Mercurial > hg > svcore
view data/fileio/test/AudioTestData.h @ 875:3e6ed8a8577b tonioni
Use a sparse time-value model only for outputs with fixed bin count of 1, not for those with unknown bin count.
(Precursor to using more than one model for outputs with unknown bin count)
author | Chris Cannam |
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date | Tue, 28 Jan 2014 18:52:22 +0000 |
parents | 02390a4c2abe |
children | a1cd5abcb38b |
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/* -*- 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> /** * 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(float 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() { float hpw = m_pulseWidth / 2.0; for (int i = 0; i < m_frameCount; ++i) { for (int c = 0; c < m_channelCount; ++c) { float s = 0.f; if (c == 0) { float phase = (i * m_sinFreq * 2.f * M_PI) / m_sampleRate; s = sinf(phase); } else if (c == 1) { int pulseNo = int((i * m_pulseFreq) / m_sampleRate); int index = (i * m_pulseFreq) - (m_sampleRate * pulseNo); if (index < m_pulseWidth) { s = 1.0 - fabsf(hpw - index) / hpw; if (pulseNo % 2) s = -s; } } else { s = c / 20.0; } m_data[i * m_channelCount + c] = s; } } } float *getInterleavedData() const { return m_data; } int getFrameCount() const { return m_frameCount; } int getChannelCount() const { return m_channelCount; } float getSampleRate () const { return m_sampleRate; } float getDuration() const { // seconds return m_duration; } private: float *m_data; int m_frameCount; int m_channelCount; float m_duration; float m_sampleRate; float m_sinFreq; float m_pulseFreq; float m_pulseWidth; }; #endif