diff data/fileio/test/AudioTestData.h @ 756:02390a4c2abe

Toward audio read tests

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/data/fileio/test/AudioTestData.h	Fri Mar 08 16:14:21 2013 +0000
@@ -0,0 +1,120 @@
+/* -*- 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
+