android devices.

- -
-

3. About implementation & Usage tips

3.1. Supported sample data formats

-

The sample data format can be chosen between 16bit signed integer -and 32bit floating point values. The default is 32bit floating point format, -which will also provide slightly better sound quality over the integer format.

-

In Windows environment, the sample data format is chosen in file -"STTypes.h" by choosing one of the following defines:

-
    -
  • #define -SOUNDTOUCH_INTEGER_SAMPLES for 16bit signed integer
    • -
  • #define SOUNDTOUCH_FLOAT_SAMPLES for 32bit floating -point
    • -
-

In GNU environment, the floating sample format is used by default, -but integer sample format can be chosen by giving the following switch -to the configure script:

-
- 
./configure --enable-integer-samples
-
-

The sample data can have either single (mono) or double (stereo) -audio channel. Stereo data is interleaved so that every other data -value is for left channel and every second for right channel. Notice -that while it'd be possible in theory to process stereo sound as two -separate mono channels, this isn't recommended because processing the -channels separately would result in losing the phase coherency between -the channels, which consequently would ruin the stereo effect.

-

Sample rates between 8000-48000H are supported.

-

3.2. Processing latency

-

The processing and latency constraints of the SoundTouch library are:

-
    -
  • Input/output processing latency for the SoundTouch processor is -around 100 ms. This is when time-stretching is used. If the rate -transposing effect alone is used, the latency requirement is much -shorter, see section 'About algorithms'.
    • -
  • Processing CD-quality sound (16bit stereo sound with 44100H -sample rate) in real-time or faster is possible starting from -processors equivalent to Intel Pentium 133Mh or better, if using the -"quick" processing algorithm. If not using the "quick" mode or if -floating point sample data are being used, several times more CPU power -is typically required.
    • -
-

3.3. About algorithms

-

SoundTouch provides three seemingly independent effects: tempo, -pitch and playback rate control. These three controls are implemented -as combination of two primary effects, sample rate transposing -and time-stretching.

-

Sample rate transposing affects both the audio stream -duration and pitch. It's implemented simply by converting the original -audio sample stream to the desired duration by interpolating from -the original audio samples. In SoundTouch, linear interpolation with -anti-alias filtering is used. Theoretically a higher-order -interpolation provide better result than 1st order linear -interpolation, but in audio application linear interpolation together -with anti-alias filtering performs subjectively about as well as -higher-order filtering would.

-

Time-stretching means changing the audio stream duration -without affecting it's pitch. SoundTouch uses WSOLA-like -time-stretching routines that operate in the time domain. Compared to -sample rate transposing, time-stretching is a much heavier operation -and also requires a longer processing "window" of sound samples used by -the processing algorithm, thus increasing the algorithm input/output -latency. Typical i/o latency for the SoundTouch time-stretch algorithm -is around 100 ms.

-

Sample rate transposing and time-stretching are then used together -to produce the tempo, pitch and rate controls:

-
    -
  • 'Tempo' control is implemented purely by -time-stretching.
    • -
  • 'Rate' control is implemented purely by sample -rate transposing.
    • -
  • 'Pitch' control is implemented as a -combination of time-stretching and sample rate transposing. For -example, to increase pitch the audio stream is first time-stretched to -longer duration (without affecting pitch) and then transposed back to -original duration by sample rate transposing, which simultaneously -reduces duration and increases pitch. The result is original duration -but increased pitch.
    • -
-

3.4 Tuning the algorithm parameters

-

The time-stretch algorithm has few parameters that can be tuned to -optimize sound quality for certain application. The current default -parameters have been chosen by iterative if-then analysis (read: "trial -and error") to obtain best subjective sound quality in pop/rock music -processing, but in applications processing different kind of sound the -default parameter set may result into a sub-optimal result.

-

The time-stretch algorithm default parameter values are set by the -following #defines in file "TDStretch.h":

-
- 
#define DEFAULT_SEQUENCE_MS     AUTOMATIC
#define DEFAULT_SEEKWINDOW_MS   AUTOMATIC
#define DEFAULT_OVERLAP_MS      8
-
-

These parameters affect to the time-stretch algorithm as follows:

-
    -
  • DEFAULT_SEQUENCE_MS: This is the default -length of a single processing sequence in milliseconds which determines -the how the original sound is chopped in the time-stretch algorithm. -Larger values mean fewer sequences are used in processing. In principle -a larger value sounds better when slowing down the tempo, but worse -when increasing the tempo and vice versa.
    -
    -By default, this setting value is calculated automatically according to -tempo value.
    -
    • -
  • DEFAULT_SEEKWINDOW_MS: The seeking window -default length in milliseconds is for the algorithm that seeks the best -possible overlapping location. This determines from how wide a sample -"window" the algorithm can use to find an optimal mixing location when -the sound sequences are to be linked back together.
    -
    -The bigger this window setting is, the higher the possibility to find a -better mixing position becomes, but at the same time large values may -cause a "drifting" sound artifact because neighboring sequences can be -chosen at more uneven intervals. If there's a disturbing artifact that -sounds as if a constant frequency was drifting around, try reducing -this setting.
    -
    -By default, this setting value is calculated automatically according to -tempo value.
    -
    • -
  • DEFAULT_OVERLAP_MS: Overlap length in -milliseconds. When the sound sequences are mixed back together to form -again a continuous sound stream, this parameter defines how much the -ends of the consecutive sequences will overlap with 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.
    • -
-

Notice that these parameters can also be set during execution time -with functions "TDStretch::setParameters()" and "SoundTouch::setSetting()".

-

The table below summaries how the parameters can be adjusted for -different applications:

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Parameter nameDefault value magnitudeLarger value affects...Smaller value affects...Effect to CPU burden
- 
SEQUENCE_MS
- 		Default value is relatively large, chosen for -slowing down music tempoLarger value is usually better for slowing down -tempo. Growing the value decelerates the "echoing" artifact when -slowing down the tempo.Smaller value might be better for speeding up -tempo. Reducing the value accelerates the "echoing" artifact when -slowing down the tempo Increasing the parameter value reduces -computation burden
- 
SEEKWINDOW_MS
- 		Default value is relatively large, chosen for -slowing down music tempoLarger value eases finding a good mixing -position, but may cause a "drifting" artifactSmaller reduce possibility to find a good mixing -position, but reduce the "drifting" artifact.Increasing the parameter value increases -computation burden
- 
OVERLAP_MS
- 		Default value is relatively large, chosen to -suit with above parameters.If you reduce the "sequence ms" setting, you -might wish to try a smaller value.Increasing the parameter value increases -computation burden
-

3.5 Performance Optimizations

-

General optimizations:

-

The time-stretch routine has a 'quick' mode that substantially -speeds up the algorithm but may slightly compromise the sound quality. -This mode is activated by calling SoundTouch::setSetting() -function with parameter id of SETTING_USE_QUICKSEEK and value -"1", i.e.

-
-

setSetting(SETTING_USE_QUICKSEEK, 1);

-
-

CPU-specific optimizations:

-

Intel x86 specific SIMD optimizations are implemented using compiler -intrinsics, providing about a 3x processing speedup for x86 compatible -processors vs. non-SIMD implementation:

-
    -
  • Intel MMX optimized routines are used with x86 CPUs when 16bit integer - sample type is used
    • -
  • Intel SSE optimized routines are used with x86 CPUs when 32bit floating - point sample type is used
    • -
-

3.5 OpenMP parallel computation

-

SoundTouch 1.9 onwards support running the algorithms parallel in several CPU -cores. Based on benchmark the experienced multi-core processing speed-up gain -ranges between +30% (on a high-spec dual-core x86 Windows PC) to 215% (on a moderately low-spec -quad-core ARM of Raspberry Pi2).

-

See an external blog article with more detailed discussion about the - -SoundTouch OpenMP optimization.

-

The parallel computing support is implemented using OpenMP spec 3.0 -instructions. These instructions are supported by Visual C++ 2008 and later, and -GCC v4.2 and later. Compilers that do not supporting OpenMP will ignore these -optimizations and routines will still work properly. Possible warnings about -unknown #pragmas are related to OpenMP support and can be safely ignored.

-

The OpenMP improvements are disabled by default, and need to be enabled by -developer during compile-time. Reason for this is that parallel processing adds -moderate runtime overhead in managing the multi-threading, so it may not be -necessary nor desirable in all applications. For example real-time processing -that is not constrained by CPU power will not benefit of speed-up provided by -the parallel processing, in the contrary it may increase power consumption due -to the increased overhead.

-

However, applications that run on low-spec multi-core CPUs and may otherwise -have possibly constrained performance will benefit of the OpenMP improvements. -This include for example multi-core embedded devices.

-

OpenMP parallel computation can be enabled before compiling SoundTouch -library as follows:

-
    -
  • Visual Studio: Open properties for the SoundTouch - sub-project, browse to C/C++ and Language - settings. Set - there "OpenMP support" to "Yes". Alternatively add - /openmp switch to command-line - parameters
    • -
  • GNU: Run the configure script with "./configure - --enable-openmp" switch, then run make as usually
    • -
  • Android: Add "-fopenmp" switches to compiler & linker - options, see README-SoundTouch-Android.html in the source code package for - more detailed instructions.
    • -
-
-

4. SoundStretch audio processing utility -

-

SoundStretch audio processing utility
- Copyright (c) Olli Parviainen 2002-2015

-

SoundStretch is a simple command-line application that can change -tempo, pitch and playback rates of WAV sound files. This program is -intended primarily to demonstrate how the "SoundTouch" library can be -used to process sound in your own program, but it can as well be used -for processing sound files.

-

4.1. SoundStretch Usage Instructions

-

SoundStretch Usage syntax:

-
- 
soundstretch infilename outfilename [switches]
-
-

Where:

- - - - - - - - - - - - - - - -
- 
"infilename"
- 		Name of the input sound data file (in .WAV audio -file format). Give "stdin" as filename to use standard input pipe.
- 
"outfilename"
- 		Name of the output sound file where the -resulting sound is saved (in .WAV audio file format). This parameter -may be omitted if you don't want to save the output (e.g. when -only calculating BPM rate with '-bpm' switch). Give "stdout" as -filename to use standard output pipe.
- 
[switches]
- 		Are one or more control switches.
-

Available control switches are:

- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- 
-tempo=n
- 		Change the sound tempo by n percents (n = -95.0 -.. +5000.0 %)
- 
-pitch=n
- 		Change the sound pitch by n semitones (n = -60.0 -.. + 60.0 semitones)
- 
-rate=n
- 		Change the sound playback rate by n percents (n -= -95.0 .. +5000.0 %)
- 
-bpm=n
- 		Detect the Beats-Per-Minute (BPM) rate of the -sound and adjust the tempo to meet 'n' BPMs. When this switch is -applied, the "-tempo" switch is ignored. If "=n" is omitted, i.e. -switch "-bpm" is used alone, then the BPM rate is estimated and -displayed, but tempo not adjusted according to the BPM value.
- 
-quick
- 		Use quicker tempo change algorithm. Gains speed -but loses sound quality.
- 
-naa
- 		Don't use anti-alias filtering in sample rate -transposing. Gains speed but loses sound quality.
- 
-license
- 		Displays the program license text (LGPL)
-

Notes:

-
    -
  • To use standard input/output pipes for processing, give "stdin" -and "stdout" as input/output filenames correspondingly. The standard -input/output pipes will still carry the audio data in .wav audio file -format.
    • -
  • The numerical switches allow both integer (e.g. "-tempo=123") -and decimal (e.g. "-tempo=123.45") numbers.
    • -
  • The "-naa" and/or "-quick" switches can be used to reduce CPU -usage while compromising some sound quality
    • -
  • The BPM detection algorithm works by detecting repeating bass or -drum patterns at low frequencies of <250Hz. A lower-than-expected -BPM figure may be reported for music with uneven or complex bass -patterns.
    • -
-

4.2. SoundStretch usage examples

-

Example 1

-

The following command increases tempo of the sound file -"originalfile.wav" by 12.5% and stores result to file -"destinationfile.wav":

-
- 
soundstretch originalfile.wav destinationfile.wav -tempo=12.5
-
-

Example 2

-

The following command decreases the sound pitch (key) of the sound -file "orig.wav" by two semitones and stores the result to file -"dest.wav":

-
- 
soundstretch orig.wav dest.wav -pitch=-2
-
-

Example 3

-

The following command processes the file "orig.wav" by decreasing -the sound tempo by 25.3% and increasing the sound pitch (key) by 1.5 -semitones. Resulting .wav audio data is directed to standard output -pipe:

-
- 
soundstretch orig.wav stdout -tempo=-25.3 -pitch=1.5
-
-

Example 4

-

The following command detects the BPM rate of the file "orig.wav" -and adjusts the tempo to match 100 beats per minute. Result is stored -to file "dest.wav":

-
- 
soundstretch orig.wav dest.wav -bpm=100
-
-

Example 5

-

The following command reads .wav sound data from standard input pipe -and estimates the BPM rate:

-
- 
soundstretch stdin -bpm
-
-

Example 6

-

The following command tunes song from original 440Hz tuning to 432Hz tuning: -this corresponds to lowering the pitch by -0.318 semitones:

-
- 
soundstretch original.wav output.wav -pitch=-0.318
-
-
-

5. Change History

-

5.1. SoundTouch library Change History

-

2.1:

-
    -
  • Refactored C# interface example
    • -
  • Disable anti-alias filter when switch - SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER defined because anti-alias - filter cause slight click if the rate change crosses zero during - processing
    • -
  • Added script for building SoundTouchDll dynamic-link-library for GNU platforms
    • -
  • Rewrote Beats-per-Minute analysis algorithm for more reliable BPM detection
    • -
  • Added BPM functions to SoundTouchDll API
    • -
  • Migrated Visual Studio project files to MSVC 201x format
    • -
  • Replaced function parameter value asserts with runtime exceptions
    • -
  • Code maintenance & style cleanup
    • -
-

2.0:

-
    -
  • Added functions to get initial processing latency, duration ratio between the original input and processed output tracks, and clarified reporting of input/output batch sizes
    • -
  • Fixed issue that added brief sequence of silence to beginning of output audio
    • -
  • Adjusted algorithm parameters to reduce reverberating effect at tempo slowdown
    • -
  • Bugfix: Fixed a glitch that could cause negative array indexing in quick seek algorithm
    • -
  • Bugfix: flush() didn't properly flush final samples from the pipeline on 2nd time in case that soundtouch object instance was recycled and used for processing a second audio stream.
    • -
  • Bugfix: Pi value had incorrect 9th/10th decimals
    • -
  • Added C# example application that uses SoundTouch dll library for processing MP3 files
    • -
-

1.9.2:

-
    -
  • Fix in GNU package configuration
    • -
-

1.9.1:

-
    -
  • Improved SoundTouch::flush() function so that it returns precisely the desired amount of samples for exact output duration control
    • -
  • Redesigned quickseek algorithm for improved sound quality when using the quickseek mode. The new quickseek algorithm can find 99% as good results as the - default full-scan mode, while the quickseek algorithm is remarkable less - CPU intensive.
    • -
  • Added adaptive integer divider scaling for improved sound quality when using integer processing algorithm -
    • -
-

1.9:

-
    -
  • Added support for parallel computation support via OpenMP primitives for better performance in multicore systems. - Benchmarks show that achieved parallel processing speedup improvement - typically range from +30% (x86 dual-core) to +180% (ARM quad-core). The - OpenMP optimizations are disabled by default, see OpenMP notes above in this - readme file how to enabled these optimizations.
    • -
  • Android: Added support for Android devices featuring X86 and MIPS CPUs, - in addition to ARM CPUs.
    • -
  • Android: More versatile Android example application that processes WAV - audio files with SoundTouch library
    • -
  • Replaced Windows-like 'BOOL' types with native 'bool'
    • -
  • Changed documentation token to "dist_doc_DATA" in Makefile.am file
    • -
  • Miscellaneous small fixes and improvements
    • -
-

1.8.0:

-
    -
  • Added support for multi-channel audio processing
    • -
  • Added support for cubic and shannon interpolation for rate and pitch shift effects besides - the original linear interpolation, to reduce aliasing at high frequencies due to interpolation. - Cubic interpolation is used as default for floating point processing, and linear interpolation for integer - processing.
    • -
  • Fixed bug in anti-alias filtering that limited stop-band attenuation to -10 dB instead of <-50dB, and - increased filter length from 32 to 64 taps to further reduce aliasing due to frequency folding.
    • -
  • Performance improvements in cross-correlation algorithm
    • -
  • Other bug and compatibility fixes
    • -
-

1.7.1:

-
    -
  • Added files for Android compilation -
-

1.7.0:

-
    -
  • Sound quality improvements/li> -
  • Improved flush() to adjust output sound stream duration to match better with - ideal duration
    • -
  • Rewrote x86 cpu feature check to resolve compatibility problems
    • -
  • Configure script automatically checks if CPU supports mmx & sse compatibility for GNU platform, and - the script support now "--enable-x86-optimizations" switch to allow disabling x86-specific optimizations.
    • -
  • Revised #define conditions for 32bit/64bit compatibility
    • -
  • gnu autoconf/automake script compatibility fixes
    • -
  • Tuned beat-per-minute detection algorithm
    • -
-

1.6.0:

-
    -
  • Added automatic cutoff threshold adaptation to beat detection -routine to better adapt BPM calculation to different types of music
    • -
  • Retired 3DNow! optimization support as 3DNow! is nowadays -obsoleted and assembler code is nuisance to maintain
    • -
  • Retired "configure" file from source code package due to -autoconf/automake versio conflicts, so that it is from now on to be -generated by invoking "boostrap" script that uses locally available -toolchain version for generating the "configure" file
    • -
  • Resolved namespace/label naming conflicts with other libraries by -replacing global labels such as INTEGER_SAMPLES with more specific -SOUNDTOUCH_INTEGER_SAMPLES etc.
    -
    • -
  • Updated windows build scripts & project files for Visual -Studio 2008 support
    • -
  • Updated SoundTouch.dll API for .NET compatibility
    • -
  • Added API for querying nominal processing input & output -sample batch sizes
    • -
-

1.5.0:

-
    -
  • Added normalization to correlation calculation and improvement -automatic seek/sequence parameter calculation to improve sound quality
    • -
  • Bugfixes: -
      -
    • Fixed negative array indexing in quick seek algorithm
      • -
    • FIR autoalias filter running too far in processing buffer
      • -
    • Check against zero sample count in rate transposing
      • -
    • Fix for x86-64 support: Removed pop/push instructions from -the cpu detection algorithm.
      • -
    • Check against empty buffers in FIFOSampleBuffer
      • -
    • Other minor fixes & code cleanup
      • -
    -
    • -
  • Fixes in compilation scripts for non-Intel platforms
    • -
  • Added Dynamic-Link-Library (DLL) version of SoundTouch library -build, provided with Delphi/Pascal wrapper for calling the dll routines -
    • -
  • Added #define PREVENT_CLICK_AT_RATE_CROSSOVER that prevents a -click artifact when crossing the nominal pitch from either positive to -negative side or vice versa
    • -
-

1.4.1:

-
    -
  • Fixed a buffer overflow bug in BPM detect algorithm routines if -processing more than 2048 samples at one call
    • -
-

1.4.0:

-
    -
  • Improved sound quality by automatic calculation of time stretch -algorithm processing parameters according to tempo setting
    • -
  • Moved BPM detection routines from SoundStretch application into -SoundTouch library
    • -
  • Bugfixes: Usage of uninitialied variables, GNU build scripts, -compiler errors due to 'const' keyword mismatch.
    • -
  • Source code cleanup
    • -
-

1.3.1:

-
    -
  • Changed static class declaration to GCC 4.x compiler compatible -syntax.
    • -
  • Enabled MMX/SSE-optimized routines also for GCC compilers. -Earlier the MMX/SSE-optimized routines were written in -compiler-specific inline assembler, now these routines are migrated to -use compiler intrinsic syntax which allows compiling the same -MMX/SSE-optimized source code with both Visual C++ and GCC compilers.
    • -
  • Set floating point as the default sample format and added switch -to the GNU configure script for selecting the other sample format.
    • -
-

1.3.0:

-
    -
  • Fixed tempo routine output duration inaccuracy due to rounding -error
    • -
  • Implemented separate processing routines for integer and -floating arithmetic to allow improvements to floating point routines -(earlier used algorithms mostly optimized for integer arithmetic also -for floating point samples)
    • -
  • Fixed a bug that distorts sound if sample rate changes during -the sound stream
    • -
  • Fixed a memory leak that appeared in MMX/SSE/3DNow! optimized -routines
    • -
  • Reduced redundant code pieces in MMX/SSE/3DNow! optimized -routines vs. the standard C routines.
    • -
  • MMX routine incompatibility with new gcc compiler versions
    • -
  • Other miscellaneous bug fixes
    • -
-

1.2.1:

-
    -
  • Added automake/autoconf scripts for GNU platforms (in courtesy -of David Durham)
    • -
  • Fixed SCALE overflow bug in rate transposer routine.
    • -
  • Fixed 64bit address space bugs.
    • -
  • Created a 'soundtouch' namespace for SAMPLETYPE definitions.
    • -
-

1.2.0:

-
    -
  • Added support for 32bit floating point sample data type with -SSE/3DNow! optimizations for Win32 platform (SSE/3DNow! optimizations -currently not supported in GCC environment)
    • -
  • Replaced 'make-gcc' script for GNU environment by master -Makefile
    • -
  • Added time-stretch routine configurability to SoundTouch main -class
    • -
  • Bugfixes
    • -
-

1.1.1:

-
    -
  • Moved SoundTouch under lesser GPL license (LGPL). This allows -using SoundTouch library in programs that aren't released under GPL -license.
    • -
  • Changed MMX routine organiation so that MMX optimized routines -are now implemented in classes that are derived from the basic classes -having the standard non-mmx routines.
    • -
  • MMX routines to support gcc version 3.
    • -
  • Replaced windows makefiles by script using the .dsw files
    • -
-

1.0.1:

-
    -
  • "mmx_gcc.cpp": Added "using namespace std" and removed "return -0" from a function with void return value to fix compiler errors when -compiling the library in Solaris environment.
    • -
  • Moved file "FIFOSampleBuffer.h" to "include" directory to allow -accessing the FIFOSampleBuffer class from external files.
    • -
-

1.0:

-
    -
  • Initial release
    • -
-

5.2. SoundStretch application Change History

-

1.9:

-
    -
  • Added support for WAV file 'fact' information chunk.
    • -
- -

1.7.0:

-
    -
  • Bugfixes in Wavfile: exception string formatting, avoid getLengthMs() integer - precision overflow, support WAV files using 24/32bit sample format.
    • -
-

1.5.0:

-
    -
  • Added "-speech" switch to activate algorithm parameters more -suitable for speech processing than the default parameters tuned for -music processing.
    • -
-

1.4.0:

-
    -
  • Moved BPM detection routines from SoundStretch application into -SoundTouch library
    • -
  • Allow using standard input/output pipes as audio processing -input/output streams
    • -
-

1.3.0:

-
    -
  • Simplified accessing WAV files with floating point sample -format.
    • -
-

1.2.1:

-
    -
  • Fixed 64bit address space bugs.
    • -
-

1.2.0:

-
    -
  • Added support for 32bit floating point sample data type
    • -
  • Restructured the BPM routines into separate library
    • -
  • Fixed big-endian conversion bugs in WAV file routines (hopefully -:)
    • -
-

1.1.1:

-
    -
  • Fixed bugs in WAV file reading & added byte-order conversion -for big-endian processors.
    • -
  • Moved SoundStretch source code under 'example' directory to -highlight difference from SoundTouch stuff.
    • -
  • Replaced windows makefiles by script using the .dsw files
    • -
  • Output file name isn't required if output isn't desired (e.g. if -using the switch '-bpm' in plain format only)
    • -
-

1.1:

-
    -
  • Fixed "Release" settings in Microsoft Visual C++ project file -(.dsp)
    • -
  • Added beats-per-minute (BPM) detection routine and command-line -switch "-bpm"
    • -
-

1.01:

-
    -
  • Initial release
    • -
-
-

6. Acknowledgements

-

Kudos for these people who have contributed to development or -submitted bugfixes:

-
    -
  • Arthur A
    • -
  • Paul Adenot
    • -
  • Richard Ash
    • -
  • Stanislav Brabec
    • -
  • Christian Budde
    • -
  • Jamie Bullock
    • -
  • Chris Bryan
    • -
  • Jacek Caban
    • -
  • Brian Cameron
    • -
  • Jason Champion
    • -
  • David Clark
    • -
  • Patrick Colis
    • -
  • Miquel Colon
    • -
  • Jim Credland
    • -
  • Sandro Cumerlato
    • -
  • Justin Frankel
    • -
  • Masa H.
    • -
  • Jason Garland
    • -
  • Takashi Iwai
    • -
  • Thomas Klausner
    • -
  • Lu Zhihe
    • -
  • Luzpaz
    • -
  • Tony Mechelynck
    • -
  • Mathias Möhl
    • -
  • Yuval Naveh
    • -
  • Mats Palmgren
    • -
  • Chandni Patel
    • -
  • Paulo Pizarro
    • -
  • Andrey Ponomarenko
    • -
  • Blaise Potard
    • -
  • Michael Pruett
    • -
  • Rajeev Puran
    • -
  • RJ Ryan
    • -
  • John Sheehy
    • -
  • Tim Shuttleworth
    • -
  • Albert Sirvent
    • -
  • Tyson Smith
    • -
  • John Stumpo
    • -
  • Mario di Vece
    • -
  • Katja Vetter
    • -
  • Wu Q.
    • -
-

Moral greetings to all other contributors and users also!

-
-

7. 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 version 2.1 -as published by the Free Software Foundation.

-

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

-

---

-

commercial license alternative also available, contact author for details.

-
-

README.html file updated in May-2018

- - diff --git a/lib/soundtouch/RateTransposer.cpp b/lib/soundtouch/RateTransposer.cpp deleted file mode 100644 index 2efaf0424df..00000000000 --- a/lib/soundtouch/RateTransposer.cpp +++ /dev/null @@ -1,307 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sample rate transposer. Changes sample rate by using linear interpolation -/// together with anti-alias filtering (first order interpolation with anti- -/// alias filtering should be quite adequate for this application) -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#include "RateTransposer.h" -#include "InterpolateLinear.h" -#include "InterpolateCubic.h" -#include "InterpolateShannon.h" -#include "AAFilter.h" - -using namespace soundtouch; - -// Define default interpolation algorithm here -TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC; - - -// Constructor -RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer) -{ - bUseAAFilter = -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - true; -#else - // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover - false; -#endif - - // Instantiates the anti-alias filter - pAAFilter = new AAFilter(64); - pTransposer = TransposerBase::newInstance(); -} - - -RateTransposer::~RateTransposer() -{ - delete pAAFilter; - delete pTransposer; -} - - -/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable -void RateTransposer::enableAAFilter(bool newMode) -{ -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover - bUseAAFilter = newMode; -#endif -} - - -/// Returns nonzero if anti-alias filter is enabled. -bool RateTransposer::isAAFilterEnabled() const -{ - return bUseAAFilter; -} - - -AAFilter *RateTransposer::getAAFilter() -{ - return pAAFilter; -} - - -// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower -// iRate, larger faster iRates. -void RateTransposer::setRate(double newRate) -{ - double fCutoff; - - pTransposer->setRate(newRate); - - // design a new anti-alias filter - if (newRate > 1.0) - { - fCutoff = 0.5 / newRate; - } - else - { - fCutoff = 0.5 * newRate; - } - pAAFilter->setCutoffFreq(fCutoff); -} - - -// Adds 'nSamples' pcs of samples from the 'samples' memory position into -// the input of the object. -void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples) -{ - processSamples(samples, nSamples); -} - - -// Transposes sample rate by applying anti-alias filter to prevent folding. -// Returns amount of samples returned in the "dest" buffer. -// The maximum amount of samples that can be returned at a time is set by -// the 'set_returnBuffer_size' function. -void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples) -{ - 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) -{ - if (!verifyNumberOfChannels(nChannels) || - (pTransposer->numChannels == nChannels)) return; - - pTransposer->setChannels(nChannels); - inputBuffer.setChannels(nChannels); - midBuffer.setChannels(nChannels); - outputBuffer.setChannels(nChannels); -} - - -// Clears all the samples in the object -void RateTransposer::clear() -{ - outputBuffer.clear(); - midBuffer.clear(); - inputBuffer.clear(); -} - - -// Returns nonzero if there aren't any samples available for outputting. -int RateTransposer::isEmpty() const -{ - int res; - - res = FIFOProcessor::isEmpty(); - if (res == 0) return 0; - return inputBuffer.isEmpty(); -} - - -/// Return approximate initial input-output latency -int RateTransposer::getLatency() const -{ - return (bUseAAFilter) ? pAAFilter->getLength() : 0; -} - - -////////////////////////////////////////////////////////////////////////////// -// -// TransposerBase - Base class for interpolation -// - -// static function to set interpolation algorithm -void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a) -{ - TransposerBase::algorithm = a; -} - - -// Transposes the sample rate of the given samples using linear interpolation. -// Returns the number of samples returned in the "dest" buffer -int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) -{ - int numSrcSamples = src.numSamples(); - int sizeDemand = (int)((double)numSrcSamples / rate) + 8; - int numOutput; - SAMPLETYPE *psrc = src.ptrBegin(); - SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand); - -#ifndef USE_MULTICH_ALWAYS - if (numChannels == 1) - { - numOutput = transposeMono(pdest, psrc, numSrcSamples); - } - else if (numChannels == 2) - { - numOutput = transposeStereo(pdest, psrc, numSrcSamples); - } - else -#endif // USE_MULTICH_ALWAYS - { - assert(numChannels > 0); - numOutput = transposeMulti(pdest, psrc, numSrcSamples); - } - dest.putSamples(numOutput); - src.receiveSamples(numSrcSamples); - return numOutput; -} - - -TransposerBase::TransposerBase() -{ - numChannels = 0; - rate = 1.0f; -} - - -TransposerBase::~TransposerBase() -{ -} - - -void TransposerBase::setChannels(int channels) -{ - numChannels = channels; - resetRegisters(); -} - - -void TransposerBase::setRate(double newRate) -{ - rate = newRate; -} - - -// static factory function -TransposerBase *TransposerBase::newInstance() -{ -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - // Notice: For integer arithmetic support only linear algorithm (due to simplest calculus) - return ::new InterpolateLinearInteger; -#else - switch (algorithm) - { - case LINEAR: - return new InterpolateLinearFloat; - - case CUBIC: - return new InterpolateCubic; - - case SHANNON: - return new InterpolateShannon; - - default: - assert(false); - return NULL; - } -#endif -} diff --git a/lib/soundtouch/RateTransposer.h b/lib/soundtouch/RateTransposer.h deleted file mode 100644 index 52b7441b89e..00000000000 --- a/lib/soundtouch/RateTransposer.h +++ /dev/null @@ -1,163 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sample rate transposer. Changes sample rate by using linear interpolation -/// together with anti-alias filtering (first order interpolation with anti- -/// alias filtering should be quite adequate for this application). -/// -/// Use either of the derived classes of 'RateTransposerInteger' or -/// 'RateTransposerFloat' for corresponding integer/floating point tranposing -/// algorithm implementation. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef RateTransposer_H -#define RateTransposer_H - -#include -#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(); - - /// Returns the output buffer object - FIFOSamplePipe *getOutput() { return &outputBuffer; }; - - /// Return anti-alias filter object - AAFilter *getAAFilter(); - - /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable - void enableAAFilter(bool newMode); - - /// Returns nonzero if anti-alias filter is enabled. - bool isAAFilterEnabled() const; - - /// Sets new target rate. Normal rate = 1.0, smaller values represent slower - /// rate, larger faster rates. - virtual void setRate(double newRate); - - /// Sets the number of channels, 1 = mono, 2 = stereo - void setChannels(int channels); - - /// Adds 'numSamples' pcs of samples from the 'samples' memory position into - /// the input of the object. - void putSamples(const SAMPLETYPE *samples, uint numSamples); - - /// Clears all the samples in the object - void clear(); - - /// Returns nonzero if there aren't any samples available for outputting. - int isEmpty() const; - - /// Return approximate initial input-output latency - int getLatency() const; -}; - -} - -#endif diff --git a/lib/soundtouch/SConscript b/lib/soundtouch/SConscript deleted file mode 100644 index 1df1e68b288..00000000000 --- a/lib/soundtouch/SConscript +++ /dev/null @@ -1,27 +0,0 @@ -#!/usr/bin/env python - -Import('env') -env = env.Clone() - -soundtouch_sources = [ - 'AAFilter.cpp', - 'BPMDetect.cpp', - 'FIFOSampleBuffer.cpp', - 'FIRFilter.cpp', - 'InterpolateCubic.cpp', - 'InterpolateLinear.cpp', - 'InterpolateShannon.cpp', - 'PeakFinder.cpp', - 'RateTransposer.cpp', - 'SoundTouch.cpp', - 'TDStretch.cpp', - # SoundTouch CPU optimizations are only for x86 - # architectures. SoundTouch automatically ignores these files - # when it is not being built for an architecture that supports - # them. - 'cpu_detect_x86.cpp', - 'mmx_optimized.cpp', - 'sse_optimized.cpp', -] - -env.StaticLibrary(target='soundtouch', source=soundtouch_sources) diff --git a/lib/soundtouch/STTypes.h b/lib/soundtouch/STTypes.h deleted file mode 100644 index 862505e7691..00000000000 --- a/lib/soundtouch/STTypes.h +++ /dev/null @@ -1,183 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Common type definitions for SoundTouch audio processing library. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef STTypes_H -#define STTypes_H - -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 -{ - /// Max allowed number of channels - #define SOUNDTOUCH_MAX_CHANNELS 16 - - /// Activate these undef's to overrule the possible sampletype - /// setting inherited from some other header file: - //#undef SOUNDTOUCH_INTEGER_SAMPLES - //#undef SOUNDTOUCH_FLOAT_SAMPLES - - /// If following flag is defined, always uses multichannel processing - /// routines also for mono and stero sound. This is for routine testing - /// purposes; output should be same with either routines, yet disabling - /// the dedicated mono/stereo processing routines will result in slower - /// runtime performance so recommendation is to keep this off. - // #define USE_MULTICH_ALWAYS - - #if (defined(__SOFTFP__) && defined(ANDROID)) - // For Android compilation: Force use of Integer samples in case that - // compilation uses soft-floating point emulation - soft-fp is way too slow - #undef SOUNDTOUCH_FLOAT_SAMPLES - #define SOUNDTOUCH_INTEGER_SAMPLES 1 - #endif - - #if !(SOUNDTOUCH_INTEGER_SAMPLES || SOUNDTOUCH_FLOAT_SAMPLES) - - /// Choose either 32bit floating point or 16bit integer sampletype - /// by choosing one of the following defines, unless this selection - /// has already been done in some other file. - //// - /// Notes: - /// - In Windows environment, choose the sample format with the - /// following defines. - /// - In GNU environment, the floating point samples are used by - /// default, but integer samples can be chosen by giving the - /// following switch to the configure script: - /// ./configure --enable-integer-samples - /// However, if you still prefer to select the sample format here - /// also in GNU environment, then please #undef the INTEGER_SAMPLE - /// and FLOAT_SAMPLE defines first as in comments above. - //#define SOUNDTOUCH_INTEGER_SAMPLES 1 //< 16bit integer samples - #define SOUNDTOUCH_FLOAT_SAMPLES 1 //< 32bit float samples - - #endif - - #if (_M_IX86 || __i386__ || __x86_64__ || _M_X64) - /// Define this to allow X86-specific assembler/intrinsic optimizations. - /// Notice that library contains also usual C++ versions of each of these - /// these routines, so if you're having difficulties getting the optimized - /// routines compiled for whatever reason, you may disable these optimizations - /// to make the library compile. - - #define SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS 1 - - /// In GNU environment, allow the user to override this setting by - /// giving the following switch to the configure script: - /// ./configure --disable-x86-optimizations - /// ./configure --enable-x86-optimizations=no - #ifdef SOUNDTOUCH_DISABLE_X86_OPTIMIZATIONS - #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS - #endif - #else - /// Always disable optimizations when not using a x86 systems. - #undef SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS - - #endif - - // If defined, allows the SIMD-optimized routines to 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 (not available in X64 mode) - #if (!_M_X64) - #define SOUNDTOUCH_ALLOW_MMX 1 - #endif - #endif - - #else - - // floating point samples - typedef float SAMPLETYPE; - // data type for sample accumulation: Use 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 - #define ST_THROW_RT_ERROR(x) {assert((const char *)x);} -#else - // use c++ standard exceptions - #include - #include - #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 diff --git a/lib/soundtouch/SoundTouch.cpp b/lib/soundtouch/SoundTouch.cpp deleted file mode 100644 index 1618884cf51..00000000000 --- a/lib/soundtouch/SoundTouch.cpp +++ /dev/null @@ -1,538 +0,0 @@ -////////////////////////////////////////////////////////////////////////////// -/// -/// SoundTouch - main class for tempo/pitch/rate adjusting routines. -/// -/// Notes: -/// - Initialize the SoundTouch object instance by setting up the sound stream -/// parameters with functions 'setSampleRate' and 'setChannels', then set -/// desired tempo/pitch/rate settings with the corresponding functions. -/// -/// - The SoundTouch class behaves like a first-in-first-out pipeline: The -/// samples that are to be processed are fed into one of the pipe by calling -/// function 'putSamples', while the ready processed samples can be read -/// from the other end of the pipeline with function 'receiveSamples'. -/// -/// - The SoundTouch processing classes require certain sized 'batches' of -/// samples in order to process the sound. For this reason the classes buffer -/// incoming samples until there are enough of samples available for -/// processing, then they carry out the processing step and consequently -/// make the processed samples available for outputting. -/// -/// - For the above reason, the processing routines introduce a certain -/// 'latency' between the input and output, so that the samples input to -/// SoundTouch may not be immediately available in the output, and neither -/// the amount of outputtable samples may not immediately be in direct -/// relationship with the amount of previously input samples. -/// -/// - The tempo/pitch/rate control parameters can be altered during processing. -/// Please notice though that they aren't currently protected by semaphores, -/// so in multi-thread application external semaphore protection may be -/// required. -/// -/// - This class utilizes classes 'TDStretch' for tempo change (without modifying -/// pitch) and 'RateTransposer' for changing the playback rate (that is, both -/// tempo and pitch in the same ratio) of the sound. The third available control -/// 'pitch' (change pitch but maintain tempo) is produced by a combination of -/// combining the two other controls. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#include - -#include "SoundTouch.h" -#include "TDStretch.h" -#include "RateTransposer.h" -#include "cpu_detect.h" - -using namespace soundtouch; - -/// test if two floating point numbers are equal -#define TEST_FLOAT_EQUAL(a, b) (fabs(a - b) < 1e-10) - - -/// Print library version string for autoconf -extern "C" void soundtouch_ac_test() -{ - printf("SoundTouch Version: %s\n",SOUNDTOUCH_VERSION); -} - - -SoundTouch::SoundTouch() -{ - // Initialize rate transposer and tempo changer instances - - pRateTransposer = new RateTransposer(); - pTDStretch = TDStretch::newInstance(); - - setOutPipe(pTDStretch); - - rate = tempo = 0; - - virtualPitch = - virtualRate = - virtualTempo = 1.0; - - calcEffectiveRateAndTempo(); - - samplesExpectedOut = 0; - samplesOutput = 0; - - channels = 0; - bSrateSet = false; -} - - -SoundTouch::~SoundTouch() -{ - delete pRateTransposer; - delete pTDStretch; -} - - -/// Get SoundTouch library version string -const char *SoundTouch::getVersionString() -{ - static const char *_version = SOUNDTOUCH_VERSION; - - return _version; -} - - -/// Get SoundTouch library version Id -uint SoundTouch::getVersionId() -{ - return SOUNDTOUCH_VERSION_ID; -} - - -// Sets the number of channels, 1 = mono, 2 = stereo -void SoundTouch::setChannels(uint numChannels) -{ - if (!verifyNumberOfChannels(numChannels)) return; - - channels = numChannels; - pRateTransposer->setChannels((int)numChannels); - pTDStretch->setChannels((int)numChannels); -} - - -// Sets new rate control value. Normal rate = 1.0, smaller values -// represent slower rate, larger faster rates. -void SoundTouch::setRate(double newRate) -{ - virtualRate = newRate; - calcEffectiveRateAndTempo(); -} - - -// Sets new rate control value as a difference in percents compared -// to the original rate (-50 .. +100 %) -void SoundTouch::setRateChange(double newRate) -{ - virtualRate = 1.0 + 0.01 * newRate; - calcEffectiveRateAndTempo(); -} - - -// Sets new tempo control value. Normal tempo = 1.0, smaller values -// represent slower tempo, larger faster tempo. -void SoundTouch::setTempo(double newTempo) -{ - virtualTempo = newTempo; - calcEffectiveRateAndTempo(); -} - - -// Sets new tempo control value as a difference in percents compared -// to the original tempo (-50 .. +100 %) -void SoundTouch::setTempoChange(double newTempo) -{ - virtualTempo = 1.0 + 0.01 * newTempo; - calcEffectiveRateAndTempo(); -} - - -// Sets new pitch control value. Original pitch = 1.0, smaller values -// represent lower pitches, larger values higher pitch. -void SoundTouch::setPitch(double newPitch) -{ - virtualPitch = newPitch; - calcEffectiveRateAndTempo(); -} - - -// Sets pitch change in octaves compared to the original pitch -// (-1.00 .. +1.00) -void SoundTouch::setPitchOctaves(double newPitch) -{ - virtualPitch = exp(0.69314718056 * newPitch); - calcEffectiveRateAndTempo(); -} - - -// Sets pitch change in semi-tones compared to the original pitch -// (-12 .. +12) -void SoundTouch::setPitchSemiTones(int newPitch) -{ - setPitchOctaves((double)newPitch / 12.0); -} - - -void SoundTouch::setPitchSemiTones(double newPitch) -{ - setPitchOctaves(newPitch / 12.0); -} - - -// Calculates 'effective' rate and tempo values from the -// nominal control values. -void SoundTouch::calcEffectiveRateAndTempo() -{ - double oldTempo = tempo; - double oldRate = rate; - - tempo = virtualTempo / virtualPitch; - rate = virtualPitch * virtualRate; - - if (!TEST_FLOAT_EQUAL(rate,oldRate)) pRateTransposer->setRate(rate); - if (!TEST_FLOAT_EQUAL(tempo, oldTempo)) pTDStretch->setTempo(tempo); - -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - if (rate <= 1.0f) - { - if (output != pTDStretch) - { - FIFOSamplePipe *tempoOut; - - assert(output == pRateTransposer); - // move samples in the current output buffer to the output of pTDStretch - tempoOut = pTDStretch->getOutput(); - tempoOut->moveSamples(*output); - // move samples in pitch transposer's store buffer to tempo changer's input - // deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore()); - - output = pTDStretch; - } - } - else -#endif - { - if (output != pRateTransposer) - { - FIFOSamplePipe *transOut; - - assert(output == pTDStretch); - // move samples in the current output buffer to the output of pRateTransposer - transOut = pRateTransposer->getOutput(); - transOut->moveSamples(*output); - // move samples in tempo changer's input to pitch transposer's input - pRateTransposer->moveSamples(*pTDStretch->getInput()); - - output = pRateTransposer; - } - } -} - - -// Sets sample rate. -void SoundTouch::setSampleRate(uint srate) -{ - // set sample rate, leave other tempo changer parameters as they are. - pTDStretch->setParameters((int)srate); - bSrateSet = true; -} - - -// Adds 'numSamples' pcs of samples from the 'samples' memory position into -// the input of the object. -void SoundTouch::putSamples(const SAMPLETYPE *samples, uint nSamples) -{ - if (bSrateSet == false) - { - ST_THROW_RT_ERROR("SoundTouch : Sample rate not defined"); - } - else if (channels == 0) - { - ST_THROW_RT_ERROR("SoundTouch : Number of channels not defined"); - } - - // accumulate how many samples are expected out from processing, given the current - // processing setting - samplesExpectedOut += (double)nSamples / ((double)rate * (double)tempo); - -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - if (rate <= 1.0f) - { - // transpose the rate down, output the transposed sound to tempo changer buffer - assert(output == pTDStretch); - pRateTransposer->putSamples(samples, nSamples); - pTDStretch->moveSamples(*pRateTransposer); - } - else -#endif - { - // evaluate the tempo changer, then transpose the rate up, - assert(output == pRateTransposer); - pTDStretch->putSamples(samples, nSamples); - pRateTransposer->moveSamples(*pTDStretch); - } -} - - -// Flushes the last samples from the processing pipeline to the output. -// Clears also the internal processing buffers. -// -// Note: This function is meant for extracting the last samples of a sound -// stream. This function may introduce additional blank samples in the end -// of the sound stream, and thus it's not recommended to call this function -// in the middle of a sound stream. -void SoundTouch::flush() -{ - int i; - int numStillExpected; - SAMPLETYPE *buff = new SAMPLETYPE[128 * channels]; - - // how many samples are still expected to output - numStillExpected = (int)((long)(samplesExpectedOut + 0.5) - samplesOutput); - if (numStillExpected < 0) numStillExpected = 0; - - memset(buff, 0, 128 * channels * sizeof(SAMPLETYPE)); - // "Push" the last active samples out from the processing pipeline by - // feeding blank samples into the processing pipeline until new, - // processed samples appear in the output (not however, more than - // 24ksamples in any case) - for (i = 0; (numStillExpected > (int)numSamples()) && (i < 200); i ++) - { - putSamples(buff, 128); - } - - adjustAmountOfSamples(numStillExpected); - - delete[] buff; - - // Clear input buffers - pTDStretch->clearInput(); - // yet leave the output intouched as that's where the - // flushed samples are! -} - - -// Changes a setting controlling the processing system behaviour. See the -// 'SETTING_...' defines for available setting ID's. -bool SoundTouch::setSetting(int settingId, int value) -{ - int sampleRate, sequenceMs, seekWindowMs, overlapMs; - - // read current tdstretch routine parameters - pTDStretch->getParameters(&sampleRate, &sequenceMs, &seekWindowMs, &overlapMs); - - switch (settingId) - { - case SETTING_USE_AA_FILTER : - // enables / disabless anti-alias filter - pRateTransposer->enableAAFilter((value != 0) ? true : false); - return true; - - case SETTING_AA_FILTER_LENGTH : - // sets anti-alias filter length - pRateTransposer->getAAFilter()->setLength(value); - return true; - - case SETTING_USE_QUICKSEEK : - // enables / disables tempo routine quick seeking algorithm - pTDStretch->enableQuickSeek((value != 0) ? true : false); - return true; - - case SETTING_SEQUENCE_MS: - // change time-stretch sequence duration parameter - pTDStretch->setParameters(sampleRate, value, seekWindowMs, overlapMs); - return true; - - case SETTING_SEEKWINDOW_MS: - // change time-stretch seek window length parameter - pTDStretch->setParameters(sampleRate, sequenceMs, value, overlapMs); - return true; - - case SETTING_OVERLAP_MS: - // change time-stretch overlap length parameter - pTDStretch->setParameters(sampleRate, sequenceMs, seekWindowMs, value); - return true; - - default : - return false; - } -} - - -// Reads a setting controlling the processing system behaviour. See the -// 'SETTING_...' defines for available setting ID's. -// -// Returns the setting value. -int SoundTouch::getSetting(int settingId) const -{ - int temp; - - switch (settingId) - { - case SETTING_USE_AA_FILTER : - return (uint)pRateTransposer->isAAFilterEnabled(); - - case SETTING_AA_FILTER_LENGTH : - return pRateTransposer->getAAFilter()->getLength(); - - case SETTING_USE_QUICKSEEK : - return (uint)pTDStretch->isQuickSeekEnabled(); - - case SETTING_SEQUENCE_MS: - pTDStretch->getParameters(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 : - { - int size = pTDStretch->getInputSampleReq(); - -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - if (rate <= 1.0) - { - // transposing done before timestretch, which impacts latency - return (int)(size * rate + 0.5); - } -#endif - return size; - } - - case SETTING_NOMINAL_OUTPUT_SEQUENCE : - { - int size = pTDStretch->getOutputBatchSize(); - - if (rate > 1.0) - { - // transposing done after timestretch, which impacts latency - return (int)(size / rate + 0.5); - } - return size; - } - - case SETTING_INITIAL_LATENCY: - { - double latency = pTDStretch->getLatency(); - int latency_tr = pRateTransposer->getLatency(); - -#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER - if (rate <= 1.0) - { - // transposing done before timestretch, which impacts latency - latency = (latency + latency_tr) * rate; - } - else -#endif - { - latency += (double)latency_tr / rate; - } - - return (int)(latency + 0.5); - } - - default : - return 0; - } -} - - -// Clears all the samples in the object's output and internal processing -// buffers. -void SoundTouch::clear() -{ - samplesExpectedOut = 0; - samplesOutput = 0; - pRateTransposer->clear(); - pTDStretch->clear(); -} - - -/// Returns number of samples currently unprocessed. -uint SoundTouch::numUnprocessedSamples() const -{ - FIFOSamplePipe * psp; - if (pTDStretch) - { - psp = pTDStretch->getInput(); - if (psp) - { - return psp->numSamples(); - } - } - return 0; -} - - -/// Output samples from beginning of the sample buffer. Copies requested samples to -/// output buffer and removes them from the sample buffer. If there are less than -/// 'numsample' samples in the buffer, returns all that available. -/// -/// \return Number of samples returned. -uint SoundTouch::receiveSamples(SAMPLETYPE *output, uint maxSamples) -{ - uint ret = FIFOProcessor::receiveSamples(output, maxSamples); - samplesOutput += (long)ret; - return ret; -} - - -/// Adjusts book-keeping so that given number of samples are removed from beginning of the -/// sample buffer without copying them anywhere. -/// -/// Used to reduce the number of samples in the buffer when accessing the sample buffer directly -/// with 'ptrBegin' function. -uint SoundTouch::receiveSamples(uint maxSamples) -{ - uint ret = FIFOProcessor::receiveSamples(maxSamples); - samplesOutput += (long)ret; - return ret; -} - - -/// Get ratio between input and output audio durations, useful for calculating -/// processed output duration: if you'll process a stream of N samples, then -/// you can expect to get out N * getInputOutputSampleRatio() samples. -double SoundTouch::getInputOutputSampleRatio() -{ - return 1.0 / (tempo * rate); -} diff --git a/lib/soundtouch/SoundTouch.h b/lib/soundtouch/SoundTouch.h deleted file mode 100644 index f2addc1b0ae..00000000000 --- a/lib/soundtouch/SoundTouch.h +++ /dev/null @@ -1,348 +0,0 @@ -////////////////////////////////////////////////////////////////////////////// -/// -/// SoundTouch - main class for tempo/pitch/rate adjusting routines. -/// -/// Notes: -/// - Initialize the SoundTouch object instance by setting up the sound stream -/// parameters with functions 'setSampleRate' and 'setChannels', then set -/// desired tempo/pitch/rate settings with the corresponding functions. -/// -/// - The SoundTouch class behaves like a first-in-first-out pipeline: The -/// samples that are to be processed are fed into one of the pipe by calling -/// function 'putSamples', while the ready processed samples can be read -/// from the other end of the pipeline with function 'receiveSamples'. -/// -/// - The SoundTouch processing classes require certain sized 'batches' of -/// samples in order to process the sound. For this reason the classes buffer -/// incoming samples until there are enough of samples available for -/// processing, then they carry out the processing step and consequently -/// make the processed samples available for outputting. -/// -/// - For the above reason, the processing routines introduce a certain -/// 'latency' between the input and output, so that the samples input to -/// SoundTouch may not be immediately available in the output, and neither -/// the amount of outputtable samples may not immediately be in direct -/// relationship with the amount of previously input samples. -/// -/// - The tempo/pitch/rate control parameters can be altered during processing. -/// Please notice though that they aren't currently protected by semaphores, -/// so in multi-thread application external semaphore protection may be -/// required. -/// -/// - This class utilizes classes 'TDStretch' for tempo change (without modifying -/// pitch) and 'RateTransposer' for changing the playback rate (that is, both -/// tempo and pitch in the same ratio) of the sound. The third available control -/// 'pitch' (change pitch but maintain tempo) is produced by a combination of -/// combining the two other controls. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef SoundTouch_H -#define SoundTouch_H - -#include "FIFOSamplePipe.h" -#include "STTypes.h" - -namespace soundtouch -{ - -/// Soundtouch library version string -#define SOUNDTOUCH_VERSION "2.1" - -/// SoundTouch library version id -#define SOUNDTOUCH_VERSION_ID (20100) - -// -// Available setting IDs for the 'setSetting' & 'get_setting' functions: - -/// Enable/disable anti-alias filter in pitch transposer (0 = disable) -#define SETTING_USE_AA_FILTER 0 - -/// Pitch transposer anti-alias filter length (8 .. 128 taps, default = 32) -#define SETTING_AA_FILTER_LENGTH 1 - -/// Enable/disable quick seeking algorithm in tempo changer routine -/// (enabling quick seeking lowers CPU utilization but causes a minor sound -/// quality compromising) -#define SETTING_USE_QUICKSEEK 2 - -/// Time-stretch algorithm single processing sequence length in milliseconds. This determines -/// to how long sequences the original sound is chopped in the time-stretch algorithm. -/// See "STTypes.h" or README for more information. -#define SETTING_SEQUENCE_MS 3 - -/// Time-stretch algorithm seeking window length in milliseconds for algorithm that finds the -/// best possible overlapping location. This determines from how wide window the algorithm -/// may look for an optimal joining location when mixing the sound sequences back together. -/// See "STTypes.h" or README for more information. -#define SETTING_SEEKWINDOW_MS 4 - -/// Time-stretch algorithm overlap length in milliseconds. When the chopped sound sequences -/// are mixed back together, to form a continuous sound stream, this parameter defines over -/// how long period the two consecutive sequences are let to overlap each other. -/// See "STTypes.h" or README for more information. -#define SETTING_OVERLAP_MS 5 - - -/// Call "getSetting" with this ID to query processing sequence size in samples. -/// This value gives approximate value of how many input samples you'll need to -/// feed into SoundTouch after initial buffering to get out a new batch of -/// output samples. -/// -/// This value does not include initial buffering at beginning of a new processing -/// stream, use SETTING_INITIAL_LATENCY to get the initial buffering size. -/// -/// Notices: -/// - This is read-only parameter, i.e. setSetting ignores this parameter -/// - This parameter value is not constant but change depending on -/// tempo/pitch/rate/samplerate settings. -#define SETTING_NOMINAL_INPUT_SEQUENCE 6 - - -/// Call "getSetting" with this ID to query nominal average processing output -/// size in samples. This value tells approcimate value how many output samples -/// SoundTouch outputs once it does DSP processing run for a batch of input samples. -/// -/// Notices: -/// - This is read-only parameter, i.e. setSetting ignores this parameter -/// - This parameter value is not constant but change depending on -/// tempo/pitch/rate/samplerate settings. -#define SETTING_NOMINAL_OUTPUT_SEQUENCE 7 - - -/// Call "getSetting" with this ID to query initial processing latency, i.e. -/// approx. how many samples you'll need to enter to SoundTouch pipeline before -/// you can expect to get first batch of ready output samples out. -/// -/// After the first output batch, you can then expect to get approx. -/// SETTING_NOMINAL_OUTPUT_SEQUENCE ready samples out for every -/// SETTING_NOMINAL_INPUT_SEQUENCE samples that you enter into SoundTouch. -/// -/// Example: -/// processing with parameter -tempo=5 -/// => initial latency = 5509 samples -/// input sequence = 4167 samples -/// output sequence = 3969 samples -/// -/// Accordingly, you can expect to feed in approx. 5509 samples at beginning of -/// the stream, and then you'll get out the first 3969 samples. After that, for -/// every approx. 4167 samples that you'll put in, you'll receive again approx. -/// 3969 samples out. -/// -/// This also means that average latency during stream processing is -/// INITIAL_LATENCY-OUTPUT_SEQUENCE/2, in the above example case 5509-3969/2 -/// = 3524 samples -/// -/// Notices: -/// - This is read-only parameter, i.e. setSetting ignores this parameter -/// - This parameter value is not constant but change depending on -/// tempo/pitch/rate/samplerate settings. -#define SETTING_INITIAL_LATENCY 8 - - -class SoundTouch : public FIFOProcessor -{ -private: - /// Rate transposer class instance - class RateTransposer *pRateTransposer; - - /// Time-stretch class instance - class TDStretch *pTDStretch; - - /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters. - double virtualRate; - - /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters. - double virtualTempo; - - /// Virtual pitch parameter. Effective rate & tempo are calculated from these parameters. - double virtualPitch; - - /// Flag: Has sample rate been set? - bool bSrateSet; - - /// Accumulator for how many samples in total will be expected as output vs. samples put in, - /// considering current processing settings. - double samplesExpectedOut; - - /// Accumulator for how many samples in total have been read out from the processing so far - long samplesOutput; - - /// Calculates effective rate & tempo valuescfrom 'virtualRate', 'virtualTempo' and - /// 'virtualPitch' parameters. - void calcEffectiveRateAndTempo(); - -protected : - /// Number of channels - uint channels; - - /// Effective 'rate' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch' - double rate; - - /// Effective 'tempo' value calculated from 'virtualRate', 'virtualTempo' and 'virtualPitch' - double tempo; - -public: - SoundTouch(); - virtual ~SoundTouch(); - - /// Get SoundTouch library version string - static const char *getVersionString(); - - /// Get SoundTouch library version Id - static uint getVersionId(); - - /// Sets new rate control value. Normal rate = 1.0, smaller values - /// represent slower rate, larger faster rates. - void setRate(double newRate); - - /// Sets new tempo control value. Normal tempo = 1.0, smaller values - /// represent slower tempo, larger faster tempo. - void setTempo(double newTempo); - - /// Sets new rate control value as a difference in percents compared - /// to the original rate (-50 .. +100 %) - void setRateChange(double newRate); - - /// Sets new tempo control value as a difference in percents compared - /// to the original tempo (-50 .. +100 %) - void setTempoChange(double newTempo); - - /// Sets new pitch control value. Original pitch = 1.0, smaller values - /// represent lower pitches, larger values higher pitch. - void setPitch(double newPitch); - - /// Sets pitch change in octaves compared to the original pitch - /// (-1.00 .. +1.00) - void setPitchOctaves(double newPitch); - - /// Sets pitch change in semi-tones compared to the original pitch - /// (-12 .. +12) - void setPitchSemiTones(int newPitch); - void setPitchSemiTones(double newPitch); - - /// Sets the number of channels, 1 = mono, 2 = stereo - void setChannels(uint numChannels); - - /// Sets sample rate. - void setSampleRate(uint srate); - - /// Get ratio between input and output audio durations, useful for calculating - /// processed output duration: if you'll process a stream of N samples, then - /// you can expect to get out N * getInputOutputSampleRatio() samples. - /// - /// This ratio will give accurate target duration ratio for a full audio track, - /// given that the the whole track is processed with same processing parameters. - /// - /// If this ratio is applied to calculate intermediate offsets inside a processing - /// stream, then this ratio is approximate and can deviate +- some tens of milliseconds - /// from ideal offset, yet by end of the audio stream the duration ratio will become - /// exact. - /// - /// Example: if processing with parameters "-tempo=15 -pitch=-3", the function - /// will return value 0.8695652... Now, if processing an audio stream whose duration - /// is exactly one million audio samples, then you can expect the processed - /// output duration be 0.869565 * 1000000 = 869565 samples. - double getInputOutputSampleRatio(); - - /// Flushes the last samples from the processing pipeline to the output. - /// Clears also the internal processing buffers. - // - /// Note: This function is meant for extracting the last samples of a sound - /// stream. This function may introduce additional blank samples in the end - /// of the sound stream, and thus it's not recommended to call this function - /// in the middle of a sound stream. - void flush(); - - /// Adds 'numSamples' pcs of samples from the 'samples' memory position into - /// the input of the object. Notice that sample rate _has_to_ be set before - /// calling this function, otherwise throws a runtime_error exception. - virtual void putSamples( - const SAMPLETYPE *samples, ///< Pointer to sample buffer. - uint numSamples ///< Number of samples in buffer. Notice - ///< that in case of stereo-sound a single sample - ///< contains data for both channels. - ); - - /// 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 successfully changed - bool setSetting(int settingId, ///< Setting ID number. see SETTING_... defines. - int value ///< New setting value. - ); - - /// Reads a setting controlling the processing system behaviour. See the - /// 'SETTING_...' defines for available setting ID's. - /// - /// \return the setting value. - int getSetting(int settingId ///< Setting ID number, see SETTING_... defines. - ) const; - - /// Returns number of samples currently unprocessed. - virtual uint numUnprocessedSamples() const; - - /// Return number of channels - uint numChannels() const - { - return channels; - } - - /// Other handy functions that are implemented in the ancestor classes (see - /// classes 'FIFOProcessor' and 'FIFOSamplePipe') - /// - /// - receiveSamples() : Use this function to receive 'ready' processed samples from SoundTouch. - /// - numSamples() : Get number of 'ready' samples that can be received with - /// function 'receiveSamples()' - /// - isEmpty() : Returns nonzero if there aren't any 'ready' samples. - /// - clear() : Clears all samples from ready/processing buffers. -}; - -} -#endif diff --git a/lib/soundtouch/TDStretch.cpp b/lib/soundtouch/TDStretch.cpp deleted file mode 100644 index ce49310900f..00000000000 --- a/lib/soundtouch/TDStretch.cpp +++ /dev/null @@ -1,1111 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo -/// while maintaining the original pitch by using a time domain WSOLA-like -/// method with several performance-increasing tweaks. -/// -/// Notes : MMX optimized functions reside in a separate, platform-specific -/// file, e.g. 'mmx_win.cpp' or 'mmx_gcc.cpp'. -/// -/// This source file contains OpenMP optimizations that allow speeding up the -/// corss-correlation algorithm by executing it in several threads / CPU cores -/// in parallel. See the following article link for more detailed discussion -/// about SoundTouch OpenMP optimizations: -/// http://www.softwarecoven.com/parallel-computing-in-embedded-mobile-devices -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#include - -#include "STTypes.h" -#include "cpu_detect.h" -#include "TDStretch.h" - -using namespace soundtouch; - -#define max(x, y) (((x) > (y)) ? (x) : (y)) - - -/***************************************************************************** - * - * Constant definitions - * - *****************************************************************************/ - -// Table for the hierarchical mixing position seeking algorithm -const short _scanOffsets[5][24]={ - { 124, 186, 248, 310, 372, 434, 496, 558, 620, 682, 744, 806, - 868, 930, 992, 1054, 1116, 1178, 1240, 1302, 1364, 1426, 1488, 0}, - {-100, -75, -50, -25, 25, 50, 75, 100, 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}, - { -20, -15, -10, -5, 5, 10, 15, 20, 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}, - { -4, -3, -2, -1, 1, 2, 3, 4, 0, 0, 0, 0, - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}, - { 121, 114, 97, 114, 98, 105, 108, 32, 104, 99, 117, 111, - 116, 100, 110, 117, 111, 115, 0, 0, 0, 0, 0, 0}}; - -/***************************************************************************** - * - * Implementation of the class 'TDStretch' - * - *****************************************************************************/ - - -TDStretch::TDStretch() : FIFOProcessor(&outputBuffer) -{ - bQuickSeek = false; - channels = 2; - - pMidBuffer = NULL; - pMidBufferUnaligned = NULL; - overlapLength = 0; - - bAutoSeqSetting = true; - bAutoSeekSetting = true; - - maxnorm = 0; - maxnormf = 1e8; - - skipFract = 0; - - tempo = 1.0f; - setParameters(44100, DEFAULT_SEQUENCE_MS, DEFAULT_SEEKWINDOW_MS, DEFAULT_OVERLAP_MS); - setTempo(1.0f); - - clear(); -} - - - -TDStretch::~TDStretch() -{ - delete[] pMidBufferUnaligned; -} - - - -// 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 (default = 82 ms) -// 'seekwindowMS' = seeking window length for scanning the best overlapping -// position (default = 28 ms) -// 'overlapMS' = overlapping length (default = 12 ms) - -void TDStretch::setParameters(int aSampleRate, int aSequenceMS, - int aSeekWindowMS, int aOverlapMS) -{ - // accept only positive parameter values - if zero or negative, use old values instead - if (aSampleRate > 0) - { - if (aSampleRate > 192000) ST_THROW_RT_ERROR("Error: Excessive samplerate"); - this->sampleRate = aSampleRate; - } - - if (aOverlapMS > 0) this->overlapMs = aOverlapMS; - - if (aSequenceMS > 0) - { - this->sequenceMs = aSequenceMS; - bAutoSeqSetting = false; - } - else if (aSequenceMS == 0) - { - // if zero, use automatic setting - bAutoSeqSetting = true; - } - - if (aSeekWindowMS > 0) - { - this->seekWindowMs = aSeekWindowMS; - bAutoSeekSetting = false; - } - else if (aSeekWindowMS == 0) - { - // if zero, use automatic setting - bAutoSeekSetting = true; - } - - calcSeqParameters(); - - calculateOverlapLength(overlapMs); - - // set tempo to recalculate 'sampleReq' - setTempo(tempo); -} - - - -/// 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 TDStretch::getParameters(int *pSampleRate, int *pSequenceMs, int *pSeekWindowMs, int *pOverlapMs) const -{ - if (pSampleRate) - { - *pSampleRate = sampleRate; - } - - if (pSequenceMs) - { - *pSequenceMs = (bAutoSeqSetting) ? (USE_AUTO_SEQUENCE_LEN) : sequenceMs; - } - - if (pSeekWindowMs) - { - *pSeekWindowMs = (bAutoSeekSetting) ? (USE_AUTO_SEEKWINDOW_LEN) : seekWindowMs; - } - - if (pOverlapMs) - { - *pOverlapMs = overlapMs; - } -} - - -// Overlaps samples in 'midBuffer' with the samples in 'pInput' -void TDStretch::overlapMono(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput) const -{ - int i; - SAMPLETYPE m1, m2; - - m1 = (SAMPLETYPE)0; - m2 = (SAMPLETYPE)overlapLength; - - for (i = 0; i < overlapLength ; i ++) - { - pOutput[i] = (pInput[i] * m1 + pMidBuffer[i] * m2 ) / overlapLength; - m1 += 1; - m2 -= 1; - } -} - - - -void TDStretch::clearMidBuffer() -{ - memset(pMidBuffer, 0, channels * sizeof(SAMPLETYPE) * overlapLength); -} - - -void TDStretch::clearInput() -{ - inputBuffer.clear(); - clearMidBuffer(); - isBeginning = true; -} - - -// Clears the sample buffers -void TDStretch::clear() -{ - outputBuffer.clear(); - clearInput(); -} - - - -// Enables/disables the quick position seeking algorithm. Zero to disable, nonzero -// to enable -void TDStretch::enableQuickSeek(bool enable) -{ - bQuickSeek = enable; -} - - -// Returns nonzero if the quick seeking algorithm is enabled. -bool TDStretch::isQuickSeekEnabled() const -{ - return bQuickSeek; -} - - -// Seeks for the optimal overlap-mixing position. -int TDStretch::seekBestOverlapPosition(const SAMPLETYPE *refPos) -{ - if (bQuickSeek) - { - return seekBestOverlapPositionQuick(refPos); - } - else - { - return seekBestOverlapPositionFull(refPos); - } -} - - -// Overlaps samples in 'midBuffer' with the samples in 'pInputBuffer' at position -// of 'ovlPos'. -inline void TDStretch::overlap(SAMPLETYPE *pOutput, const SAMPLETYPE *pInput, uint ovlPos) const -{ -#ifndef USE_MULTICH_ALWAYS - if (channels == 1) - { - // mono sound. - overlapMono(pOutput, pInput + ovlPos); - } - else if (channels == 2) - { - // stereo sound - overlapStereo(pOutput, pInput + 2 * ovlPos); - } - else -#endif // USE_MULTICH_ALWAYS - { - assert(channels > 0); - overlapMulti(pOutput, pInput + channels * ovlPos); - } -} - - -// Seeks for the optimal overlap-mixing position. The 'stereo' version of the -// routine -// -// The best position is determined as the position where the two overlapped -// sample sequences are 'most alike', in terms of the highest cross-correlation -// value over the overlapping period -int TDStretch::seekBestOverlapPositionFull(const SAMPLETYPE *refPos) -{ - int bestOffs; - double bestCorr; - int i; - double norm; - - bestCorr = -FLT_MAX; - bestOffs = 0; - - // Scans for the best correlation value by testing each possible position - // over the permitted range. - bestCorr = calcCrossCorr(refPos, pMidBuffer, norm); - bestCorr = (bestCorr + 0.1) * 0.75; - - #pragma omp parallel for - for (i = 1; i < seekLength; i ++) - { - double corr; - // Calculates correlation value for the mixing position corresponding to 'i' -#ifdef _OPENMP - // in parallel OpenMP mode, can't use norm accumulator version as parallel executor won't - // iterate the loop in sequential order - corr = calcCrossCorr(refPos + channels * i, pMidBuffer, norm); -#else - // In non-parallel version call "calcCrossCorrAccumulate" that is otherwise same - // as "calcCrossCorr", but saves time by reusing & updating previously stored - // "norm" value - corr = calcCrossCorrAccumulate(refPos + channels * i, pMidBuffer, norm); -#endif - // heuristic rule to slightly favour values close to mid of the range - double tmp = (double)(2 * i - seekLength) / (double)seekLength; - corr = ((corr + 0.1) * (1.0 - 0.25 * tmp * tmp)); - - // Checks for the highest correlation value - if (corr > bestCorr) - { - // For optimal performance, enter critical section only in case that best value found. - // in such case repeat 'if' condition as it's possible that parallel execution may have - // updated the bestCorr value in the mean time - #pragma omp critical - if (corr > bestCorr) - { - bestCorr = corr; - bestOffs = i; - } - } - } - -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - adaptNormalizer(); -#endif - - // clear cross correlation routine state if necessary (is so e.g. in MMX routines). - clearCrossCorrState(); - - return bestOffs; -} - - -// Quick seek algorithm for improved runtime-performance: First roughly scans through the -// correlation area, and then scan surroundings of two best preliminary correlation candidates -// with improved precision -// -// Based on testing: -// - This algorithm gives on average 99% as good match as the full algorithm -// - this quick seek algorithm finds the best match on ~90% of cases -// - on those 10% of cases when this algorithm doesn't find best match, -// it still finds on average ~90% match vs. the best possible match -int TDStretch::seekBestOverlapPositionQuick(const SAMPLETYPE *refPos) -{ -#define _MIN(a, b) (((a) < (b)) ? (a) : (b)) -#define SCANSTEP 16 -#define SCANWIND 8 - - int bestOffs; - int i; - int bestOffs2; - float bestCorr, corr; - float bestCorr2; - double norm; - - // note: 'float' types used in this function in case that the platform would need to use software-fp - - bestCorr = - bestCorr2 = -FLT_MAX; - bestOffs = - bestOffs2 = SCANWIND; - - // Scans for the best correlation value by testing each possible position - // over the permitted range. Look for two best matches on the first pass to - // increase possibility of ideal match. - // - // Begin from "SCANSTEP" instead of SCANWIND to make the calculation - // catch the 'middlepoint' of seekLength vector as that's the a-priori - // expected best match position - // - // Roughly: - // - 15% of cases find best result directly on the first round, - // - 75% cases find better match on 2nd round around the best match from 1st round - // - 10% cases find better match on 2nd round around the 2nd-best-match from 1st round - for (i = SCANSTEP; i < seekLength - SCANWIND - 1; i += SCANSTEP) - { - // Calculates correlation value for the mixing position corresponding - // to 'i' - corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm); - // heuristic rule to slightly favour values close to mid of the seek range - float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength; - corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp)); - - // Checks for the highest correlation value - if (corr > bestCorr) - { - // found new best match. keep the previous best as 2nd best match - bestCorr2 = bestCorr; - bestOffs2 = bestOffs; - bestCorr = corr; - bestOffs = i; - } - else if (corr > bestCorr2) - { - // not new best, but still new 2nd best match - bestCorr2 = corr; - bestOffs2 = i; - } - } - - // Scans surroundings of the found best match with small stepping - int end = _MIN(bestOffs + SCANWIND + 1, seekLength); - for (i = bestOffs - SCANWIND; i < end; i++) - { - if (i == bestOffs) continue; // this offset already calculated, thus skip - - // Calculates correlation value for the mixing position corresponding - // to 'i' - corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm); - // heuristic rule to slightly favour values close to mid of the range - float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength; - corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp)); - - // Checks for the highest correlation value - if (corr > bestCorr) - { - bestCorr = corr; - bestOffs = i; - } - } - - // Scans surroundings of the 2nd best match with small stepping - end = _MIN(bestOffs2 + SCANWIND + 1, seekLength); - for (i = bestOffs2 - SCANWIND; i < end; i++) - { - if (i == bestOffs2) continue; // this offset already calculated, thus skip - - // Calculates correlation value for the mixing position corresponding - // to 'i' - corr = (float)calcCrossCorr(refPos + channels*i, pMidBuffer, norm); - // heuristic rule to slightly favour values close to mid of the range - float tmp = (float)(2 * i - seekLength - 1) / (float)seekLength; - corr = ((corr + 0.1f) * (1.0f - 0.25f * tmp * tmp)); - - // Checks for the highest correlation value - if (corr > bestCorr) - { - bestCorr = corr; - bestOffs = i; - } - } - - // clear cross correlation routine state if necessary (is so e.g. in MMX routines). - clearCrossCorrState(); - -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - adaptNormalizer(); -#endif - - return bestOffs; -} - - - - -/// For integer algorithm: adapt normalization factor divider with music so that -/// it'll not be pessimistically restrictive that can degrade quality on quieter sections -/// yet won't cause integer overflows either -void TDStretch::adaptNormalizer() -{ - // Do not adapt normalizer over too silent sequences to avoid averaging filter depleting to - // too low values during pauses in music - if ((maxnorm > 1000) || (maxnormf > 40000000)) - { - //norm averaging filter - maxnormf = 0.9f * maxnormf + 0.1f * (float)maxnorm; - - if ((maxnorm > 800000000) && (overlapDividerBitsNorm < 16)) - { - // large values, so increase divider - overlapDividerBitsNorm++; - if (maxnorm > 1600000000) overlapDividerBitsNorm++; // extra large value => extra increase - } - else if ((maxnormf < 1000000) && (overlapDividerBitsNorm > 0)) - { - // extra small values, decrease divider - overlapDividerBitsNorm--; - } - } - - maxnorm = 0; -} - - -/// clear cross correlation routine state if necessary -void TDStretch::clearCrossCorrState() -{ - // default implementation is empty. -} - - -/// Calculates processing sequence length according to tempo setting -void TDStretch::calcSeqParameters() -{ - // Adjust tempo param according to tempo, so that variating processing sequence length is used - // at various tempo settings, between the given low...top limits - #define AUTOSEQ_TEMPO_LOW 0.5 // auto setting low tempo range (-50%) - #define AUTOSEQ_TEMPO_TOP 2.0 // auto setting top tempo range (+100%) - - // sequence-ms setting values at above low & top tempo - #define AUTOSEQ_AT_MIN 90.0 - #define AUTOSEQ_AT_MAX 40.0 - #define AUTOSEQ_K ((AUTOSEQ_AT_MAX - AUTOSEQ_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW)) - #define AUTOSEQ_C (AUTOSEQ_AT_MIN - (AUTOSEQ_K) * (AUTOSEQ_TEMPO_LOW)) - - // seek-window-ms setting values at above low & top tempoq - #define AUTOSEEK_AT_MIN 20.0 - #define AUTOSEEK_AT_MAX 15.0 - #define AUTOSEEK_K ((AUTOSEEK_AT_MAX - AUTOSEEK_AT_MIN) / (AUTOSEQ_TEMPO_TOP - AUTOSEQ_TEMPO_LOW)) - #define AUTOSEEK_C (AUTOSEEK_AT_MIN - (AUTOSEEK_K) * (AUTOSEQ_TEMPO_LOW)) - - #define CHECK_LIMITS(x, mi, ma) (((x) < (mi)) ? (mi) : (((x) > (ma)) ? (ma) : (x))) - - double seq, seek; - - if (bAutoSeqSetting) - { - seq = AUTOSEQ_C + AUTOSEQ_K * tempo; - seq = CHECK_LIMITS(seq, AUTOSEQ_AT_MAX, AUTOSEQ_AT_MIN); - sequenceMs = (int)(seq + 0.5); - } - - if (bAutoSeekSetting) - { - seek = AUTOSEEK_C + AUTOSEEK_K * tempo; - seek = CHECK_LIMITS(seek, AUTOSEEK_AT_MAX, AUTOSEEK_AT_MIN); - seekWindowMs = (int)(seek + 0.5); - } - - // Update seek window lengths - seekWindowLength = (sampleRate * sequenceMs) / 1000; - if (seekWindowLength < 2 * overlapLength) - { - seekWindowLength = 2 * overlapLength; - } - seekLength = (sampleRate * seekWindowMs) / 1000; -} - - - -// Sets new target tempo. Normal tempo = 'SCALE', smaller values represent slower -// tempo, larger faster tempo. -void TDStretch::setTempo(double newTempo) -{ - int intskip; - - tempo = newTempo; - - // Calculate new sequence duration - calcSeqParameters(); - - // Calculate ideal skip length (according to tempo value) - nominalSkip = tempo * (seekWindowLength - overlapLength); - intskip = (int)(nominalSkip + 0.5); - - // Calculate how many samples are needed in the 'inputBuffer' to - // process another batch of samples - //sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength / 2; - sampleReq = max(intskip + overlapLength, seekWindowLength) + seekLength; -} - - - -// Sets the number of channels, 1 = mono, 2 = stereo -void TDStretch::setChannels(int numChannels) -{ - if (!verifyNumberOfChannels(numChannels) || - (channels == numChannels)) return; - - channels = numChannels; - inputBuffer.setChannels(channels); - outputBuffer.setChannels(channels); - - // re-init overlap/buffer - overlapLength=0; - setParameters(sampleRate); -} - - -// nominal tempo, no need for processing, just pass the samples through -// to outputBuffer -/* -void TDStretch::processNominalTempo() -{ - assert(tempo == 1.0f); - - if (bMidBufferDirty) - { - // If there are samples in pMidBuffer waiting for overlapping, - // do a single sliding overlapping with them in order to prevent a - // clicking distortion in the output sound - if (inputBuffer.numSamples() < overlapLength) - { - // wait until we've got overlapLength input samples - return; - } - // Mix the samples in the beginning of 'inputBuffer' with the - // samples in 'midBuffer' using sliding overlapping - overlap(outputBuffer.ptrEnd(overlapLength), inputBuffer.ptrBegin(), 0); - outputBuffer.putSamples(overlapLength); - inputBuffer.receiveSamples(overlapLength); - clearMidBuffer(); - // now we've caught the nominal sample flow and may switch to - // bypass mode - } - - // Simply bypass samples from input to output - outputBuffer.moveSamples(inputBuffer); -} -*/ - - -// Processes as many processing frames of the samples 'inputBuffer', store -// the result into 'outputBuffer' -void TDStretch::processSamples() -{ - int ovlSkip; - int offset = 0; - int temp; - - /* Removed this small optimization - can introduce a click to sound when tempo setting - crosses the nominal value - if (tempo == 1.0f) - { - // tempo not changed from the original, so bypass the processing - processNominalTempo(); - return; - } - */ - - // Process samples as long as there are enough samples in 'inputBuffer' - // to form a processing frame. - while ((int)inputBuffer.numSamples() >= sampleReq) - { - if (isBeginning == false) - { - // apart from the very beginning of the track, - // scan for the best overlapping position & do overlap-add - offset = seekBestOverlapPosition(inputBuffer.ptrBegin()); - - // Mix the samples in the 'inputBuffer' at position of 'offset' with the - // samples in 'midBuffer' using sliding overlapping - // ... first partially overlap with the end of the previous sequence - // (that's in 'midBuffer') - overlap(outputBuffer.ptrEnd((uint)overlapLength), inputBuffer.ptrBegin(), (uint)offset); - outputBuffer.putSamples((uint)overlapLength); - offset += overlapLength; - } - else - { - // Adjust processing offset at beginning of track by not perform initial overlapping - // and compensating that in the 'input buffer skip' calculation - isBeginning = false; - int skip = (int)(tempo * overlapLength + 0.5); - - #ifdef SOUNDTOUCH_ALLOW_NONEXACT_SIMD_OPTIMIZATION - #ifdef SOUNDTOUCH_ALLOW_SSE - // if SSE mode, round the skip amount to value corresponding to aligned memory address - if (channels == 1) - { - skip &= -4; - } - else if (channels == 2) - { - skip &= -2; - } - #endif - #endif - skipFract -= skip; - assert(nominalSkip >= -skipFract); - } - - // ... then copy sequence samples from 'inputBuffer' to output: - - // crosscheck that we don't have buffer overflow... - if ((int)inputBuffer.numSamples() < (offset + seekWindowLength - overlapLength)) - { - continue; // just in case, shouldn't really happen - } - - // length of sequence - temp = (seekWindowLength - 2 * overlapLength); - outputBuffer.putSamples(inputBuffer.ptrBegin() + channels * offset, (uint)temp); - - // Copies the end of the current sequence from 'inputBuffer' to - // 'midBuffer' for being mixed with the beginning of the next - // processing sequence and so on - assert((offset + temp + overlapLength) <= (int)inputBuffer.numSamples()); - memcpy(pMidBuffer, inputBuffer.ptrBegin() + channels * (offset + temp), - channels * sizeof(SAMPLETYPE) * overlapLength); - - // Remove the processed samples from the input buffer. Update - // the difference between integer & nominal skip step to 'skipFract' - // in order to prevent the error from accumulating over time. - skipFract += nominalSkip; // real skip size - ovlSkip = (int)skipFract; // rounded to integer skip - skipFract -= ovlSkip; // maintain the fraction part, i.e. real vs. integer skip - inputBuffer.receiveSamples((uint)ovlSkip); - } -} - - -// Adds 'numsamples' pcs of samples from the 'samples' memory position into -// the input of the object. -void TDStretch::putSamples(const SAMPLETYPE *samples, uint nSamples) -{ - // Add the samples into the input buffer - inputBuffer.putSamples(samples, nSamples); - // Process the samples in input buffer - processSamples(); -} - - - -/// Set new overlap length parameter & reallocate RefMidBuffer if necessary. -void TDStretch::acceptNewOverlapLength(int newOverlapLength) -{ - int prevOvl; - - assert(newOverlapLength >= 0); - prevOvl = overlapLength; - overlapLength = newOverlapLength; - - if (overlapLength > prevOvl) - { - delete[] pMidBufferUnaligned; - - pMidBufferUnaligned = new SAMPLETYPE[overlapLength * channels + 16 / sizeof(SAMPLETYPE)]; - // ensure that 'pMidBuffer' is aligned to 16 byte boundary for efficiency - pMidBuffer = (SAMPLETYPE *)SOUNDTOUCH_ALIGN_POINTER_16(pMidBufferUnaligned); - - clearMidBuffer(); - } -} - - -// 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. -void * TDStretch::operator new(size_t s) -{ - // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead! - ST_THROW_RT_ERROR("Error in TDStretch::new: Don't use 'new TDStretch' directly, use 'newInstance' member instead!"); - return newInstance(); -} - - -TDStretch * TDStretch::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 TDStretchMMX; - } - else -#endif // SOUNDTOUCH_ALLOW_MMX - - -#ifdef SOUNDTOUCH_ALLOW_SSE - if (uExtensions & SUPPORT_SSE) - { - // SSE support - return ::new TDStretchSSE; - } - else -#endif // SOUNDTOUCH_ALLOW_SSE - - { - // ISA optimizations not supported, use plain C version - return ::new TDStretch; - } -} - - -////////////////////////////////////////////////////////////////////////////// -// -// Integer arithmetic specific algorithm implementations. -// -////////////////////////////////////////////////////////////////////////////// - -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - -// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Stereo' -// version of the routine. -void TDStretch::overlapStereo(short *poutput, const short *input) const -{ - int i; - short temp; - int cnt2; - - for (i = 0; i < overlapLength ; i ++) - { - temp = (short)(overlapLength - i); - cnt2 = 2 * i; - poutput[cnt2] = (input[cnt2] * i + pMidBuffer[cnt2] * temp ) / overlapLength; - poutput[cnt2 + 1] = (input[cnt2 + 1] * i + pMidBuffer[cnt2 + 1] * temp ) / overlapLength; - } -} - - -// Overlaps samples in 'midBuffer' with the samples in 'input'. The 'Multi' -// version of the routine. -void TDStretch::overlapMulti(SAMPLETYPE *poutput, const SAMPLETYPE *input) const -{ - SAMPLETYPE m1=(SAMPLETYPE)0; - SAMPLETYPE m2; - int i=0; - - for (m2 = (SAMPLETYPE)overlapLength; m2; m2 --) - { - for (int c = 0; c < channels; c ++) - { - poutput[i] = (input[i] * m1 + pMidBuffer[i] * m2) / overlapLength; - i++; - } - - m1++; - } -} - -// Calculates the x having the closest 2^x value for the given value -static int _getClosest2Power(double value) -{ - return (int)(log(value) / log(2.0) + 0.5); -} - - -/// Calculates overlap period length in samples. -/// Integer version rounds overlap length to closest power of 2 -/// for a divide scaling operation. -void TDStretch::calculateOverlapLength(int aoverlapMs) -{ - int newOvl; - - assert(aoverlapMs >= 0); - - // calculate overlap length so that it's power of 2 - thus it's easy to do - // integer division by right-shifting. Term "-1" at end is to account for - // the extra most significatnt bit left unused in result by signed multiplication - overlapDividerBitsPure = _getClosest2Power((sampleRate * aoverlapMs) / 1000.0) - 1; - if (overlapDividerBitsPure > 9) overlapDividerBitsPure = 9; - if (overlapDividerBitsPure < 3) overlapDividerBitsPure = 3; - newOvl = (int)pow(2.0, (int)overlapDividerBitsPure + 1); // +1 => account for -1 above - - acceptNewOverlapLength(newOvl); - - overlapDividerBitsNorm = overlapDividerBitsPure; - - // calculate sloping divider so that crosscorrelation operation won't - // overflow 32-bit register. Max. sum of the crosscorrelation sum without - // divider would be 2^30*(N^3-N)/3, where N = overlap length - slopingDivider = (newOvl * newOvl - 1) / 3; -} - - -double TDStretch::calcCrossCorr(const short *mixingPos, const short *compare, double &norm) -{ - long corr; - unsigned long lnorm; - int i; - - corr = lnorm = 0; - // Same routine for stereo and mono. For stereo, unroll loop for better - // efficiency and gives slightly better resolution against rounding. - // For mono it same routine, just unrolls loop by factor of 4 - for (i = 0; i < channels * overlapLength; i += 4) - { - corr += (mixingPos[i] * compare[i] + - mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm; // notice: do intermediate division here to avoid integer overflow - corr += (mixingPos[i + 2] * compare[i + 2] + - mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBitsNorm; - lnorm += (mixingPos[i] * mixingPos[i] + - mixingPos[i + 1] * mixingPos[i + 1]) >> overlapDividerBitsNorm; // notice: do intermediate division here to avoid integer overflow - lnorm += (mixingPos[i + 2] * mixingPos[i + 2] + - mixingPos[i + 3] * mixingPos[i + 3]) >> overlapDividerBitsNorm; - } - - if (lnorm > maxnorm) - { - // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode - #pragma omp critical - if (lnorm > maxnorm) - { - maxnorm = lnorm; - } - } - // Normalize result by dividing by sqrt(norm) - this step is easiest - // done using floating point operation - norm = (double)lnorm; - return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm); -} - - -/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value -double TDStretch::calcCrossCorrAccumulate(const short *mixingPos, const short *compare, double &norm) -{ - long corr; - unsigned long lnorm; - int i; - - // cancel first normalizer tap from previous round - lnorm = 0; - for (i = 1; i <= channels; i ++) - { - lnorm -= (mixingPos[-i] * mixingPos[-i]) >> overlapDividerBitsNorm; - } - - corr = 0; - // Same routine for stereo and mono. For stereo, unroll loop for better - // efficiency and gives slightly better resolution against rounding. - // For mono it same routine, just unrolls loop by factor of 4 - for (i = 0; i < channels * overlapLength; i += 4) - { - corr += (mixingPos[i] * compare[i] + - mixingPos[i + 1] * compare[i + 1]) >> overlapDividerBitsNorm; // notice: do intermediate division here to avoid integer overflow - corr += (mixingPos[i + 2] * compare[i + 2] + - mixingPos[i + 3] * compare[i + 3]) >> overlapDividerBitsNorm; - } - - // update normalizer with last samples of this round - for (int j = 0; j < channels; j ++) - { - i --; - lnorm += (mixingPos[i] * mixingPos[i]) >> overlapDividerBitsNorm; - } - - norm += (double)lnorm; - if (norm > maxnorm) - { - maxnorm = (unsigned long)norm; - } - - // Normalize result by dividing by sqrt(norm) - this step is easiest - // done using floating point operation - return (double)corr / sqrt((norm < 1e-9) ? 1.0 : norm); -} - -#endif // SOUNDTOUCH_INTEGER_SAMPLES - -////////////////////////////////////////////////////////////////////////////// -// -// Floating point arithmetic specific algorithm implementations. -// - -#ifdef SOUNDTOUCH_FLOAT_SAMPLES - -// Overlaps samples in 'midBuffer' with the samples in 'pInput' -void TDStretch::overlapStereo(float *pOutput, const float *pInput) const -{ - int i; - float fScale; - float f1; - float f2; - - fScale = 1.0f / (float)overlapLength; - - f1 = 0; - f2 = 1.0f; - - for (i = 0; i < 2 * (int)overlapLength ; i += 2) - { - pOutput[i + 0] = pInput[i + 0] * f1 + pMidBuffer[i + 0] * f2; - pOutput[i + 1] = pInput[i + 1] * f1 + pMidBuffer[i + 1] * f2; - - f1 += fScale; - f2 -= fScale; - } -} - - -// Overlaps samples in 'midBuffer' with the samples in 'input'. -void TDStretch::overlapMulti(float *pOutput, const float *pInput) const -{ - int i; - float fScale; - float f1; - float f2; - - fScale = 1.0f / (float)overlapLength; - - f1 = 0; - f2 = 1.0f; - - i=0; - for (int i2 = 0; i2 < overlapLength; i2 ++) - { - // note: Could optimize this slightly by taking into account that always channels > 2 - for (int c = 0; c < channels; c ++) - { - pOutput[i] = pInput[i] * f1 + pMidBuffer[i] * f2; - i++; - } - f1 += fScale; - f2 -= fScale; - } -} - - -/// Calculates overlapInMsec period length in samples. -void TDStretch::calculateOverlapLength(int overlapInMsec) -{ - int newOvl; - - assert(overlapInMsec >= 0); - newOvl = (sampleRate * overlapInMsec) / 1000; - if (newOvl < 16) newOvl = 16; - - // must be divisible by 8 - newOvl -= newOvl % 8; - - acceptNewOverlapLength(newOvl); -} - - -/// Calculate cross-correlation -double TDStretch::calcCrossCorr(const float *mixingPos, const float *compare, double &anorm) -{ - double corr; - double norm; - int i; - - corr = norm = 0; - // Same routine for stereo and mono. For Stereo, unroll by factor of 2. - // For mono it's same routine yet unrollsd by factor of 4. - for (i = 0; i < channels * overlapLength; i += 4) - { - corr += mixingPos[i] * compare[i] + - mixingPos[i + 1] * compare[i + 1]; - - norm += mixingPos[i] * mixingPos[i] + - mixingPos[i + 1] * mixingPos[i + 1]; - - // unroll the loop for better CPU efficiency: - corr += mixingPos[i + 2] * compare[i + 2] + - mixingPos[i + 3] * compare[i + 3]; - - norm += mixingPos[i + 2] * mixingPos[i + 2] + - mixingPos[i + 3] * mixingPos[i + 3]; - } - - anorm = norm; - return corr / sqrt((norm < 1e-9 ? 1.0 : norm)); -} - - -/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value -double TDStretch::calcCrossCorrAccumulate(const float *mixingPos, const float *compare, double &norm) -{ - double corr; - int i; - - corr = 0; - - // cancel first normalizer tap from previous round - for (i = 1; i <= channels; i ++) - { - norm -= mixingPos[-i] * mixingPos[-i]; - } - - // Same routine for stereo and mono. For Stereo, unroll by factor of 2. - // For mono it's same routine yet unrollsd by factor of 4. - for (i = 0; i < channels * overlapLength; i += 4) - { - corr += mixingPos[i] * compare[i] + - mixingPos[i + 1] * compare[i + 1] + - mixingPos[i + 2] * compare[i + 2] + - mixingPos[i + 3] * compare[i + 3]; - } - - // update normalizer with last samples of this round - for (int j = 0; j < channels; j ++) - { - i --; - norm += mixingPos[i] * mixingPos[i]; - } - - return corr / sqrt((norm < 1e-9 ? 1.0 : norm)); -} - - -#endif // SOUNDTOUCH_FLOAT_SAMPLES diff --git a/lib/soundtouch/TDStretch.h b/lib/soundtouch/TDStretch.h deleted file mode 100644 index 4118f9f474c..00000000000 --- a/lib/soundtouch/TDStretch.h +++ /dev/null @@ -1,279 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo -/// while maintaining the original pitch by using a time domain WSOLA-like method -/// with several performance-increasing tweaks. -/// -/// Note : MMX/SSE optimized functions reside in separate, platform-specific files -/// 'mmx_optimized.cpp' and 'sse_optimized.cpp' -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef TDStretch_H -#define TDStretch_H - -#include -#include "STTypes.h" -#include "RateTransposer.h" -#include "FIFOSamplePipe.h" - -namespace soundtouch -{ - -/// Default values for sound processing parameters: -/// Notice that the default parameters are tuned for contemporary popular music -/// processing. For speech processing applications these parameters suit better: -/// #define DEFAULT_SEQUENCE_MS 40 -/// #define DEFAULT_SEEKWINDOW_MS 15 -/// #define DEFAULT_OVERLAP_MS 8 -/// - -/// Default length of a single processing sequence, in milliseconds. This determines to how -/// long sequences the original sound is chopped in the time-stretch algorithm. -/// -/// The larger this value is, the lesser sequences are used in processing. In principle -/// a bigger value sounds better when slowing down tempo, but worse when increasing tempo -/// and vice versa. -/// -/// Increasing this value reduces computational burden & vice versa. -//#define DEFAULT_SEQUENCE_MS 40 -#define DEFAULT_SEQUENCE_MS USE_AUTO_SEQUENCE_LEN - -/// Giving this value for the sequence length sets automatic parameter value -/// according to tempo setting (recommended) -#define USE_AUTO_SEQUENCE_LEN 0 - -/// Seeking window default length in milliseconds for algorithm that finds the best possible -/// overlapping location. This determines from how wide window the algorithm may look for an -/// optimal joining location when mixing the sound sequences back together. -/// -/// The bigger this window setting is, the higher the possibility to find a better mixing -/// position will become, but at the same time large values may cause a "drifting" artifact -/// because consequent sequences will be taken at more uneven intervals. -/// -/// If there's a disturbing artifact that sounds as if a constant frequency was drifting -/// around, try reducing this setting. -/// -/// Increasing this value increases computational burden & vice versa. -//#define DEFAULT_SEEKWINDOW_MS 15 -#define DEFAULT_SEEKWINDOW_MS USE_AUTO_SEEKWINDOW_LEN - -/// Giving this value for the seek window length sets automatic parameter value -/// according to tempo setting (recommended) -#define USE_AUTO_SEEKWINDOW_LEN 0 - -/// Overlap length in milliseconds. When the chopped sound sequences are mixed back together, -/// to form a continuous sound stream, this parameter defines over how long period the two -/// consecutive sequences are let to overlap each other. -/// -/// This shouldn't be that critical parameter. If you reduce the DEFAULT_SEQUENCE_MS setting -/// by a large amount, you might wish to try a smaller value on this. -/// -/// Increasing this value increases computational burden & vice versa. -#define DEFAULT_OVERLAP_MS 8 - - -/// Class that does the time-stretch (tempo change) effect for the processed -/// sound. -class TDStretch : public FIFOProcessor -{ -protected: - int channels; - int sampleReq; - - int overlapLength; - int seekLength; - int seekWindowLength; - int overlapDividerBitsNorm; - int overlapDividerBitsPure; - int slopingDivider; - int sampleRate; - int sequenceMs; - int seekWindowMs; - int overlapMs; - - unsigned long maxnorm; - float maxnormf; - - double tempo; - double nominalSkip; - double skipFract; - - bool bQuickSeek; - bool bAutoSeqSetting; - bool bAutoSeekSetting; - bool isBeginning; - - SAMPLETYPE *pMidBuffer; - SAMPLETYPE *pMidBufferUnaligned; - - FIFOSampleBuffer outputBuffer; - FIFOSampleBuffer inputBuffer; - - void acceptNewOverlapLength(int newOverlapLength); - - virtual void clearCrossCorrState(); - void calculateOverlapLength(int overlapMs); - - virtual double calcCrossCorr(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm); - virtual double calcCrossCorrAccumulate(const SAMPLETYPE *mixingPos, const SAMPLETYPE *compare, double &norm); - - virtual int seekBestOverlapPositionFull(const SAMPLETYPE *refPos); - virtual int seekBestOverlapPositionQuick(const SAMPLETYPE *refPos); - virtual int seekBestOverlapPosition(const SAMPLETYPE *refPos); - - virtual void overlapStereo(SAMPLETYPE *output, const SAMPLETYPE *input) const; - virtual void overlapMono(SAMPLETYPE *output, const SAMPLETYPE *input) const; - virtual void overlapMulti(SAMPLETYPE *output, const SAMPLETYPE *input) const; - - void clearMidBuffer(); - void overlap(SAMPLETYPE *output, const SAMPLETYPE *input, uint ovlPos) const; - - void calcSeqParameters(); - void adaptNormalizer(); - - /// Changes the tempo of the given sound samples. - /// Returns amount of samples returned in the "output" buffer. - /// The maximum amount of samples that can be returned at a time is set by - /// the 'set_returnBuffer_size' function. - void processSamples(); - -public: - TDStretch(); - virtual ~TDStretch(); - - /// 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; - } - - /// return approximate initial input-output latency - int getLatency() const - { - return sampleReq; - } -}; - - -// Implementation-specific class declarations: - -#ifdef SOUNDTOUCH_ALLOW_MMX - /// Class that implements MMX optimized routines for 16bit integer samples type. - class TDStretchMMX : public TDStretch - { - protected: - double calcCrossCorr(const short *mixingPos, const short *compare, double &norm); - 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 diff --git a/lib/soundtouch/cpu_detect.h b/lib/soundtouch/cpu_detect.h deleted file mode 100644 index 0cdc22356f9..00000000000 --- a/lib/soundtouch/cpu_detect.h +++ /dev/null @@ -1,55 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// A header file for detecting the Intel MMX instructions set extension. -/// -/// Please see 'mmx_win.cpp', 'mmx_cpp.cpp' and 'mmx_non_x86.cpp' for the -/// routine implementations for x86 Windows, x86 gnu version and non-x86 -/// platforms, respectively. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _CPU_DETECT_H_ -#define _CPU_DETECT_H_ - -#include "STTypes.h" - -#define SUPPORT_MMX 0x0001 -#define SUPPORT_3DNOW 0x0002 -#define SUPPORT_ALTIVEC 0x0004 -#define SUPPORT_SSE 0x0008 -#define SUPPORT_SSE2 0x0010 - -/// 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_ diff --git a/lib/soundtouch/cpu_detect_x86.cpp b/lib/soundtouch/cpu_detect_x86.cpp deleted file mode 100644 index b1286106ebc..00000000000 --- a/lib/soundtouch/cpu_detect_x86.cpp +++ /dev/null @@ -1,130 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Generic version of the x86 CPU extension detection routine. -/// -/// This file is for GNU & other non-Windows compilers, see 'cpu_detect_x86_win.cpp' -/// for the Microsoft compiler version. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include "cpu_detect.h" -#include "STTypes.h" - - -#if defined(SOUNDTOUCH_ALLOW_X86_OPTIMIZATIONS) - - #if defined(__GNUC__) && defined(__i386__) - // gcc - #include "cpuid.h" - #elif defined(_M_IX86) - // windows non-gcc - #include - #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 -} diff --git a/lib/soundtouch/mmx_optimized.cpp b/lib/soundtouch/mmx_optimized.cpp deleted file mode 100644 index 741ba4f22e5..00000000000 --- a/lib/soundtouch/mmx_optimized.cpp +++ /dev/null @@ -1,396 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// MMX optimized routines. All MMX optimized functions have been gathered into -/// this single source code file, regardless to their class or original source -/// code file, in order to ease porting the library to other compiler and -/// processor platforms. -/// -/// The MMX-optimizations are programmed using MMX compiler intrinsics that -/// are supported both by Microsoft Visual C++ and GCC compilers, so this file -/// should compile with both toolsets. -/// -/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++ -/// 6.0 processor pack" update to support compiler intrinsic syntax. The update -/// is available for download at Microsoft Developers Network, see here: -/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include "STTypes.h" - -#ifdef SOUNDTOUCH_ALLOW_MMX -// MMX routines available only with integer sample type - -using namespace soundtouch; - -////////////////////////////////////////////////////////////////////////////// -// -// implementation of MMX optimized functions of class 'TDStretchMMX' -// -////////////////////////////////////////////////////////////////////////////// - -#include "TDStretch.h" -#include -#include -#include - - -// Calculates cross correlation of two buffers -double TDStretchMMX::calcCrossCorr(const short *pV1, const short *pV2, double &dnorm) -{ - const __m64 *pVec1, *pVec2; - __m64 shifter; - __m64 accu, normaccu; - long corr, norm; - int i; - - pVec1 = (__m64*)pV1; - pVec2 = (__m64*)pV2; - - shifter = _m_from_int(overlapDividerBitsNorm); - normaccu = accu = _mm_setzero_si64(); - - // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples - // during each round for improved CPU-level parallellization. - for (i = 0; i < channels * overlapLength / 16; i ++) - { - __m64 temp, temp2; - - // dictionary of instructions: - // _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3] - // _mm_add_pi32 : 2*32bit add - // _m_psrad : 32bit right-shift - - temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter), - _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter)); - temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec1[0]), shifter), - _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec1[1]), shifter)); - accu = _mm_add_pi32(accu, temp); - normaccu = _mm_add_pi32(normaccu, temp2); - - temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter), - _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter)); - temp2 = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec1[2]), shifter), - _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec1[3]), shifter)); - accu = _mm_add_pi32(accu, temp); - normaccu = _mm_add_pi32(normaccu, temp2); - - pVec1 += 4; - pVec2 += 4; - } - - // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1 - // and finally store the result into the variable "corr" - - accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32)); - corr = _m_to_int(accu); - - normaccu = _mm_add_pi32(normaccu, _mm_srli_si64(normaccu, 32)); - norm = _m_to_int(normaccu); - - // Clear MMS state - _m_empty(); - - if (norm > (long)maxnorm) - { - // modify 'maxnorm' inside critical section to avoid multi-access conflict if in OpenMP mode - #pragma omp critical - if (norm > (long)maxnorm) - { - maxnorm = norm; - } - } - - // Normalize result by dividing by sqrt(norm) - this step is easiest - // done using floating point operation - dnorm = (double)norm; - - return (double)corr / sqrt(dnorm < 1e-9 ? 1.0 : dnorm); - // Note: Warning about the missing EMMS instruction is harmless - // as it'll be called elsewhere. -} - - -/// Update cross-correlation by accumulating "norm" coefficient by previously calculated value -double TDStretchMMX::calcCrossCorrAccumulate(const short *pV1, const short *pV2, double &dnorm) -{ - const __m64 *pVec1, *pVec2; - __m64 shifter; - __m64 accu; - long corr, lnorm; - int i; - - // cancel first normalizer tap from previous round - lnorm = 0; - for (i = 1; i <= channels; i ++) - { - lnorm -= (pV1[-i] * pV1[-i]) >> overlapDividerBitsNorm; - } - - pVec1 = (__m64*)pV1; - pVec2 = (__m64*)pV2; - - shifter = _m_from_int(overlapDividerBitsNorm); - accu = _mm_setzero_si64(); - - // Process 4 parallel sets of 2 * stereo samples or 4 * mono samples - // during each round for improved CPU-level parallellization. - for (i = 0; i < channels * overlapLength / 16; i ++) - { - __m64 temp; - - // dictionary of instructions: - // _m_pmaddwd : 4*16bit multiply-add, resulting two 32bits = [a0*b0+a1*b1 ; a2*b2+a3*b3] - // _mm_add_pi32 : 2*32bit add - // _m_psrad : 32bit right-shift - - temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[0], pVec2[0]), shifter), - _mm_sra_pi32(_mm_madd_pi16(pVec1[1], pVec2[1]), shifter)); - accu = _mm_add_pi32(accu, temp); - - temp = _mm_add_pi32(_mm_sra_pi32(_mm_madd_pi16(pVec1[2], pVec2[2]), shifter), - _mm_sra_pi32(_mm_madd_pi16(pVec1[3], pVec2[3]), shifter)); - accu = _mm_add_pi32(accu, temp); - - pVec1 += 4; - pVec2 += 4; - } - - // copy hi-dword of mm0 to lo-dword of mm1, then sum mmo+mm1 - // and finally store the result into the variable "corr" - - accu = _mm_add_pi32(accu, _mm_srli_si64(accu, 32)); - corr = _m_to_int(accu); - - // Clear MMS state - _m_empty(); - - // update normalizer with last samples of this round - pV1 = (short *)pVec1; - for (int j = 1; j <= channels; j ++) - { - lnorm += (pV1[-j] * pV1[-j]) >> overlapDividerBitsNorm; - } - dnorm += (double)lnorm; - - if (lnorm > (long)maxnorm) - { - maxnorm = lnorm; - } - - // Normalize result by dividing by sqrt(norm) - this step is easiest - // done using floating point operation - return (double)corr / sqrt((dnorm < 1e-9) ? 1.0 : dnorm); -} - - -void TDStretchMMX::clearCrossCorrState() -{ - // Clear MMS state - _m_empty(); - //_asm EMMS; -} - - -// MMX-optimized version of the function overlapStereo -void TDStretchMMX::overlapStereo(short *output, const short *input) const -{ - const __m64 *pVinput, *pVMidBuf; - __m64 *pVdest; - __m64 mix1, mix2, adder, shifter; - int i; - - pVinput = (const __m64*)input; - pVMidBuf = (const __m64*)pMidBuffer; - pVdest = (__m64*)output; - - // mix1 = mixer values for 1st stereo sample - // mix1 = mixer values for 2nd stereo sample - // adder = adder for updating mixer values after each round - - mix1 = _mm_set_pi16(0, overlapLength, 0, overlapLength); - adder = _mm_set_pi16(1, -1, 1, -1); - mix2 = _mm_add_pi16(mix1, adder); - adder = _mm_add_pi16(adder, adder); - - // Overlaplength-division by shifter. "+1" is to account for "-1" deduced in - // overlapDividerBits calculation earlier. - shifter = _m_from_int(overlapDividerBitsPure + 1); - - for (i = 0; i < overlapLength / 4; i ++) - { - __m64 temp1, temp2; - - // load & shuffle data so that input & mixbuffer data samples are paired - temp1 = _mm_unpacklo_pi16(pVMidBuf[0], pVinput[0]); // = i0l m0l i0r m0r - temp2 = _mm_unpackhi_pi16(pVMidBuf[0], pVinput[0]); // = i1l m1l i1r m1r - - // temp = (temp .* mix) >> shifter - temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter); - temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter); - pVdest[0] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit - - // update mix += adder - mix1 = _mm_add_pi16(mix1, adder); - mix2 = _mm_add_pi16(mix2, adder); - - // --- second round begins here --- - - // load & shuffle data so that input & mixbuffer data samples are paired - temp1 = _mm_unpacklo_pi16(pVMidBuf[1], pVinput[1]); // = i2l m2l i2r m2r - temp2 = _mm_unpackhi_pi16(pVMidBuf[1], pVinput[1]); // = i3l m3l i3r m3r - - // temp = (temp .* mix) >> shifter - temp1 = _mm_sra_pi32(_mm_madd_pi16(temp1, mix1), shifter); - temp2 = _mm_sra_pi32(_mm_madd_pi16(temp2, mix2), shifter); - pVdest[1] = _mm_packs_pi32(temp1, temp2); // pack 2*2*32bit => 4*16bit - - // update mix += adder - mix1 = _mm_add_pi16(mix1, adder); - mix2 = _mm_add_pi16(mix2, adder); - - pVinput += 2; - pVMidBuf += 2; - pVdest += 2; - } - - _m_empty(); // clear MMS state -} - - -////////////////////////////////////////////////////////////////////////////// -// -// implementation of MMX optimized functions of class 'FIRFilter' -// -////////////////////////////////////////////////////////////////////////////// - -#include "FIRFilter.h" - - -FIRFilterMMX::FIRFilterMMX() : FIRFilter() -{ - filterCoeffsAlign = 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; -} - -#else - -// workaround to not complain about empty module -bool _dontcomplain_mmx_empty; - -#endif // SOUNDTOUCH_ALLOW_MMX diff --git a/lib/soundtouch/readme.md b/lib/soundtouch/readme.md deleted file mode 100644 index d6f8ef0b9e4..00000000000 --- a/lib/soundtouch/readme.md +++ /dev/null @@ -1,50 +0,0 @@ -# SoundTouch library - -SoundTouch is an open-source audio processing library that allows changing the sound tempo, pitch and playback rate parameters independently from each other: -* Change **tempo** while maintaining the original pitch -* Change **pitch** while maintaining the original tempo -* Change **playback rate** that affects both tempo and pitch at the -same time -* Change any combination of tempo/pitch/rate - -Visit [SoundTouch website](https://www.surina.net/soundtouch) and see the [README file](README.html) for more information and audio examples. - -## Example - -Use SoundStretch example app for modifying wav audio files, for example as follows: - -``` -soundstretch my_original_file.wav output_file.wav -tempo=+15 -pitch=-3 -``` - -See the [README file](README.html) for more usage examples and instructions how to build SoundTouch + SoundStretch. - -Ready [SoundStretch application executables](https://www.surina.net/soundtouch/download.html) are available for download for Windows and Mac OS. - -## Language & Platforms - -SoundTouch is written in C++ and compiles in virtually any platform: -* Windows -* Mac OS -* Linux & Unices (including also Raspberry, Beaglebone, Yocto etc embedded Linux flavors) -* Android -* iOS -* embedded systems - -The source code package includes dynamic library import modules for C#, Java and Pascal/Delphi languages. - -## License - -SoundTouch is released under LGPL v2.1: - -This library is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License version 2.1 as published by the Free Software Foundation. - -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 - -See [LGPL v2.1 full license text ](https://www.gnu.org/licenses/old-licenses/lgpl-2.1.html) for details. - --- - -Also commercial license free of GPL limitations available upon request diff --git a/lib/soundtouch/soundtouch_config.h b/lib/soundtouch/soundtouch_config.h deleted file mode 100644 index cfd8aa27476..00000000000 --- a/lib/soundtouch/soundtouch_config.h +++ /dev/null @@ -1,3 +0,0 @@ -/* Mixxx requires floating-point sample processing */ -#define SOUNDTOUCH_FLOAT_SAMPLES 1 -#undef SOUNDTOUCH_INTEGER_SAMPLES diff --git a/lib/soundtouch/sse_optimized.cpp b/lib/soundtouch/sse_optimized.cpp deleted file mode 100644 index 0dc637015f9..00000000000 --- a/lib/soundtouch/sse_optimized.cpp +++ /dev/null @@ -1,365 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// SSE optimized routines for Pentium-III, Athlon-XP and later CPUs. All SSE -/// optimized functions have been gathered into this single source -/// code file, regardless to their class or original source code file, in order -/// to ease porting the library to other compiler and processor platforms. -/// -/// The SSE-optimizations are programmed using SSE compiler intrinsics that -/// are supported both by Microsoft Visual C++ and GCC compilers, so this file -/// should compile with both toolsets. -/// -/// NOTICE: If using Visual Studio 6.0, you'll need to install the "Visual C++ -/// 6.0 processor pack" update to support SSE instruction set. The update is -/// available for download at Microsoft Developers Network, see here: -/// http://msdn.microsoft.com/en-us/vstudio/aa718349.aspx -/// -/// If the above URL is expired or removed, go to "http://msdn.microsoft.com" and -/// perform a search with keywords "processor pack". -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include "cpu_detect.h" -#include "STTypes.h" - -using namespace soundtouch; - -#ifdef SOUNDTOUCH_ALLOW_SSE - -// SSE routines available only with float sample type - -////////////////////////////////////////////////////////////////////////////// -// -// implementation of SSE optimized functions of class 'TDStretchSSE' -// -////////////////////////////////////////////////////////////////////////////// - -#include "TDStretch.h" -#include -#include - -// 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 diff --git a/vamp-plugins/plugins/MixxxBpmDetection.cpp b/vamp-plugins/plugins/MixxxBpmDetection.cpp index 177081f6992..eda64e542f5 100644 --- a/vamp-plugins/plugins/MixxxBpmDetection.cpp +++ b/vamp-plugins/plugins/MixxxBpmDetection.cpp @@ -175,7 +175,7 @@ std::string MixxxBpmDetection::getCurrentProgram() const { return ""; // no programs } -void MixxxBpmDetection::selectProgram(std::string name) { +void MixxxBpmDetection::selectProgram(std::string /*name*/) { } MixxxBpmDetection::OutputList MixxxBpmDetection::getOutputDescriptors() const { @@ -198,7 +198,7 @@ MixxxBpmDetection::OutputList MixxxBpmDetection::getOutputDescriptors() const { return list; } -bool MixxxBpmDetection::initialise(size_t channels, size_t stepSize, size_t blockSize) { +bool MixxxBpmDetection::initialise(size_t channels, size_t /*stepSize*/, size_t blockSize) { if (channels < getMinChannelCount() || channels > getMaxChannelCount()) return false; @@ -212,7 +212,7 @@ void MixxxBpmDetection::reset() { // Clear buffers, reset stored values, etc } -MixxxBpmDetection::FeatureSet MixxxBpmDetection::process(const float *const *inputBuffers, Vamp::RealTime timestamp) { +MixxxBpmDetection::FeatureSet MixxxBpmDetection::process(const float *const *inputBuffers, Vamp::RealTime /*timestamp*/) { if(m_pDetector != NULL) { m_pDetector->inputSamples(inputBuffers[0], m_iBlockSize); } diff --git a/vamp-plugins/plugins/MixxxBpmDetection.h b/vamp-plugins/plugins/MixxxBpmDetection.h index 35aa2a55450..0bab123ce9a 100644 --- a/vamp-plugins/plugins/MixxxBpmDetection.h +++ b/vamp-plugins/plugins/MixxxBpmDetection.h @@ -2,38 +2,38 @@ #define _MIXXXBPMDETECTION_H_ #include -#include "BPMDetect.h" +#include class MixxxBpmDetection : public Vamp::Plugin { public: MixxxBpmDetection(float inputSampleRate); - virtual ~MixxxBpmDetection(); + ~MixxxBpmDetection() override; - std::string getIdentifier() const; - std::string getName() const; - std::string getDescription() const; - std::string getMaker() const; - int getPluginVersion() const; - std::string getCopyright() const; + std::string getIdentifier() const override; + std::string getName() const override; + std::string getDescription() const override; + std::string getMaker() const override; + int getPluginVersion() const override; + std::string getCopyright() const override; - InputDomain getInputDomain() const; - size_t getPreferredBlockSize() const; - size_t getPreferredStepSize() const; - size_t getMinChannelCount() const; - size_t getMaxChannelCount() const; + InputDomain getInputDomain() const override; + size_t getPreferredBlockSize() const override; + size_t getPreferredStepSize() const override; + size_t getMinChannelCount() const override; + size_t getMaxChannelCount() const override; - ParameterList getParameterDescriptors() const; - float getParameter(std::string identifier) const; - void setParameter(std::string identifier, float value); + ParameterList getParameterDescriptors() const override; + float getParameter(std::string identifier) const override; + void setParameter(std::string identifier, float value) override; - ProgramList getPrograms() const; - std::string getCurrentProgram() const; - void selectProgram(std::string name); + ProgramList getPrograms() const override; + std::string getCurrentProgram() const override; + void selectProgram(std::string name) override; - OutputList getOutputDescriptors() const; + OutputList getOutputDescriptors() const override; - bool initialise(size_t channels, size_t stepSize, size_t blockSize); - void reset(); + bool initialise(size_t channels, size_t stepSize, size_t blockSize) override; + void reset() override; FeatureSet process(const float *const *inputBuffers, Vamp::RealTime timestamp);