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https://github.com/cookiengineer/audacity
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Remove some old erratta, and do a major tidy up of line endings and properties on source files
This commit is contained in:
ra
@@ -1,311 +1,311 @@
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////////////////////////////////////////////////////////////////////////////////
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///
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/// Beats-per-minute (BPM) detection routine.
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||||
///
|
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/// The beat detection algorithm works as follows:
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/// - 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
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/// 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
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/// function, calculates it's precise location and converts this reading to bpm's.
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||||
///
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||||
/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai 'at' iki.fi
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/// SoundTouch WWW: http://www.surina.net/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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||||
//
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||||
// Last changed : $Date: 2006-09-18 07:31:48 $
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// File revision : $Revision: 1.2 $
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//
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// $Id: BPMDetect.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
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//
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||||
////////////////////////////////////////////////////////////////////////////////
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||||
//
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||||
// License :
|
||||
//
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||||
// SoundTouch audio processing library
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||||
// Copyright (c) Olli Parviainen
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||||
//
|
||||
// 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
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||||
//
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////////////////////////////////////////////////////////////////////////////////
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#include <math.h>
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#include <assert.h>
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#include <string.h>
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#include "FIFOSampleBuffer.h"
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#include "PeakFinder.h"
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#include "BPMDetect.h"
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using namespace soundtouch;
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#define INPUT_BLOCK_SAMPLES 2048
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#define DECIMATED_BLOCK_SAMPLES 256
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typedef unsigned short ushort;
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||||
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/// decay constant for calculating RMS volume sliding average approximation
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/// (time constant is about 10 sec)
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const float avgdecay = 0.99986f;
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/// Normalization coefficient for calculating RMS sliding average approximation.
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const float avgnorm = (1 - avgdecay);
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||||
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||||
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||||
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BPMDetect::BPMDetect(int numChannels, int sampleRate)
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{
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xcorr = NULL;
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buffer = new FIFOSampleBuffer();
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decimateSum = 0;
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decimateCount = 0;
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decimateBy = 0;
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||||
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this->sampleRate = sampleRate;
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this->channels = numChannels;
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envelopeAccu = 0;
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// Initialize RMS volume accumulator to RMS level of 3000 (out of 32768) that's
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// a typical RMS signal level value for song data. This value is then adapted
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// to the actual level during processing.
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#ifdef INTEGER_SAMPLES
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// integer samples
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RMSVolumeAccu = (3000 * 3000) / avgnorm;
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#else
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// float samples, scaled to range [-1..+1[
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RMSVolumeAccu = (0.092f * 0.092f) / avgnorm;
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#endif
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init(numChannels, sampleRate);
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}
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BPMDetect::~BPMDetect()
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||||
{
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delete[] xcorr;
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delete buffer;
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}
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||||
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||||
|
||||
/// 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
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||||
/// (it'd also be difficult to design a high-quality filter with steep cut-off at very
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||||
/// narrow band)
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int BPMDetect::decimate(SAMPLETYPE *dest, const SAMPLETYPE *src, int numsamples)
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{
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int count, outcount;
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LONG_SAMPLETYPE out;
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assert(decimateBy != 0);
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outcount = 0;
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for (count = 0; count < numsamples; count ++)
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{
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decimateSum += src[count];
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decimateCount ++;
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if (decimateCount >= decimateBy)
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{
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// Store every Nth sample only
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||||
out = (LONG_SAMPLETYPE)(decimateSum / decimateBy);
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decimateSum = 0;
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||||
decimateCount = 0;
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||||
#ifdef INTEGER_SAMPLES
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// check ranges for sure (shouldn't actually be necessary)
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||||
if (out > 32767)
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||||
{
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||||
out = 32767;
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||||
}
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else if (out < -32768)
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||||
{
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out = -32768;
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}
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#endif // INTEGER_SAMPLES
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dest[outcount] = (SAMPLETYPE)out;
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outcount ++;
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}
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}
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return outcount;
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}
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// Calculates autocorrelation function of the sample history buffer
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void BPMDetect::updateXCorr(int process_samples)
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||||
{
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int offs;
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SAMPLETYPE *pBuffer;
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assert(buffer->numSamples() >= (uint)(process_samples + windowLen));
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pBuffer = buffer->ptrBegin();
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for (offs = windowStart; offs < windowLen; offs ++)
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{
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LONG_SAMPLETYPE sum;
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||||
int i;
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||||
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sum = 0;
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||||
for (i = 0; i < process_samples; i ++)
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||||
{
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sum += pBuffer[i] * pBuffer[i + offs]; // scaling the sub-result shouldn't be necessary
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||||
}
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// 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.
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||||
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xcorr[offs] += (float)sum;
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}
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}
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||||
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// Calculates envelope of the sample data
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void BPMDetect::calcEnvelope(SAMPLETYPE *samples, int numsamples)
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{
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const float decay = 0.7f; // decay constant for smoothing the envelope
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const float norm = (1 - decay);
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int i;
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LONG_SAMPLETYPE out;
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float val;
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for (i = 0; i < numsamples; i ++)
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{
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// calc average RMS volume
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RMSVolumeAccu *= avgdecay;
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val = (float)fabs((float)samples[i]);
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RMSVolumeAccu += val * val;
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// cut amplitudes that are below 2 times average RMS volume
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// (we're interested in peak values, not the silent moments)
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val -= 2 * (float)sqrt(RMSVolumeAccu * avgnorm);
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val = (val > 0) ? val : 0;
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// smooth amplitude envelope
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envelopeAccu *= decay;
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envelopeAccu += val;
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out = (LONG_SAMPLETYPE)(envelopeAccu * norm);
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#ifdef INTEGER_SAMPLES
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// cut peaks (shouldn't be necessary though)
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if (out > 32767) out = 32767;
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#endif // INTEGER_SAMPLES
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samples[i] = (SAMPLETYPE)out;
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}
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}
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void BPMDetect::inputSamples(SAMPLETYPE *samples, int numSamples)
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{
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SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
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// convert from stereo to mono if necessary
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if (channels == 2)
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{
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int i;
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for (i = 0; i < numSamples; i ++)
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||||
{
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samples[i] = (samples[i * 2] + samples[i * 2 + 1]) / 2;
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}
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}
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// decimate
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numSamples = decimate(decimated, samples, numSamples);
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// envelope new samples and add them to buffer
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calcEnvelope(decimated, numSamples);
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buffer->putSamples(decimated, numSamples);
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// when the buffer has enought samples for processing...
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if ((int)buffer->numSamples() > windowLen)
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||||
{
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int processLength;
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// how many samples are processed
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processLength = buffer->numSamples() - windowLen;
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// ... calculate autocorrelations for oldest samples...
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updateXCorr(processLength);
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// ... and remove them from the buffer
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buffer->receiveSamples(processLength);
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}
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}
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||||
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||||
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void BPMDetect::init(int numChannels, int sampleRate)
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{
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this->sampleRate = sampleRate;
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// choose decimation factor so that result is approx. 500 Hz
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decimateBy = sampleRate / 500;
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assert(decimateBy > 0);
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assert(INPUT_BLOCK_SAMPLES < decimateBy * DECIMATED_BLOCK_SAMPLES);
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// Calculate window length & starting item according to desired min & max bpms
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windowLen = (60 * sampleRate) / (decimateBy * MIN_BPM);
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windowStart = (60 * sampleRate) / (decimateBy * MAX_BPM);
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assert(windowLen > windowStart);
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// allocate new working objects
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xcorr = new float[windowLen];
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memset(xcorr, 0, windowLen * sizeof(float));
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// we do processing in mono mode
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buffer->setChannels(1);
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buffer->clear();
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}
|
||||
|
||||
|
||||
|
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float BPMDetect::getBpm()
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{
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float peakPos;
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PeakFinder peakFinder;
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|
||||
// find peak position
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||||
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
|
||||
|
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assert(decimateBy != 0);
|
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if (peakPos < 1e-6) return 0.0; // detection failed.
|
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|
||||
// calculate BPM
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return 60.0f * (((float)sampleRate / (float)decimateBy) / peakPos);
|
||||
}
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
///
|
||||
/// 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: 2006-09-18 07:31:48 $
|
||||
// File revision : $Revision: 1.2 $
|
||||
//
|
||||
// $Id: BPMDetect.cpp,v 1.2 2006-09-18 07:31:48 richardash1981 Exp $
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// License :
|
||||
//
|
||||
// SoundTouch audio processing library
|
||||
// Copyright (c) Olli Parviainen
|
||||
//
|
||||
// This library is free software; you can redistribute it and/or
|
||||
// modify it under the terms of the GNU Lesser General Public
|
||||
// License as published by the Free Software Foundation; either
|
||||
// version 2.1 of the License, or (at your option) any later version.
|
||||
//
|
||||
// This library is distributed in the hope that it will be useful,
|
||||
// but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
// Lesser General Public License for more details.
|
||||
//
|
||||
// You should have received a copy of the GNU Lesser General Public
|
||||
// License along with this library; if not, write to the Free Software
|
||||
// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
||||
//
|
||||
////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
#include <math.h>
|
||||
#include <assert.h>
|
||||
#include <string.h>
|
||||
#include "FIFOSampleBuffer.h"
|
||||
#include "PeakFinder.h"
|
||||
#include "BPMDetect.h"
|
||||
|
||||
using namespace soundtouch;
|
||||
|
||||
#define INPUT_BLOCK_SAMPLES 2048
|
||||
#define DECIMATED_BLOCK_SAMPLES 256
|
||||
|
||||
typedef unsigned short ushort;
|
||||
|
||||
/// 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 sampleRate)
|
||||
{
|
||||
xcorr = NULL;
|
||||
|
||||
buffer = new FIFOSampleBuffer();
|
||||
|
||||
decimateSum = 0;
|
||||
decimateCount = 0;
|
||||
decimateBy = 0;
|
||||
|
||||
this->sampleRate = sampleRate;
|
||||
this->channels = numChannels;
|
||||
|
||||
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
|
||||
|
||||
init(numChannels, sampleRate);
|
||||
}
|
||||
|
||||
|
||||
|
||||
BPMDetect::~BPMDetect()
|
||||
{
|
||||
delete[] xcorr;
|
||||
delete buffer;
|
||||
}
|
||||
|
||||
|
||||
/// 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(decimateBy != 0);
|
||||
outcount = 0;
|
||||
for (count = 0; count < numsamples; count ++)
|
||||
{
|
||||
decimateSum += src[count];
|
||||
|
||||
decimateCount ++;
|
||||
if (decimateCount >= decimateBy)
|
||||
{
|
||||
// Store every Nth sample only
|
||||
out = (LONG_SAMPLETYPE)(decimateSum / decimateBy);
|
||||
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(SAMPLETYPE *samples, int numSamples)
|
||||
{
|
||||
SAMPLETYPE decimated[DECIMATED_BLOCK_SAMPLES];
|
||||
|
||||
// convert from stereo to mono if necessary
|
||||
if (channels == 2)
|
||||
{
|
||||
int i;
|
||||
|
||||
for (i = 0; i < numSamples; i ++)
|
||||
{
|
||||
samples[i] = (samples[i * 2] + samples[i * 2 + 1]) / 2;
|
||||
}
|
||||
}
|
||||
|
||||
// decimate
|
||||
numSamples = decimate(decimated, samples, numSamples);
|
||||
|
||||
// envelope new samples and add them to buffer
|
||||
calcEnvelope(decimated, numSamples);
|
||||
buffer->putSamples(decimated, numSamples);
|
||||
|
||||
// when the buffer has enought samples for processing...
|
||||
if ((int)buffer->numSamples() > windowLen)
|
||||
{
|
||||
int processLength;
|
||||
|
||||
// how many samples are processed
|
||||
processLength = buffer->numSamples() - windowLen;
|
||||
|
||||
// ... calculate autocorrelations for oldest samples...
|
||||
updateXCorr(processLength);
|
||||
// ... and remove them from the buffer
|
||||
buffer->receiveSamples(processLength);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
void BPMDetect::init(int numChannels, int sampleRate)
|
||||
{
|
||||
this->sampleRate = sampleRate;
|
||||
|
||||
// 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));
|
||||
|
||||
// we do processing in mono mode
|
||||
buffer->setChannels(1);
|
||||
buffer->clear();
|
||||
}
|
||||
|
||||
|
||||
|
||||
float BPMDetect::getBpm()
|
||||
{
|
||||
float peakPos;
|
||||
PeakFinder peakFinder;
|
||||
|
||||
// find peak position
|
||||
peakPos = peakFinder.detectPeak(xcorr, windowStart, windowLen);
|
||||
|
||||
assert(decimateBy != 0);
|
||||
if (peakPos < 1e-6) return 0.0; // detection failed.
|
||||
|
||||
// calculate BPM
|
||||
return 60.0f * (((float)sampleRate / (float)decimateBy) / peakPos);
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user