annotate dsp/tempotracking/DownBeat.h @ 209:ccd2019190bf msvc

Some MSVC fixes, including (temporarily, probably) renaming the FFT source file to avoid getting it mixed up with the Vamp SDK one in our object dir
author Chris Cannam
date Thu, 01 Feb 2018 16:34:08 +0000
parents 915365c58758
children 701233f8ed41
rev   line source
cannam@54 1 /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
cannam@54 2
cannam@54 3 /*
cannam@54 4 QM DSP Library
cannam@54 5
cannam@54 6 Centre for Digital Music, Queen Mary, University of London.
cannam@54 7 This file copyright 2008-2009 Matthew Davies and QMUL.
Chris@84 8
Chris@84 9 This program is free software; you can redistribute it and/or
Chris@84 10 modify it under the terms of the GNU General Public License as
Chris@84 11 published by the Free Software Foundation; either version 2 of the
Chris@84 12 License, or (at your option) any later version. See the file
Chris@84 13 COPYING included with this distribution for more information.
cannam@54 14 */
cannam@54 15
cannam@54 16 #ifndef DOWNBEAT_H
cannam@54 17 #define DOWNBEAT_H
cannam@54 18
cannam@54 19 #include <vector>
Chris@98 20 #include <cstddef>
cannam@54 21
cannam@54 22 #include "dsp/rateconversion/Decimator.h"
cannam@54 23
cannam@54 24 using std::vector;
cannam@54 25
cannam@64 26 class FFTReal;
cannam@64 27
cannam@54 28 /**
cannam@54 29 * This class takes an input audio signal and a sequence of beat
cannam@54 30 * locations (calculated e.g. by TempoTrackV2) and estimates which of
cannam@54 31 * the beat locations are downbeats (first beat of the bar).
cannam@54 32 *
cannam@54 33 * The input audio signal is expected to have been downsampled to a
cannam@54 34 * very low sampling rate (e.g. 2700Hz). A utility function for
cannam@54 35 * downsampling and buffering incoming block-by-block audio is
cannam@54 36 * provided.
cannam@54 37 */
cannam@54 38 class DownBeat
cannam@54 39 {
cannam@54 40 public:
cannam@54 41 /**
cannam@54 42 * Construct a downbeat locator that will operate on audio at the
cannam@54 43 * downsampled by the given decimation factor from the given
cannam@54 44 * original sample rate, plus beats extracted from the same audio
cannam@54 45 * at the given original sample rate with the given frame
cannam@54 46 * increment.
cannam@54 47 *
cannam@54 48 * decimationFactor must be a power of two no greater than 64, and
cannam@54 49 * dfIncrement must be a multiple of decimationFactor.
cannam@54 50 */
cannam@54 51 DownBeat(float originalSampleRate,
cannam@54 52 size_t decimationFactor,
cannam@54 53 size_t dfIncrement);
cannam@54 54 ~DownBeat();
cannam@54 55
cannam@55 56 void setBeatsPerBar(int bpb);
cannam@55 57
cannam@54 58 /**
cannam@54 59 * Estimate which beats are down-beats.
cannam@54 60 *
cannam@54 61 * audio contains the input audio stream after downsampling, and
cannam@54 62 * audioLength contains the number of samples in this downsampled
cannam@54 63 * stream.
cannam@54 64 *
cannam@54 65 * beats contains a series of beat positions expressed in
cannam@54 66 * multiples of the df increment at the audio's original sample
cannam@54 67 * rate, as described to the constructor.
cannam@54 68 *
cannam@54 69 * The returned downbeat array contains a series of indices to the
cannam@54 70 * beats array.
cannam@54 71 */
cannam@55 72 void findDownBeats(const float *audio, // downsampled
cannam@54 73 size_t audioLength, // after downsampling
cannam@54 74 const vector<double> &beats,
cannam@54 75 vector<int> &downbeats);
cannam@56 76
cannam@56 77 /**
cannam@56 78 * Return the beat spectral difference function. This is
cannam@56 79 * calculated during findDownBeats, so this function can only be
cannam@56 80 * meaningfully called after that has completed. The returned
cannam@56 81 * vector contains one value for each of the beat times passed in
cannam@56 82 * to findDownBeats, less one. Each value contains the spectral
cannam@56 83 * difference between region prior to the beat's nominal position
cannam@56 84 * and the region following it.
cannam@56 85 */
cannam@56 86 void getBeatSD(vector<double> &beatsd) const;
cannam@54 87
cannam@54 88 /**
cannam@54 89 * For your downsampling convenience: call this function
cannam@54 90 * repeatedly with input audio blocks containing dfIncrement
cannam@54 91 * samples at the original sample rate, to decimate them to the
cannam@54 92 * downsampled rate and buffer them within the DownBeat class.
cannam@54 93 *
cannam@54 94 * Call getBufferedAudio() to retrieve the results after all
cannam@54 95 * blocks have been processed.
cannam@54 96 */
cannam@55 97 void pushAudioBlock(const float *audio);
cannam@54 98
cannam@54 99 /**
cannam@54 100 * Retrieve the accumulated audio produced by pushAudioBlock calls.
cannam@54 101 */
cannam@55 102 const float *getBufferedAudio(size_t &length) const;
cannam@55 103
cannam@55 104 /**
cannam@55 105 * Clear any buffered downsampled audio data.
cannam@55 106 */
cannam@55 107 void resetAudioBuffer();
cannam@54 108
cannam@54 109 private:
cannam@54 110 typedef vector<int> i_vec_t;
cannam@54 111 typedef vector<vector<int> > i_mat_t;
cannam@54 112 typedef vector<double> d_vec_t;
cannam@54 113 typedef vector<vector<double> > d_mat_t;
cannam@54 114
cannam@54 115 void makeDecimators();
cannam@54 116 double measureSpecDiff(d_vec_t oldspec, d_vec_t newspec);
cannam@54 117
cannam@55 118 int m_bpb;
cannam@54 119 float m_rate;
cannam@54 120 size_t m_factor;
cannam@54 121 size_t m_increment;
cannam@54 122 Decimator *m_decimator1;
cannam@54 123 Decimator *m_decimator2;
cannam@55 124 float *m_buffer;
cannam@55 125 float *m_decbuf;
cannam@54 126 size_t m_bufsiz;
cannam@54 127 size_t m_buffill;
cannam@54 128 size_t m_beatframesize;
cannam@54 129 double *m_beatframe;
cannam@64 130 FFTReal *m_fft;
cannam@54 131 double *m_fftRealOut;
cannam@54 132 double *m_fftImagOut;
cannam@56 133 d_vec_t m_beatsd;
cannam@54 134 };
cannam@54 135
cannam@54 136 #endif