From e06ddba339f7354868d243802e560048d7631fb5 Mon Sep 17 00:00:00 2001 From: Be Date: Sat, 14 Aug 2021 17:57:59 -0500 Subject: devendor SoundTouch --- CMakeLists.txt | 28 +- lib/soundtouch/AAFilter.cpp | 222 ------- lib/soundtouch/AAFilter.h | 93 --- lib/soundtouch/BPMDetect.cpp | 574 ----------------- lib/soundtouch/BPMDetect.h | 205 ------ lib/soundtouch/COPYING.TXT | 458 -------------- lib/soundtouch/FIFOSampleBuffer.cpp | 267 -------- lib/soundtouch/FIFOSampleBuffer.h | 177 ------ lib/soundtouch/FIFOSamplePipe.h | 230 ------- lib/soundtouch/FIRFilter.cpp | 324 ---------- lib/soundtouch/FIRFilter.h | 139 ----- lib/soundtouch/InterpolateCubic.cpp | 196 ------ lib/soundtouch/InterpolateCubic.h | 63 -- lib/soundtouch/InterpolateLinear.cpp | 296 --------- lib/soundtouch/InterpolateLinear.h | 88 --- lib/soundtouch/InterpolateShannon.cpp | 181 ------ lib/soundtouch/InterpolateShannon.h | 68 -- lib/soundtouch/PeakFinder.cpp | 277 -------- lib/soundtouch/PeakFinder.h | 90 --- lib/soundtouch/README.html | 923 --------------------------- lib/soundtouch/RateTransposer.cpp | 307 --------- lib/soundtouch/RateTransposer.h | 163 ----- lib/soundtouch/STTypes.h | 183 ------ lib/soundtouch/SoundTouch.cpp | 538 ---------------- lib/soundtouch/SoundTouch.h | 348 ----------- lib/soundtouch/TDStretch.cpp | 1111 --------------------------------- lib/soundtouch/TDStretch.h | 279 --------- lib/soundtouch/cpu_detect.h | 55 -- lib/soundtouch/cpu_detect_x86.cpp | 130 ---- lib/soundtouch/mmx_optimized.cpp | 396 ------------ lib/soundtouch/readme.md | 52 -- lib/soundtouch/soundtouch_config.h | 3 - lib/soundtouch/sse_optimized.cpp | 365 ----------- 33 files changed, 2 insertions(+), 8827 deletions(-) delete mode 100644 lib/soundtouch/AAFilter.cpp delete mode 100644 lib/soundtouch/AAFilter.h delete mode 100644 lib/soundtouch/BPMDetect.cpp delete mode 100644 lib/soundtouch/BPMDetect.h delete mode 100644 lib/soundtouch/COPYING.TXT delete mode 100644 lib/soundtouch/FIFOSampleBuffer.cpp delete mode 100644 lib/soundtouch/FIFOSampleBuffer.h delete mode 100644 lib/soundtouch/FIFOSamplePipe.h delete mode 100644 lib/soundtouch/FIRFilter.cpp delete mode 100644 lib/soundtouch/FIRFilter.h delete mode 100644 lib/soundtouch/InterpolateCubic.cpp delete mode 100644 lib/soundtouch/InterpolateCubic.h delete mode 100644 lib/soundtouch/InterpolateLinear.cpp delete mode 100644 lib/soundtouch/InterpolateLinear.h delete mode 100644 lib/soundtouch/InterpolateShannon.cpp delete mode 100644 lib/soundtouch/InterpolateShannon.h delete mode 100644 lib/soundtouch/PeakFinder.cpp delete mode 100644 lib/soundtouch/PeakFinder.h delete mode 100644 lib/soundtouch/README.html delete mode 100644 lib/soundtouch/RateTransposer.cpp delete mode 100644 lib/soundtouch/RateTransposer.h delete mode 100644 lib/soundtouch/STTypes.h delete mode 100644 lib/soundtouch/SoundTouch.cpp delete mode 100644 lib/soundtouch/SoundTouch.h delete mode 100644 lib/soundtouch/TDStretch.cpp delete mode 100644 lib/soundtouch/TDStretch.h delete mode 100644 lib/soundtouch/cpu_detect.h delete mode 100644 lib/soundtouch/cpu_detect_x86.cpp delete mode 100644 lib/soundtouch/mmx_optimized.cpp delete mode 100644 lib/soundtouch/readme.md delete mode 100644 lib/soundtouch/soundtouch_config.h delete mode 100644 lib/soundtouch/sse_optimized.cpp diff --git a/CMakeLists.txt b/CMakeLists.txt index 26df0e6472..730b264f4c 100644 --- a/CMakeLists.txt +++ b/CMakeLists.txt @@ -2376,32 +2376,8 @@ if(SndFile_SUPPORTS_SET_COMPRESSION_LEVEL) endif() # SoundTouch -find_package(SoundTouch) -default_option(SoundTouch_STATIC "Link libSoundTouch statically" "NOT SoundTouch_FOUND OR SoundTouch_VERSION VERSION_LESS 2.1.2") -if(SoundTouch_STATIC) - message(STATUS "Preparing internal libSoundTouch") - add_library(SoundTouch STATIC EXCLUDE_FROM_ALL - lib/soundtouch/AAFilter.cpp - lib/soundtouch/BPMDetect.cpp - lib/soundtouch/FIFOSampleBuffer.cpp - lib/soundtouch/FIRFilter.cpp - lib/soundtouch/InterpolateCubic.cpp - lib/soundtouch/InterpolateLinear.cpp - lib/soundtouch/InterpolateShannon.cpp - lib/soundtouch/PeakFinder.cpp - lib/soundtouch/RateTransposer.cpp - lib/soundtouch/SoundTouch.cpp - lib/soundtouch/TDStretch.cpp - lib/soundtouch/cpu_detect_x86.cpp - lib/soundtouch/mmx_optimized.cpp - lib/soundtouch/sse_optimized.cpp - ) - target_include_directories(SoundTouch SYSTEM PUBLIC lib) - target_link_libraries(mixxx-lib PRIVATE SoundTouch) -else() - message(STATUS "Linking libSoundTouch dynamically") - target_link_libraries(mixxx-lib PRIVATE SoundTouch::SoundTouch) -endif() +find_package(SoundTouch 2.1.2 REQUIRED) +target_link_libraries(mixxx-lib PRIVATE SoundTouch::SoundTouch) # TagLib find_package(TagLib 1.11 REQUIRED) diff --git a/lib/soundtouch/AAFilter.cpp b/lib/soundtouch/AAFilter.cpp deleted file mode 100644 index 76a3da65d4..0000000000 --- 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 8e5697f796..0000000000 --- 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 3ecda49f61..0000000000 --- a/lib/soundtouch/BPMDetect.cpp +++ /dev/null @@ -1,574 +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; - if ((decimateBy <= 0) || (decimateBy * DECIMATED_BLOCK_SIZE < INPUT_BLOCK_SIZE)) - { - ST_THROW_RT_ERROR("Too small samplerate"); - } - - // Calculate window length & starting item according to desired min & max bpms - windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM); - windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM_RANGE); - - assert(windowLen > windowStart); - - // allocate new working objects - xcorr = new float[windowLen]; - memset(xcorr, 0, windowLen * sizeof(float)); - - pos = 0; - peakPos = 0; - peakVal = 0; - init_scaler = 1; - beatcorr_ringbuffpos = 0; - beatcorr_ringbuff = new float[windowLen]; - memset(beatcorr_ringbuff, 0, windowLen * sizeof(float)); - - // allocate processing buffer - buffer = new FIFOSampleBuffer(); - // we do processing in mono mode - buffer->setChannels(1); - buffer->clear(); - - // calculate hamming windows - hamw = new float[XCORR_UPDATE_SEQUENCE]; - hamming(hamw, XCORR_UPDATE_SEQUENCE); - hamw2 = new float[XCORR_UPDATE_SEQUENCE / 2]; - hamming(hamw2, XCORR_UPDATE_SEQUENCE / 2); -} - - -BPMDetect::~BPMDetect() -{ - delete[] xcorr; - delete[] beatcorr_ringbuff; - delete[] hamw; - delete[] hamw2; - delete buffer; -} - - -/// convert to mono, low-pass filter & decimate to about 500 Hz. -/// return number of outputted samples. -/// -/// Decimation is used to remove the unnecessary frequencies and thus to reduce -/// the amount of data needed to be processed as calculating autocorrelation -/// function is a very-very heavy operation. -/// -/// Anti-alias filtering is done simply by averaging the samples. This is really a -/// poor-man's anti-alias filtering, but it's not so critical in this kind of application -/// (it'd also be difficult to design a high-quality filter with steep cut-off at very -/// narrow band) -int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples) -{ - int count, outcount; - LONG_SAMPLETYPE out; - - assert(channels > 0); - assert(decimateBy > 0); - outcount = 0; - for (count = 0; count < numsamples; count ++) - { - int j; - - // convert to mono and accumulate - for (j = 0; j < channels; j ++) - { - decimateSum += src[j]; - } - src += j; - - decimateCount ++; - if (decimateCount >= decimateBy) - { - // Store every Nth sample only - out = (LONG_SAMPLETYPE)(decimateSum / (decimateBy * channels)); - decimateSum = 0; - decimateCount = 0; -#ifdef SOUNDTOUCH_INTEGER_SAMPLES - // check ranges for sure (shouldn't actually be necessary) - if (out > 32767) - { - out = 32767; - } - else if (out < -32768) - { - out = -32768; - } -#endif // SOUNDTOUCH_INTEGER_SAMPLES - dest[outcount] = (SAMPLETYPE)out; - outcount ++; - } - } - return outcount; -} - - -// Calculates autocorrelation function of the sample history buffer -void BPMDetect::updateXCorr(int process_samples) -{ - int offs; - SAMPLETYPE *pBuffer; - - assert(buffer->numSamples() >= (uint)(process_samples + windowLen)); - assert(process_samples == XCORR_UPDATE_SEQUENCE); - - pBuffer = buffer->ptrBegin(); - - // calculate decay factor for xcorr filtering - float xcorr_decay = (float)pow(0.5, 1.0 / (XCORR_DECAY_TIME_CONSTANT * TARGET_SRATE / process_samples)); - - // prescale pbuffer - float tmp[XCORR_UPDATE_SEQUENCE]; - for (int i = 0; i < process_samples; i++) - { - tmp[i] = hamw[i] * hamw[i] * pBuffer[i]; - } - - #pragma omp parallel for - for (offs = windowStart; offs < windowLen; offs ++) - { - 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 8ece78448d..0000000000 --- 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 bbd24e6430..0000000000 --- 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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