diff dsp/tempotracking/DownBeat.h @ 279:c8908cdc8c32

* First cut at Matthew's downbeat estimator -- untested so far

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/dsp/tempotracking/DownBeat.h	Tue Feb 10 12:52:43 2009 +0000
@@ -0,0 +1,107 @@
+/* -*- 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;
+
+/**
+ * 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();
+
+    /**
+     * 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 double *audio, // downsampled
+                       size_t audioLength, // after downsampling
+                       const vector<double> &beats,
+                       vector<int> &downbeats);
+    
+    /**
+     * 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 double *audio);
+    
+    /**
+     * Retrieve the accumulated audio produced by pushAudioBlock calls.
+     */
+    const double *getBufferedAudio(size_t &length) const;
+
+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);
+
+    float m_rate;
+    size_t m_factor;
+    size_t m_increment;
+    Decimator *m_decimator1;
+    Decimator *m_decimator2;
+    double *m_buffer;
+    double *m_decbuf;
+    size_t m_bufsiz;
+    size_t m_buffill;
+    size_t m_beatframesize;
+    double *m_beatframe;
+    double *m_fftRealOut;
+    double *m_fftImagOut;
+};
+
+#endif