view data/fileio/test/AudioTestData.h @ 1196:c7b9c902642f spectrogram-minor-refactor

Fix threshold in spectrogram -- it wasn't working in the last release.
There is a new protocol for this. Formerly the threshold parameter had a
range from -50dB to 0 with the default at -50, and -50 treated internally
as "no threshold". However, there was a hardcoded, hidden internal threshold
for spectrogram colour mapping at -80dB with anything below this being rounded
to zero. Now the threshold parameter has range -81 to -1 with the default
at -80, -81 is treated internally as "no threshold", and there is no hidden
internal threshold. So the default behaviour is the same as before, an
effective -80dB threshold, but it is now possible to change this in both
directions. Sessions reloaded from prior versions may look slightly different
because, if the session says there should be no threshold, there will now
actually be no threshold instead of having the hidden internal one.

Still need to do something in the UI to make it apparent that the -81dB
setting removes the threshold entirely. This is at least no worse than the
previous, also obscured, magic -50dB setting.

/* -*- 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