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diff --git a/libs/qm-dsp/dsp/tempotracking/DownBeat.cpp b/libs/qm-dsp/dsp/tempotracking/DownBeat.cpp
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+++ b/libs/qm-dsp/dsp/tempotracking/DownBeat.cpp
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+/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    QM DSP Library
+
+    Centre for Digital Music, Queen Mary, University of London.
+    This file copyright 2008-2009 Matthew Davies and QMUL.
+
+    This program is free software; you can redistribute it and/or
+    modify it under the terms of the GNU General Public License as
+    published by the Free Software Foundation; either version 2 of the
+    License, or (at your option) any later version.  See the file
+    COPYING included with this distribution for more information.
+*/
+
+#include "DownBeat.h"
+
+#include "maths/MathAliases.h"
+#include "maths/MathUtilities.h"
+#include "maths/KLDivergence.h"
+#include "dsp/transforms/FFT.h"
+
+#include <iostream>
+#include <cstdlib>
+
+DownBeat::DownBeat(float originalSampleRate,
+                   size_t decimationFactor,
+                   size_t dfIncrement) :
+    m_bpb(0),
+    m_rate(originalSampleRate),
+    m_factor(decimationFactor),
+    m_increment(dfIncrement),
+    m_decimator1(0),
+    m_decimator2(0),
+    m_buffer(0),
+    m_decbuf(0),
+    m_bufsiz(0),
+    m_buffill(0),
+    m_beatframesize(0),
+    m_beatframe(0)
+{
+    // beat frame size is next power of two up from 1.3 seconds at the
+    // downsampled rate (happens to produce 4096 for 44100 or 48000 at
+    // 16x decimation, which is our expected normal situation)
+    m_beatframesize = MathUtilities::nextPowerOfTwo
+        (int((m_rate / decimationFactor) * 1.3));
+//    std::cerr << "rate = " << m_rate << ", bfs = " << m_beatframesize << std::endl;
+    m_beatframe = new double[m_beatframesize];
+    m_fftRealOut = new double[m_beatframesize];
+    m_fftImagOut = new double[m_beatframesize];
+    m_fft = new FFTReal(m_beatframesize);
+}
+
+DownBeat::~DownBeat()
+{
+    delete m_decimator1;
+    delete m_decimator2;
+    if (m_buffer) free(m_buffer);
+    delete[] m_decbuf;
+    delete[] m_beatframe;
+    delete[] m_fftRealOut;
+    delete[] m_fftImagOut;
+    delete m_fft;
+}
+
+void
+DownBeat::setBeatsPerBar(int bpb)
+{
+    m_bpb = bpb;
+}
+
+void
+DownBeat::makeDecimators()
+{
+//    std::cerr << "m_factor = " << m_factor << std::endl;
+    if (m_factor < 2) return;
+    size_t highest = Decimator::getHighestSupportedFactor();
+    if (m_factor <= highest) {
+        m_decimator1 = new Decimator(m_increment, m_factor);
+//        std::cerr << "DownBeat: decimator 1 factor " << m_factor << ", size " << m_increment << std::endl;
+        return;
+    }
+    m_decimator1 = new Decimator(m_increment, highest);
+//    std::cerr << "DownBeat: decimator 1 factor " << highest << ", size " << m_increment << std::endl;
+    m_decimator2 = new Decimator(m_increment / highest, m_factor / highest);
+//    std::cerr << "DownBeat: decimator 2 factor " << m_factor / highest << ", size " << m_increment / highest << std::endl;
+    m_decbuf = new float[m_increment / highest];
+}
+
+void
+DownBeat::pushAudioBlock(const float *audio)
+{
+    if (m_buffill + (m_increment / m_factor) > m_bufsiz) {
+        if (m_bufsiz == 0) m_bufsiz = m_increment * 16;
+        else m_bufsiz = m_bufsiz * 2;
+        if (!m_buffer) {
+            m_buffer = (float *)malloc(m_bufsiz * sizeof(float));
+        } else {
+//            std::cerr << "DownBeat::pushAudioBlock: realloc m_buffer to " << m_bufsiz << std::endl;
+            m_buffer = (float *)realloc(m_buffer, m_bufsiz * sizeof(float));
+        }
+    }
+    if (!m_decimator1 && m_factor > 1) makeDecimators();
+//    float rmsin = 0, rmsout = 0;
+//    for (int i = 0; i < m_increment; ++i) {
+//        rmsin += audio[i] * audio[i];
+//    }
+    if (m_decimator2) {
+        m_decimator1->process(audio, m_decbuf);
+        m_decimator2->process(m_decbuf, m_buffer + m_buffill);
+    } else if (m_decimator1) {
+        m_decimator1->process(audio, m_buffer + m_buffill);
+    } else {
+        // just copy across (m_factor is presumably 1)
+        for (size_t i = 0; i < m_increment; ++i) {
+            (m_buffer + m_buffill)[i] = audio[i];
+        }
+    }
+//    for (int i = 0; i < m_increment / m_factor; ++i) {
+//        rmsout += m_buffer[m_buffill + i] * m_buffer[m_buffill + i];
+//    }
+//    std::cerr << "pushAudioBlock: rms in " << sqrt(rmsin) << ", out " << sqrt(rmsout) << std::endl;
+    m_buffill += m_increment / m_factor;
+}
+    
+const float *
+DownBeat::getBufferedAudio(size_t &length) const
+{
+    length = m_buffill;
+    return m_buffer;
+}
+
+void
+DownBeat::resetAudioBuffer()
+{
+    if (m_buffer) free(m_buffer);
+    m_buffer = 0;
+    m_buffill = 0;
+    m_bufsiz = 0;
+}
+
+void
+DownBeat::findDownBeats(const float *audio,
+                        size_t audioLength,
+                        const d_vec_t &beats,
+                        i_vec_t &downbeats)
+{
+    // FIND DOWNBEATS BY PARTITIONING THE INPUT AUDIO FILE INTO BEAT SEGMENTS
+    // WHERE THE AUDIO FRAMES ARE DOWNSAMPLED  BY A FACTOR OF 16 (fs ~= 2700Hz)
+    // THEN TAKING THE JENSEN-SHANNON DIVERGENCE BETWEEN BEAT SYNCHRONOUS SPECTRAL FRAMES
+
+    // IMPLEMENTATION (MOSTLY) FOLLOWS:
+    //  DAVIES AND PLUMBLEY "A SPECTRAL DIFFERENCE APPROACH TO EXTRACTING DOWNBEATS IN MUSICAL AUDIO"
+    //  EUSIPCO 2006, FLORENCE, ITALY
+
+    d_vec_t newspec(m_beatframesize / 2); // magnitude spectrum of current beat
+    d_vec_t oldspec(m_beatframesize / 2); // magnitude spectrum of previous beat
+
+    m_beatsd.clear();
+
+    if (audioLength == 0) return;
+
+    for (size_t i = 0; i + 1 < beats.size(); ++i) {
+
+        // Copy the extents of the current beat from downsampled array
+        // into beat frame buffer
+
+        size_t beatstart = (beats[i] * m_increment) / m_factor;
+        size_t beatend = (beats[i+1] * m_increment) / m_factor;
+        if (beatend >= audioLength) beatend = audioLength - 1;
+        if (beatend < beatstart) beatend = beatstart;
+        size_t beatlen = beatend - beatstart;
+
+        // Also apply a Hanning window to the beat frame buffer, sized
+        // to the beat extents rather than the frame size.  (Because
+        // the size varies, it's easier to do this by hand than use
+        // our Window abstraction.)
+
+//        std::cerr << "beatlen = " << beatlen << std::endl;
+
+//        float rms = 0;
+        for (size_t j = 0; j < beatlen && j < m_beatframesize; ++j) {
+            double mul = 0.5 * (1.0 - cos(TWO_PI * (double(j) / double(beatlen))));
+            m_beatframe[j] = audio[beatstart + j] * mul;
+//            rms += m_beatframe[j] * m_beatframe[j];
+        }
+//        rms = sqrt(rms);
+//        std::cerr << "beat " << i << ": audio rms " << rms << std::endl;
+
+        for (size_t j = beatlen; j < m_beatframesize; ++j) {
+            m_beatframe[j] = 0.0;
+        }
+
+        // Now FFT beat frame
+        
+        m_fft->process(false, m_beatframe, m_fftRealOut, m_fftImagOut);
+        
+        // Calculate magnitudes
+
+        for (size_t j = 0; j < m_beatframesize/2; ++j) {
+            newspec[j] = sqrt(m_fftRealOut[j] * m_fftRealOut[j] +
+                              m_fftImagOut[j] * m_fftImagOut[j]);
+        }
+
+        // Preserve peaks by applying adaptive threshold
+
+        MathUtilities::adaptiveThreshold(newspec);
+
+        // Calculate JS divergence between new and old spectral frames
+
+        if (i > 0) { // otherwise we have no previous frame
+            m_beatsd.push_back(measureSpecDiff(oldspec, newspec));
+//            std::cerr << "specdiff: " << m_beatsd[m_beatsd.size()-1] << std::endl;
+        }
+
+        // Copy newspec across to old
+
+        for (size_t j = 0; j < m_beatframesize/2; ++j) {
+            oldspec[j] = newspec[j];
+        }
+    }
+
+    // We now have all spectral difference measures in specdiff
+
+    int timesig = m_bpb;
+    if (timesig == 0) timesig = 4;
+
+    d_vec_t dbcand(timesig); // downbeat candidates
+
+    for (int beat = 0; beat < timesig; ++beat) {
+        dbcand[beat] = 0;
+    }
+
+   // look for beat transition which leads to greatest spectral change
+   for (int beat = 0; beat < timesig; ++beat) {
+       int count = 0;
+       for (int example = beat-1; example < (int)m_beatsd.size(); example += timesig) {
+           if (example < 0) continue;
+           dbcand[beat] += (m_beatsd[example]) / timesig;
+           ++count;
+       }
+       if (count > 0) dbcand[beat] /= count;
+//        std::cerr << "dbcand[" << beat << "] = " << dbcand[beat] << std::endl;
+   }
+
+    // first downbeat is beat at index of maximum value of dbcand
+    int dbind = MathUtilities::getMax(dbcand);
+
+    // remaining downbeats are at timesig intervals from the first
+    for (int i = dbind; i < (int)beats.size(); i += timesig) {
+        downbeats.push_back(i);
+    }
+}
+
+double
+DownBeat::measureSpecDiff(d_vec_t oldspec, d_vec_t newspec)
+{
+    // JENSEN-SHANNON DIVERGENCE BETWEEN SPECTRAL FRAMES
+
+    unsigned int SPECSIZE = 512;   // ONLY LOOK AT FIRST 512 SAMPLES OF SPECTRUM. 
+    if (SPECSIZE > oldspec.size()/4) {
+        SPECSIZE = oldspec.size()/4;
+    }
+    double SD = 0.;
+    double sd1 = 0.;
+
+    double sumnew = 0.;
+    double sumold = 0.;
+  
+    for (unsigned int i = 0;i < SPECSIZE;i++)
+    {
+        newspec[i] +=EPS;
+        oldspec[i] +=EPS;
+        
+        sumnew+=newspec[i];
+        sumold+=oldspec[i];
+    } 
+    
+    for (unsigned int i = 0;i < SPECSIZE;i++)
+    {
+        newspec[i] /= (sumnew);
+        oldspec[i] /= (sumold);
+        
+        // IF ANY SPECTRAL VALUES ARE 0 (SHOULDN'T BE ANY!) SET THEM TO 1
+        if (newspec[i] == 0)
+        {
+            newspec[i] = 1.;
+        }
+        
+        if (oldspec[i] == 0)
+        {
+            oldspec[i] = 1.;
+        }
+        
+        // JENSEN-SHANNON CALCULATION
+        sd1 = 0.5*oldspec[i] + 0.5*newspec[i];	
+        SD = SD + (-sd1*log(sd1)) + (0.5*(oldspec[i]*log(oldspec[i]))) + (0.5*(newspec[i]*log(newspec[i])));
+    }
+    
+    return SD;
+}
+
+void
+DownBeat::getBeatSD(vector<double> &beatsd) const
+{
+    for (int i = 0; i < (int)m_beatsd.size(); ++i) beatsd.push_back(m_beatsd[i]);
+}
+