From 8adfb3bd99b4dcff2459756af090a640fd7a4b4a Mon Sep 17 00:00:00 2001 From: yo mama Date: Fri, 19 Jun 2015 16:24:27 -0400 Subject: clone --- .../soundtouch/source/SoundTouch/BPMDetect.cpp | 311 +++++++++++++++++++++ 1 file changed, 311 insertions(+) create mode 100644 pysoundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp (limited to 'pysoundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp') diff --git a/pysoundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp b/pysoundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp new file mode 100644 index 0000000..4e7d386 --- /dev/null +++ b/pysoundtouch/soundtouch/source/SoundTouch/BPMDetect.cpp @@ -0,0 +1,311 @@ +//////////////////////////////////////////////////////////////////////////////// +/// +/// 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: 2009-02-21 18:00:14 +0200 (Sat, 21 Feb 2009) $ +// File revision : $Revision: 4 $ +// +// $Id: BPMDetect.cpp 63 2009-02-21 16:00:14Z 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 +#define RESOL_FACTOR 1 + +/// 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); + + + +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 INTEGER_SAMPLES + // integer samples + RMSVolumeAccu = (3000 * 3000) / avgnorm; +#else + // float samples, scaled to range [-1..+1[ + RMSVolumeAccu = (0.092f * 0.092f) / avgnorm; +#endif + + // choose decimation factor so that result is approx. 500 Hz + decimateBy = sampleRate / (500 * RESOL_FACTOR); + 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 INTEGER_SAMPLES + // check ranges for sure (shouldn't actually be necessary) + if (out > 32767) + { + out = 32767; + } + else if (out < -32768) + { + out = -32768; + } +#endif // 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 float decay = 0.7f; // decay constant for smoothing the envelope + const float norm = (1 - decay); + + int i; + LONG_SAMPLETYPE out; + float 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 2 times average RMS volume + // (we're interested in peak values, not the silent moments) + val -= 2 * (float)sqrt(RMSVolumeAccu * avgnorm); + val = (val > 0) ? val : 0; + + // smooth amplitude envelope + envelopeAccu *= decay; + envelopeAccu += val; + out = (LONG_SAMPLETYPE)(envelopeAccu * norm); + +#ifdef INTEGER_SAMPLES + // cut peaks (shouldn't be necessary though) + if (out > 32767) out = 32767; +#endif // INTEGER_SAMPLES + samples[i] = (SAMPLETYPE)out; + } +} + + + +void BPMDetect::inputSamples(const SAMPLETYPE *samples, int numSamples) +{ + SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES]; + + // iterate so that max INPUT_BLOCK_SAMPLES processed per iteration + while (numSamples > 0) + { + int block; + int decSamples; + + block = (numSamples > INPUT_BLOCK_SAMPLES) ? INPUT_BLOCK_SAMPLES : numSamples; + + // decimate. note that converts to mono at the same time + decSamples = decimate(decimated, samples, block); + samples += block * channels; + numSamples -= block; + + // envelope new samples and add them to buffer + calcEnvelope(decimated, decSamples); + buffer->putSamples(decimated, decSamples); + } + + // when the buffer has enought samples for processing... + if ((int)buffer->numSamples() > windowLen) + { + int processLength; + + // how many samples are processed + processLength = (int)buffer->numSamples() - windowLen; + + // ... calculate autocorrelations for oldest samples... + updateXCorr(processLength); + // ... and remove them from the buffer + buffer->receiveSamples(processLength); + } +} + + + +float BPMDetect::getBpm() +{ + double peakPos; + PeakFinder peakFinder; + + // find peak position + peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen, 1); + printf("HERE: %lf\n", peakPos); + + assert(decimateBy != 0); + if (peakPos < 1e-6) return 0.0; // detection failed. + + // calculate BPM + return (float)(60.0 * (((double)sampleRate / (double)decimateBy) / peakPos)); +} -- cgit v1.2.3-70-g09d2