Mercurial > hg > svcore
view data/fileio/test/AudioTestData.h @ 1116:7f8eb7379280 3.0-integration
Necessary include
author | Chris Cannam |
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date | Tue, 04 Aug 2015 10:49:24 +0100 |
parents | a1cd5abcb38b |
children | 48e9f538e6e9 |
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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> #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