annotate data/fileio/test/AudioTestData.h @ 1247:8f076d02569a piper

Make SVDEBUG always write to a log file -- formerly this was disabled in NDEBUG builds. I think there's little use to that, it just means that we keep adding more cerr debug output because we aren't getting the log we need. And SVDEBUG logging is not usually used in tight loops, I don't think the performance overhead is too serious.

Also update the About box.

/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */

/*
    Sonic Visualiser
    An audio file viewer and annotation editor.
    Centre for Digital Music, Queen Mary, University of London.
    This file copyright 2013 Chris Cannam.
    
    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.
*/

#ifndef AUDIO_TEST_DATA_H
#define AUDIO_TEST_DATA_H

#include <cmath>

#include "base/BaseTypes.h"

/**
 * Class that generates a single fixed test pattern to a given sample
 * rate and number of channels.
 *
 * The test pattern is two seconds long and consists of:
 *
 * -- in channel 0, a 600Hz sinusoid with peak amplitude 1.0
 *
 * -- in channel 1, four triangular forms with peaks at +1.0, -1.0,
 *    +1.0, -1.0 respectively, of 10ms width, starting at 0.0, 0.5,
 *    1.0 and 1.5 seconds; silence elsewhere
 *
 * -- in subsequent channels, a flat DC offset at +(channelNo / 20.0)
 */
class AudioTestData
{
public:
    AudioTestData(double rate, int channels) :
	m_channelCount(channels),
	m_duration(2.0),
	m_sampleRate(rate),
	m_sinFreq(600.0),
	m_pulseFreq(2)
    {
	m_frameCount = lrint(m_duration * m_sampleRate);
	m_data = new float[m_frameCount * m_channelCount];
	m_pulseWidth = 0.01 * m_sampleRate;
	generate();
    }

    ~AudioTestData() {
	delete[] m_data;
    }

    void generate() {

	double hpw = m_pulseWidth / 2.0;

	for (int i = 0; i < m_frameCount; ++i) {
	    for (int c = 0; c < m_channelCount; ++c) {

		double s = 0.0;

		if (c == 0) {

		    double phase = (i * m_sinFreq * 2.0 * M_PI) / m_sampleRate;
		    s = sin(phase);

		} else if (c == 1) {

		    int pulseNo = int((i * m_pulseFreq) / m_sampleRate);
		    int index = int(round((i * m_pulseFreq) -
                                          (m_sampleRate * pulseNo)));
		    if (index < m_pulseWidth) {
			s = 1.0 - fabs(hpw - index) / hpw;
			if (pulseNo % 2) s = -s;
		    }

		} else {

		    s = c / 20.0;
		}

		m_data[i * m_channelCount + c] = float(s);
	    }
	}
    }

    float *getInterleavedData() const {
	return m_data;
    }

    sv_frame_t getFrameCount() const { 
	return m_frameCount;
    }

    int getChannelCount() const {
	return m_channelCount;
    }

    sv_samplerate_t getSampleRate () const {
	return m_sampleRate;
    }

    double getDuration() const { // seconds
	return m_duration;
    }

private:
    float *m_data;
    sv_frame_t m_frameCount;
    int m_channelCount;
    double m_duration;
    sv_samplerate_t m_sampleRate;
    double m_sinFreq;
    double m_pulseFreq;
    double m_pulseWidth;
};

#endif