annotate dsp/tempotracking/DownBeat.h @ 73:dcb555b90924

* Key detector: when returning key strengths, use the peak value of the
  three underlying chromagram correlations (from 36-bin chromagram)
  corresponding to each key, instead of the mean.
  Rationale: This is the same method as used when returning the key value,
  and it's nice to have the same results in both returned value and plot.
  The peak performed better than the sum with a simple test set of triads,
  so it seems reasonable to change the plot to match the key output rather
  than the other way around.

* FFT: kiss_fftr returns only the non-conjugate bins, synthesise the rest
  rather than leaving them (perhaps dangerously) undefined.  Fixes an
  uninitialised data error in chromagram that could cause garbage results
  from key detector.

* Constant Q: remove precalculated values again, I reckon they're not
  proving such a good tradeoff.

/* -*- 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.
    All rights reserved.
*/

#ifndef DOWNBEAT_H
#define DOWNBEAT_H

#include <vector>

#include "dsp/rateconversion/Decimator.h"

using std::vector;

class FFTReal;

/**
 * This class takes an input audio signal and a sequence of beat
 * locations (calculated e.g. by TempoTrackV2) and estimates which of
 * the beat locations are downbeats (first beat of the bar).
 * 
 * The input audio signal is expected to have been downsampled to a
 * very low sampling rate (e.g. 2700Hz).  A utility function for
 * downsampling and buffering incoming block-by-block audio is
 * provided.
 */
class DownBeat
{
public:
    /**
     * Construct a downbeat locator that will operate on audio at the
     * downsampled by the given decimation factor from the given
     * original sample rate, plus beats extracted from the same audio
     * at the given original sample rate with the given frame
     * increment.
     *
     * decimationFactor must be a power of two no greater than 64, and
     * dfIncrement must be a multiple of decimationFactor.
     */
    DownBeat(float originalSampleRate,
             size_t decimationFactor,
             size_t dfIncrement);
    ~DownBeat();

    void setBeatsPerBar(int bpb);

    /**
     * Estimate which beats are down-beats.
     * 
     * audio contains the input audio stream after downsampling, and
     * audioLength contains the number of samples in this downsampled
     * stream.
     *
     * beats contains a series of beat positions expressed in
     * multiples of the df increment at the audio's original sample
     * rate, as described to the constructor.
     *
     * The returned downbeat array contains a series of indices to the
     * beats array.
     */
    void findDownBeats(const float *audio, // downsampled
                       size_t audioLength, // after downsampling
                       const vector<double> &beats,
                       vector<int> &downbeats);

    /**
     * Return the beat spectral difference function.  This is
     * calculated during findDownBeats, so this function can only be
     * meaningfully called after that has completed.  The returned
     * vector contains one value for each of the beat times passed in
     * to findDownBeats, less one.  Each value contains the spectral
     * difference between region prior to the beat's nominal position
     * and the region following it.
     */
    void getBeatSD(vector<double> &beatsd) const;
    
    /**
     * For your downsampling convenience: call this function
     * repeatedly with input audio blocks containing dfIncrement
     * samples at the original sample rate, to decimate them to the
     * downsampled rate and buffer them within the DownBeat class.
     *     
     * Call getBufferedAudio() to retrieve the results after all
     * blocks have been processed.
     */
    void pushAudioBlock(const float *audio);
    
    /**
     * Retrieve the accumulated audio produced by pushAudioBlock calls.
     */
    const float *getBufferedAudio(size_t &length) const;

    /**
     * Clear any buffered downsampled audio data.
     */
    void resetAudioBuffer();

private:
    typedef vector<int> i_vec_t;
    typedef vector<vector<int> > i_mat_t;
    typedef vector<double> d_vec_t;
    typedef vector<vector<double> > d_mat_t;

    void makeDecimators();
    double measureSpecDiff(d_vec_t oldspec, d_vec_t newspec);

    int m_bpb;
    float m_rate;
    size_t m_factor;
    size_t m_increment;
    Decimator *m_decimator1;
    Decimator *m_decimator2;
    float *m_buffer;
    float *m_decbuf;
    size_t m_bufsiz;
    size_t m_buffill;
    size_t m_beatframesize;
    double *m_beatframe;
    FFTReal *m_fft;
    double *m_fftRealOut;
    double *m_fftImagOut;
    d_vec_t m_beatsd;
};

#endif