From 6349013134574ccf825f44b7d3cd2d368418f10f Mon Sep 17 00:00:00 2001 From: RJ Ryan Date: Sat, 13 Dec 2014 13:09:27 -0500 Subject: Upgrade from SoundTouch 1.6.0 to 1.8.0. Fixes a portability-related segfault on Windows x64 reported in Bug #1402219. --- lib/soundtouch-1.6.0/AAFilter.cpp | 184 ----- lib/soundtouch-1.6.0/AAFilter.h | 91 --- lib/soundtouch-1.6.0/BPMDetect.cpp | 377 ---------- lib/soundtouch-1.6.0/BPMDetect.h | 170 ----- lib/soundtouch-1.6.0/COPYING.TXT | 458 ------------ lib/soundtouch-1.6.0/FIFOSampleBuffer.cpp | 262 ------- lib/soundtouch-1.6.0/FIFOSampleBuffer.h | 174 ----- lib/soundtouch-1.6.0/FIFOSamplePipe.h | 221 ------ lib/soundtouch-1.6.0/FIRFilter.cpp | 260 ------- lib/soundtouch-1.6.0/FIRFilter.h | 145 ---- lib/soundtouch-1.6.0/PeakFinder.cpp | 239 ------- lib/soundtouch-1.6.0/PeakFinder.h | 93 --- lib/soundtouch-1.6.0/RateTransposer.cpp | 628 ---------------- lib/soundtouch-1.6.0/RateTransposer.h | 159 ----- lib/soundtouch-1.6.0/STTypes.h | 146 ---- lib/soundtouch-1.6.0/SoundTouch.cpp | 486 ------------- lib/soundtouch-1.6.0/SoundTouch.h | 277 -------- lib/soundtouch-1.6.0/TDStretch.cpp | 1026 --------------------------- lib/soundtouch-1.6.0/TDStretch.h | 277 -------- lib/soundtouch-1.6.0/cpu_detect.h | 62 -- lib/soundtouch-1.6.0/cpu_detect_x86_gcc.cpp | 140 ---- lib/soundtouch-1.6.0/cpu_detect_x86_win.cpp | 139 ---- lib/soundtouch-1.6.0/mmx_optimized.cpp | 320 --------- lib/soundtouch-1.6.0/sse_optimized.cpp | 428 ----------- lib/soundtouch-1.8.0/AAFilter.cpp | 236 ++++++ lib/soundtouch-1.8.0/AAFilter.h | 100 +++ lib/soundtouch-1.8.0/BPMDetect.cpp | 370 ++++++++++ lib/soundtouch-1.8.0/BPMDetect.h | 164 +++++ lib/soundtouch-1.8.0/COPYING.TXT | 458 ++++++++++++ lib/soundtouch-1.8.0/FIFOSampleBuffer.cpp | 274 +++++++ lib/soundtouch-1.8.0/FIFOSampleBuffer.h | 184 +++++ lib/soundtouch-1.8.0/FIFOSamplePipe.h | 234 ++++++ lib/soundtouch-1.8.0/FIRFilter.cpp | 333 +++++++++ lib/soundtouch-1.8.0/FIRFilter.h | 150 ++++ lib/soundtouch-1.8.0/InterpolateCubic.cpp | 200 ++++++ lib/soundtouch-1.8.0/InterpolateCubic.h | 67 ++ lib/soundtouch-1.8.0/InterpolateLinear.cpp | 299 ++++++++ lib/soundtouch-1.8.0/InterpolateLinear.h | 92 +++ lib/soundtouch-1.8.0/InterpolateShannon.cpp | 185 +++++ lib/soundtouch-1.8.0/InterpolateShannon.h | 72 ++ lib/soundtouch-1.8.0/PeakFinder.cpp | 276 +++++++ lib/soundtouch-1.8.0/PeakFinder.h | 97 +++ lib/soundtouch-1.8.0/RateTransposer.cpp | 302 ++++++++ lib/soundtouch-1.8.0/RateTransposer.h | 179 +++++ lib/soundtouch-1.8.0/STTypes.h | 184 +++++ lib/soundtouch-1.8.0/SoundTouch.cpp | 504 +++++++++++++ lib/soundtouch-1.8.0/SoundTouch.h | 277 ++++++++ lib/soundtouch-1.8.0/TDStretch.cpp | 947 ++++++++++++++++++++++++ lib/soundtouch-1.8.0/TDStretch.h | 272 +++++++ lib/soundtouch-1.8.0/cpu_detect.h | 62 ++ lib/soundtouch-1.8.0/cpu_detect_x86.cpp | 138 ++++ lib/soundtouch-1.8.0/mmx_optimized.cpp | 384 ++++++++++ lib/soundtouch-1.8.0/soundtouch_config.h | 15 + lib/soundtouch-1.8.0/sse_optimized.cpp | 370 ++++++++++ 54 files changed, 7425 insertions(+), 6762 deletions(-) delete mode 100644 lib/soundtouch-1.6.0/AAFilter.cpp delete mode 100644 lib/soundtouch-1.6.0/AAFilter.h delete mode 100644 lib/soundtouch-1.6.0/BPMDetect.cpp delete mode 100644 lib/soundtouch-1.6.0/BPMDetect.h delete mode 100644 lib/soundtouch-1.6.0/COPYING.TXT delete mode 100644 lib/soundtouch-1.6.0/FIFOSampleBuffer.cpp delete mode 100644 lib/soundtouch-1.6.0/FIFOSampleBuffer.h delete mode 100644 lib/soundtouch-1.6.0/FIFOSamplePipe.h delete mode 100644 lib/soundtouch-1.6.0/FIRFilter.cpp delete mode 100644 lib/soundtouch-1.6.0/FIRFilter.h delete mode 100644 lib/soundtouch-1.6.0/PeakFinder.cpp delete mode 100644 lib/soundtouch-1.6.0/PeakFinder.h delete mode 100644 lib/soundtouch-1.6.0/RateTransposer.cpp delete mode 100644 lib/soundtouch-1.6.0/RateTransposer.h delete mode 100644 lib/soundtouch-1.6.0/STTypes.h delete mode 100644 lib/soundtouch-1.6.0/SoundTouch.cpp delete mode 100644 lib/soundtouch-1.6.0/SoundTouch.h delete mode 100644 lib/soundtouch-1.6.0/TDStretch.cpp delete mode 100644 lib/soundtouch-1.6.0/TDStretch.h delete mode 100644 lib/soundtouch-1.6.0/cpu_detect.h delete mode 100644 lib/soundtouch-1.6.0/cpu_detect_x86_gcc.cpp delete mode 100644 lib/soundtouch-1.6.0/cpu_detect_x86_win.cpp delete mode 100644 lib/soundtouch-1.6.0/mmx_optimized.cpp delete mode 100644 lib/soundtouch-1.6.0/sse_optimized.cpp create mode 100644 lib/soundtouch-1.8.0/AAFilter.cpp create mode 100644 lib/soundtouch-1.8.0/AAFilter.h create mode 100644 lib/soundtouch-1.8.0/BPMDetect.cpp create mode 100644 lib/soundtouch-1.8.0/BPMDetect.h create mode 100644 lib/soundtouch-1.8.0/COPYING.TXT create mode 100644 lib/soundtouch-1.8.0/FIFOSampleBuffer.cpp create mode 100644 lib/soundtouch-1.8.0/FIFOSampleBuffer.h create mode 100644 lib/soundtouch-1.8.0/FIFOSamplePipe.h create mode 100644 lib/soundtouch-1.8.0/FIRFilter.cpp create mode 100644 lib/soundtouch-1.8.0/FIRFilter.h create mode 100644 lib/soundtouch-1.8.0/InterpolateCubic.cpp create mode 100644 lib/soundtouch-1.8.0/InterpolateCubic.h create mode 100644 lib/soundtouch-1.8.0/InterpolateLinear.cpp create mode 100644 lib/soundtouch-1.8.0/InterpolateLinear.h create mode 100644 lib/soundtouch-1.8.0/InterpolateShannon.cpp create mode 100644 lib/soundtouch-1.8.0/InterpolateShannon.h create mode 100644 lib/soundtouch-1.8.0/PeakFinder.cpp create mode 100644 lib/soundtouch-1.8.0/PeakFinder.h create mode 100644 lib/soundtouch-1.8.0/RateTransposer.cpp create mode 100644 lib/soundtouch-1.8.0/RateTransposer.h create mode 100644 lib/soundtouch-1.8.0/STTypes.h create mode 100644 lib/soundtouch-1.8.0/SoundTouch.cpp create mode 100644 lib/soundtouch-1.8.0/SoundTouch.h create mode 100644 lib/soundtouch-1.8.0/TDStretch.cpp create mode 100644 lib/soundtouch-1.8.0/TDStretch.h create mode 100644 lib/soundtouch-1.8.0/cpu_detect.h create mode 100644 lib/soundtouch-1.8.0/cpu_detect_x86.cpp create mode 100644 lib/soundtouch-1.8.0/mmx_optimized.cpp create mode 100644 lib/soundtouch-1.8.0/soundtouch_config.h create mode 100644 lib/soundtouch-1.8.0/sse_optimized.cpp (limited to 'lib') diff --git a/lib/soundtouch-1.6.0/AAFilter.cpp b/lib/soundtouch-1.6.0/AAFilter.cpp deleted file mode 100644 index 96abda49cc..0000000000 --- a/lib/soundtouch-1.6.0/AAFilter.cpp +++ /dev/null @@ -1,184 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// FIR low-pass (anti-alias) filter with filter coefficient design routine and -/// MMX optimization. -/// -/// Anti-alias filter is used to prevent folding of high frequencies when -/// transposing the sample rate with interpolation. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2009-01-11 13:34:24 +0200 (Sun, 11 Jan 2009) $ -// File revision : $Revision: 4 $ -// -// $Id: AAFilter.cpp 45 2009-01-11 11:34:24Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#include "AAFilter.h" -#include "FIRFilter.h" - -using namespace soundtouch; - -#define PI 3.141592655357989 -#define TWOPI (2 * PI) - -/***************************************************************************** - * - * 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 fc2, 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]; - - fc2 = 2.0 * cutoffFreq; - wc = PI * fc2; - 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 = fc2 * 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 ++) - { - // scale & round to nearest integer - temp = work[i] * scaleCoeff; - 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); - - 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); -} - - -uint AAFilter::getLength() const -{ - return pFIR->getLength(); -} diff --git a/lib/soundtouch-1.6.0/AAFilter.h b/lib/soundtouch-1.6.0/AAFilter.h deleted file mode 100644 index d5c8ce4ca3..0000000000 --- a/lib/soundtouch-1.6.0/AAFilter.h +++ /dev/null @@ -1,91 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Sampled sound tempo changer/time stretch algorithm. Changes the sound tempo -/// while maintaining the original pitch by using a time domain WSOLA-like method -/// with several performance-increasing tweaks. -/// -/// Anti-alias filter is used to prevent folding of high frequencies when -/// transposing the sample rate with interpolation. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2008-02-10 18:26:55 +0200 (Sun, 10 Feb 2008) $ -// File revision : $Revision: 4 $ -// -// $Id: AAFilter.h 11 2008-02-10 16:26:55Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef AAFilter_H -#define AAFilter_H - -#include "STTypes.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; -}; - -} - -#endif diff --git a/lib/soundtouch-1.6.0/BPMDetect.cpp b/lib/soundtouch-1.6.0/BPMDetect.cpp deleted file mode 100644 index b36ac6a499..0000000000 --- a/lib/soundtouch-1.6.0/BPMDetect.cpp +++ /dev/null @@ -1,377 +0,0 @@ -//////////////////////////////////////////////////////////////////////////////// -/// -/// Beats-per-minute (BPM) detection routine. -/// -/// The beat detection algorithm works as follows: -/// - Use function 'inputSamples' to input a chunks of samples to the class for -/// analysis. It's a good idea to enter a large sound file or stream in smallish -/// chunks of around few kilosamples in order not to extinguish too much RAM memory. -/// - Inputted sound data is decimated to approx 500 Hz to reduce calculation burden, -/// which is basically ok as low (bass) frequencies mostly determine the beat rate. -/// Simple averaging is used for anti-alias filtering because the resulting signal -/// quality isn't of that high importance. -/// - Decimated sound data is enveloped, i.e. the amplitude shape is detected by -/// taking absolute value that's smoothed by sliding average. Signal levels that -/// are below a couple of times the general RMS amplitude level are cut away to -/// leave only notable peaks there. -/// - Repeating sound patterns (e.g. beats) are detected by calculating short-term -/// autocorrelation function of the enveloped signal. -/// - After whole sound data file has been analyzed as above, the bpm level is -/// detected by function 'getBpm' that finds the highest peak of the autocorrelation -/// function, calculates it's precise location and converts this reading to bpm's. -/// -/// Author : Copyright (c) Olli Parviainen -/// Author e-mail : oparviai 'at' iki.fi -/// SoundTouch WWW: http://www.surina.net/soundtouch -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2011-07-13 11:07:14 +0300 (Wed, 13 Jul 2011) $ -// File revision : $Revision: 4 $ -// -// $Id: BPMDetect.cpp 105 2011-07-13 08:07:14Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#include -#include -#include -#include "FIFOSampleBuffer.h" -#include "PeakFinder.h" -#include "BPMDetect.h" - -using namespace soundtouch; - -#define INPUT_BLOCK_SAMPLES 2048 -#define DECIMATED_BLOCK_SAMPLES 256 - -/// decay constant for calculating RMS volume sliding average approximation -/// (time constant is about 10 sec) -const float avgdecay = 0.99986f; - -/// Normalization coefficient for calculating RMS sliding average approximation. -const float avgnorm = (1 - avgdecay); - - -//////////////////////////////////////////////////////////////////////////////// - -// Enable following define to create bpm analysis file: - -// #define _CREATE_BPM_DEBUG_FILE - -#ifdef _CREATE_BPM_DEBUG_FILE - - #define DEBUGFILE_NAME "c:\\temp\\soundtouch-bpm-debug.txt" - - static void _SaveDebugData(const float *data, int minpos, int maxpos, double coeff) - { - FILE *fptr = fopen(DEBUGFILE_NAME, "wt"); - int i; - - if (fptr) - { - printf("\n\nWriting BPM debug data into file " DEBUGFILE_NAME "\n\n"); - for (i = minpos; i < maxpos; i ++) - { - fprintf(fptr, "%d\t%.1lf\t%f\n", i, coeff / (double)i, data[i]); - } - fclose(fptr); - } - } -#else - #define _SaveDebugData(a,b,c,d) -#endif - -//////////////////////////////////////////////////////////////////////////////// - - -BPMDetect::BPMDetect(int numChannels, int aSampleRate) -{ - this->sampleRate = aSampleRate; - this->channels = numChannels; - - decimateSum = 0; - decimateCount = 0; - - envelopeAccu = 0; - - // Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's - // a typical RMS signal level value for song data. This value is then adapted - // to the actual level during processing. -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - // integer samples - RMSVolumeAccu = (3000 * 3000) / avgnorm; -#else - // float samples, scaled to range [-1..+1[ - RMSVolumeAccu = (0.092f * 0.092f) / avgnorm; -#endif - - cutCoeff = 1.75; - aboveCutAccu = 0; - totalAccu = 0; - - // choose decimation factor so that result is approx. 500 Hz - decimateBy = sampleRate / 500; - assert(decimateBy > 0); - assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES); - - // Calculate window length & starting item according to desired min & max bpms - windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM); - windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM); - - assert(windowLen > windowStart); - - // allocate new working objects - xcorr = new float[windowLen]; - memset(xcorr, 0, windowLen * sizeof(float)); - - // allocate processing buffer - buffer = new FIFOSampleBuffer(); - // we do processing in mono mode - buffer->setChannels(1); - buffer->clear(); -} - - - -BPMDetect::~BPMDetect() -{ - delete[] xcorr; - delete buffer; -} - - - -/// convert to mono, low-pass filter & decimate to about 500 Hz. -/// return number of outputted samples. -/// -/// Decimation is used to remove the unnecessary frequencies and thus to reduce -/// the amount of data needed to be processed as calculating autocorrelation -/// function is a very-very heavy operation. -/// -/// Anti-alias filtering is done simply by averaging the samples. This is really a -/// poor-man's anti-alias filtering, but it's not so critical in this kind of application -/// (it'd also be difficult to design a high-quality filter with steep cut-off at very -/// narrow band) -int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples) -{ - int count, outcount; - LONG_SAMPLETYPE out; - - assert(channels > 0); - assert(decimateBy > 0); - outcount = 0; - for (count = 0; count < numsamples; count ++) - { - int j; - - // convert to mono and accumulate - for (j = 0; j < channels; j ++) - { - decimateSum += src[j]; - } - src += j; - - decimateCount ++; - if (decimateCount >= decimateBy) - { - // Store every Nth sample only - out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels)); - decimateSum = 0; - decimateCount = 0; -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - // check ranges for sure (shouldn't actually be necessary) - if (out > 32767) - { - out = 32767; - } - else if (out < -32768) - { - out = -32768; - } -#endif // SOUNDTOUCH_INTEGER_SAMPLES - dest[outcount] = (SAMPLETYPE)out; - outcount ++; - } - } - return outcount; -} - - - -// Calculates autocorrelation function of the sample history buffer -void BPMDetect::updateXCorr(int process_samples) -{ - int offs; - SAMPLETYPE *pBuffer; - - assert(buffer->numSamples() >= (uint)(process_samples + windowLen)); - - pBuffer = buffer->ptrBegin(); - for (offs = windowStart; offs < windowLen; offs ++) - { - LONG_SAMPLETYPE sum; - int i; - - sum = 0; - for (i = 0; i < process_samples; i ++) - { - sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary - } -// xcorr[offs] *= xcorr_decay; // decay 'xcorr' here with suitable coefficients - // if it's desired that the system adapts automatically to - // various bpms, e.g. in processing continouos music stream. - // The 'xcorr_decay' should be a value that's smaller than but - // close to one, and should also depend on 'process_samples' value. - - xcorr[offs] += (float)sum; - } -} - - -// Calculates envelope of the sample data -void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples) -{ - const static double decay = 0.7f; // decay constant for smoothing the envelope - const static double norm = (1 - decay); - - int i; - LONG_SAMPLETYPE out; - double val; - - for (i = 0; i < numsamples; i ++) - { - // calc average RMS volume - RMSVolumeAccu *= avgdecay; - val = (float)fabs((float)samples[i]); - RMSVolumeAccu += val * val; - - // cut amplitudes that are below cutoff ~2 times RMS volume - // (we're interested in peak values, not the silent moments) - val -= cutCoeff * sqrt(RMSVolumeAccu * avgnorm); - if (val > 0) - { - aboveCutAccu += 1.0; // sample above threshold - } - else - { - val = 0; - } - - totalAccu += 1.0; - - // maintain sliding statistic what proportion of 'val' samples is - // above cutoff threshold - aboveCutAccu *= 0.99931; // 2 sec time constant - totalAccu *= 0.99931; - - if (totalAccu > 500) - { - // after initial settling, auto-adjust cutoff level so that ~8% of - // values are above the threshold - double d = (aboveCutAccu / totalAccu) - 0.08; - cutCoeff += 0.001 * d; - } - - // smooth amplitude envelope - envelopeAccu *= decay; - envelopeAccu += val; - out = (LONG_SAMPLETYPE)(envelopeAccu * norm); - -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - // cut peaks (shouldn't be necessary though) - if (out > 32767) out = 32767; -#endif // SOUNDTOUCH_INTEGER_SAMPLES - samples[i] = (SAMPLETYPE)out; - } - - // check that cutoff doesn't get too small - it can be just silent sequence! - if (cutCoeff < 1.5) - { - cutCoeff = 1.5; - } -} - - - -void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples) -{ - SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES]; - - // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration - while (numSamples > 0) - { - int block; - int decSamples; - - block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples; - - // decimate. note that converts to mono at the same time - decSamples = decimate(decimated, samples, block); - samples += block * channels; - numSamples -= block; - - // envelope new samples and add them to buffer - calcEnvelope(decimated, decSamples); - buffer->putSamples(decimated, decSamples); - } - - // when the buffer has enought samples for processing... - if ((int)buffer->numSamples() > windowLen) - { - int processLength; - - // how many samples are processed - processLength = (int)buffer->numSamples() - windowLen; - - // ... calculate autocorrelations for oldest samples... - updateXCorr(processLength); - // ... and remove them from the buffer - buffer->receiveSamples(processLength); - } -} - - - -float BPMDetect::getBpm() -{ - double peakPos; - double coeff; - PeakFinder peakFinder; - - coeff = 60.0 * ((double)sampleRate / (double)decimateBy); - - // save bpm debug analysis data if debug data enabled - _SaveDebugData(xcorr, windowStart, windowLen, coeff); - - // find peak position - peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen); - - assert(decimateBy != 0); - if (peakPos < 1e-9) return 0.0; // detection failed. - - // calculate BPM - return (float) (coeff / peakPos); -} diff --git a/lib/soundtouch-1.6.0/BPMDetect.h b/lib/soundtouch-1.6.0/BPMDetect.h deleted file mode 100644 index 83e87c0e0e..0000000000 --- a/lib/soundtouch-1.6.0/BPMDetect.h +++ /dev/null @@ -1,170 +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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2010-01-06 20:59:23 +0200 (Wed, 06 Jan 2010) $ -// File revision : $Revision: 4 $ -// -// $Id: BPMDetect.h 86 2010-01-06 18:59:23Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#ifndef _BPMDetect_H_ -#define _BPMDetect_H_ - -#include "STTypes.h" -#include "FIFOSampleBuffer.h" - -namespace soundtouch -{ - -/// Minimum allowed BPM rate. Used to restrict accepted result above a reasonable limit. -#define MIN_BPM 29 - -/// Maximum allowed BPM rate. Used to restrict accepted result below a reasonable limit. -#define MAX_BPM 230 - - -/// Class for calculating BPM rate for audio data. -class BPMDetect -{ -protected: - /// Auto-correlation accumulator bins. - float *xcorr; - - /// Amplitude envelope sliding average approximation level accumulator - double envelopeAccu; - - /// RMS volume sliding average approximation level accumulator - double RMSVolumeAccu; - - /// Level below which to cut off signals - double cutCoeff; - - /// Accumulator for accounting what proportion of samples exceed cutCoeff level - double aboveCutAccu; - - /// Accumulator for total samples to calculate proportion of samples that exceed cutCoeff level - double totalAccu; - - /// Sample average counter. - int decimateCount; - - /// Sample average accumulator for FIFO-like decimation. - soundtouch::LONG_SAMPLETYPE decimateSum; - - /// Decimate sound by this coefficient to reach approx. 500 Hz. - int decimateBy; - - /// Auto-correlation window length - int windowLen; - - /// Number of channels (1 = mono, 2 = stereo) - int channels; - - /// sample rate - int sampleRate; - - /// Beginning of auto-correlation window: Autocorrelation isn't being updated for - /// the first these many correlation bins. - int windowStart; - - /// FIFO-buffer for decimated processing samples. - soundtouch::FIFOSampleBuffer *buffer; - - /// Updates auto-correlation function for given number of decimated samples that - /// are read from the internal 'buffer' pipe (samples aren't removed from the pipe - /// though). - void updateXCorr(int process_samples /// How many samples are processed. - ); - - /// Decimates samples to approx. 500 Hz. - /// - /// \return Number of output samples. - int decimate(soundtouch::SAMPLETYPE *dest, ///< Destination buffer - const soundtouch::SAMPLETYPE *src, ///< Source sample buffer - int numsamples ///< Number of source samples. - ); - - /// Calculates amplitude envelope for the buffer of samples. - /// Result is output to 'samples'. - void calcEnvelope(soundtouch::SAMPLETYPE *samples, ///< Pointer to input/output data buffer - int numsamples ///< Number of samples in buffer - ); - -public: - /// Constructor. - BPMDetect(int numChannels, ///< Number of channels in sample data. - int sampleRate ///< Sample rate in Hz. - ); - - /// Destructor. - virtual ~BPMDetect(); - - /// Inputs a block of samples for analyzing: Envelopes the samples and then - /// updates the autocorrelation estimation. When whole song data has been input - /// in smaller blocks using this function, read the resulting bpm with 'getBpm' - /// function. - /// - /// Notice that data in 'samples' array can be disrupted in processing. - void inputSamples(const soundtouch::SAMPLETYPE *samples, ///< Pointer to input/working data buffer - int numSamples ///< Number of samples in buffer - ); - - - /// Analyzes the results and returns the BPM rate. Use this function to read result - /// after whole song data has been input to the class by consecutive calls of - /// 'inputSamples' function. - /// - /// \return Beats-per-minute rate, or zero if detection failed. - float getBpm(); -}; - -} - -#endif // _BPMDetect_H_ diff --git a/lib/soundtouch-1.6.0/COPYING.TXT b/lib/soundtouch-1.6.0/COPYING.TXT deleted file mode 100644 index 5b2161be20..0000000000 --- a/lib/soundtouch-1.6.0/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 -/// -//////////////////////////////////////////////////////////////////////////////// -// -// Last changed : $Date: 2009-02-27 19:24:42 +0200 (Fri, 27 Feb 2009) $ -// File revision : $Revision: 4 $ -// -// $Id: FIFOSampleBuffer.cpp 68 2009-02-27 17:24:42Z oparviai $ -// -//////////////////////////////////////////////////////////////////////////////// -// -// License : -// -// SoundTouch audio processing library -// Copyright (c) Olli Parviainen -// -// This library is free software; you can redistribute it and/or -// modify it under the terms of the GNU Lesser General Public -// License as published by the Free Software Foundation; either -// version 2.1 of the License, or (at your option) any later version. -// -// This library is distributed in the hope that it will be useful, -// but WITHOUT ANY WARRANTY; without even the implied warranty of -// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -// Lesser General Public License for more details. -// -// You should have received a copy of the GNU Lesser General Public -// License along with this library; if not, write to the Free Software -// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -// -//////////////////////////////////////////////////////////////////////////////// - -#include -#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; - - assert(numChannels > 0); - usedBytes = channels * samplesInBuffer; - channels = (uint)numChannels; - samplesInBuffer = usedBytes / channels; -} - - -// if output location pointer 'bufferPos' isn't zero, 'rewinds' the buffer and -// zeroes this pointer by copying samples from the 'bufferPos' pointer -// location on to the beginning of the buffer. -void FIFOSampleBuffer::rewind() -{ - if (buffer && bufferPos) - { - memmove(buffer, ptrBegin(), sizeof(SAMPLETYPE) * channels * samplesInBuffer); - bufferPos = 0; - } -} - - -// Adds 'numSamples' pcs of samples from the 'samples' memory position to -// the sample buffer. -void FIFOSampleBuffer::putSamples(const SAMPLETYPE *samples, uint nSamples) -{ - memcpy(ptrEnd(nSamples), samples, sizeof(SAMPLETYPE) * nSamples * channels); - samplesInBuffer += nSamples; -} - - -// Increases the number of samples in the buffer without copying any actual -// samples. -// -// This function is used to update the number of samples in the sample buffer -// when accessing the buffer directly with 'ptrEnd' function. Please be -// careful though! -void FIFOSampleBuffer::putSamples(uint nSamples) -{ - uint req; -