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1 /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
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2
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3 /*
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4 Sonic Visualiser
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5 An audio file viewer and annotation editor.
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6 Centre for Digital Music, Queen Mary, University of London.
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7 This file copyright 2013 Chris Cannam.
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8
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9 This program is free software; you can redistribute it and/or
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10 modify it under the terms of the GNU General Public License as
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11 published by the Free Software Foundation; either version 2 of the
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12 License, or (at your option) any later version. See the file
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13 COPYING included with this distribution for more information.
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14 */
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15
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16 #ifndef TEST_AUDIO_FILE_READER_H
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17 #define TEST_AUDIO_FILE_READER_H
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18
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19 #include "../AudioFileReaderFactory.h"
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20 #include "../AudioFileReader.h"
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21 #include "../WavFileWriter.h"
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22
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23 #include "AudioTestData.h"
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24
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25 #include <cmath>
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26
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27 #include <QObject>
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28 #include <QtTest>
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29 #include <QDir>
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30
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31 #include <iostream>
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32
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33 using namespace std;
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34
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35 static QString audioDir = "svcore/data/fileio/test/testfiles";
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36 static QString diffDir = "svcore/data/fileio/test/diffs";
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37
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38 class AudioFileReaderTest : public QObject
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39 {
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40 Q_OBJECT
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41
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42 const char *strOf(QString s) {
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43 return strdup(s.toLocal8Bit().data());
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44 }
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45
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46 void getFileMetadata(QString filename,
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47 QString &extension,
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48 sv_samplerate_t &rate,
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49 int &channels,
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50 int &bitdepth) {
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51
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52 QStringList fileAndExt = filename.split(".");
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53 QStringList bits = fileAndExt[0].split("-");
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54
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55 extension = fileAndExt[1];
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56 rate = bits[0].toInt();
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57 channels = bits[1].toInt();
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58 bitdepth = 16;
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59 if (bits.length() > 2) {
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60 bitdepth = bits[2].toInt();
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61 }
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62 }
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63
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64 void getExpectedThresholds(QString format,
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65 QString filename,
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66 bool resampled,
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67 bool gapless,
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68 bool normalised,
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69 double &maxLimit,
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70 double &rmsLimit) {
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71
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72 QString extension;
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73 sv_samplerate_t fileRate;
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74 int channels;
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75 int bitdepth;
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76 getFileMetadata(filename, extension, fileRate, channels, bitdepth);
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77
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78 if (normalised) {
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79
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80 if (format == "ogg") {
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81
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82 // Our ogg is not especially high quality and is
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83 // actually further from the original if normalised
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84
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85 maxLimit = 0.1;
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86 rmsLimit = 0.03;
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87
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88 } else if (format == "aac") {
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89
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90 // Terrible performance for this test, load of spill
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91 // from one channel to the other. I guess they know
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92 // what they're doing, it's perceptual after all, but
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93 // it does make this check a bit superfluous, you
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94 // could probably pass it with a signal that sounds
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95 // nothing like the original
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96 maxLimit = 0.2;
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97 rmsLimit = 0.1;
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98
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99 } else if (format == "mp3") {
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100
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101 if (resampled && !gapless) {
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102
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103 // We expect worse figures here, because the
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104 // combination of uncompensated encoder delay +
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105 // resampling results in a fractional delay which
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106 // means the decoded signal is slightly out of
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107 // phase compared to the test signal
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108
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109 maxLimit = 0.1;
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110 rmsLimit = 0.05;
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111
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112 } else {
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113
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114 maxLimit = 0.05;
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115 rmsLimit = 0.01;
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116 }
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117
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118 } else {
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119
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120 // lossless formats (wav, aiff, flac, apple_lossless)
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121
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122 if (bitdepth >= 16 && !resampled) {
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123 maxLimit = 1e-3;
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124 rmsLimit = 3e-4;
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125 } else {
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126 maxLimit = 0.01;
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127 rmsLimit = 5e-3;
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128 }
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129 }
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130
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131 } else { // !normalised
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132
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133 if (format == "ogg") {
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134
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135 maxLimit = 0.06;
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136 rmsLimit = 0.03;
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137
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138 } else if (format == "aac") {
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139
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140 maxLimit = 0.1;
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141 rmsLimit = 0.1;
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142
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143 } else if (format == "mp3") {
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144
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145 // all mp3 figures are worse when not normalising
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146 maxLimit = 0.1;
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147 rmsLimit = 0.05;
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148
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149 } else {
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150
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151 // lossless formats (wav, aiff, flac, apple_lossless)
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152
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153 if (bitdepth >= 16 && !resampled) {
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154 maxLimit = 1e-3;
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155 rmsLimit = 3e-4;
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156 } else {
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157 maxLimit = 0.02;
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158 rmsLimit = 0.01;
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159 }
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160 }
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161 }
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162 }
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163
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164 QString testName(QString format, QString filename, int rate, bool norm, bool gapless) {
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165 return QString("%1/%2 at %3%4%5")
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166 .arg(format)
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167 .arg(filename)
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168 .arg(rate)
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169 .arg(norm ? " normalised": "")
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170 .arg(gapless ? "" : " non-gapless");
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171 }
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172
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173 private slots:
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174 void init()
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175 {
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176 if (!QDir(audioDir).exists()) {
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177 cerr << "ERROR: Audio test file directory \"" << audioDir << "\" does not exist" << endl;
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178 QVERIFY2(QDir(audioDir).exists(), "Audio test file directory not found");
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179 }
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180 if (!QDir(diffDir).exists() && !QDir().mkpath(diffDir)) {
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181 cerr << "ERROR: Audio diff directory \"" << diffDir << "\" does not exist and could not be created" << endl;
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182 QVERIFY2(QDir(diffDir).exists(), "Audio diff directory not found and could not be created");
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183 }
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184 }
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185
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186 void read_data()
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187 {
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188 QTest::addColumn<QString>("format");
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189 QTest::addColumn<QString>("audiofile");
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190 QTest::addColumn<int>("rate");
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191 QTest::addColumn<bool>("normalised");
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192 QTest::addColumn<bool>("gapless");
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193 QStringList dirs = QDir(audioDir).entryList(QDir::Dirs |
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194 QDir::NoDotAndDotDot);
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195 for (QString format: dirs) {
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196 QStringList files = QDir(QDir(audioDir).filePath(format))
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197 .entryList(QDir::Files);
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198 int readRates[] = { 44100, 48000 };
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199 bool norms[] = { false, true };
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200 bool gaplesses[] = { true, false };
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201 foreach (QString filename, files) {
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202 for (int rate: readRates) {
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203 for (bool norm: norms) {
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204 for (bool gapless: gaplesses) {
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205
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206 if (format != "mp3" && !gapless) {
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207 continue;
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208 }
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209
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210 QString desc = testName
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211 (format, filename, rate, norm, gapless);
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212
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213 QTest::newRow(strOf(desc))
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214 << format << filename << rate << norm << gapless;
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215 }
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216 }
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217 }
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218 }
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219 }
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220 }
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221
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222 void read()
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223 {
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224 QFETCH(QString, format);
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225 QFETCH(QString, audiofile);
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226 QFETCH(int, rate);
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227 QFETCH(bool, normalised);
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228 QFETCH(bool, gapless);
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229
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230 sv_samplerate_t readRate(rate);
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231
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232 // cerr << "\naudiofile = " << audiofile << endl;
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233
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234 AudioFileReaderFactory::Parameters params;
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235 params.targetRate = readRate;
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236 params.normalisation = (normalised ?
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237 AudioFileReaderFactory::Normalisation::Peak :
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238 AudioFileReaderFactory::Normalisation::None);
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239 params.gaplessMode = (gapless ?
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240 AudioFileReaderFactory::GaplessMode::Gapless :
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241 AudioFileReaderFactory::GaplessMode::Gappy);
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242
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243 AudioFileReader *reader =
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244 AudioFileReaderFactory::createReader
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245 (audioDir + "/" + format + "/" + audiofile, params);
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246
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247 if (!reader) {
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248 #if ( QT_VERSION >= 0x050000 )
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249 QSKIP("Unsupported file, skipping");
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250 #else
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251 QSKIP("Unsupported file, skipping", SkipSingle);
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252 #endif
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253 }
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254
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255 QString extension;
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256 sv_samplerate_t fileRate;
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257 int channels;
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258 int fileBitdepth;
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259 getFileMetadata(audiofile, extension, fileRate, channels, fileBitdepth);
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260
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261 QCOMPARE((int)reader->getChannelCount(), channels);
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262 QCOMPARE(reader->getNativeRate(), fileRate);
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263 QCOMPARE(reader->getSampleRate(), readRate);
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264
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265 AudioTestData tdata(readRate, channels);
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266
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267 float *reference = tdata.getInterleavedData();
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268 sv_frame_t refFrames = tdata.getFrameCount();
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269
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270 // The reader should give us exactly the expected number of
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271 // frames, except for mp3/aac files. We ask for quite a lot
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272 // more, though, so we can (a) check that we only get the
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273 // expected number back (if this is not mp3/aac) or (b) take
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274 // into account silence at beginning and end (if it is).
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275 floatvec_t test = reader->getInterleavedFrames(0, refFrames + 5000);
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276 sv_frame_t read = test.size() / channels;
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277
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278 bool perceptual = (extension == "mp3" ||
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279 extension == "aac" ||
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280 extension == "m4a");
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281
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282 if (perceptual && !gapless) {
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283 // allow silence at start and end
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284 QVERIFY(read >= refFrames);
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285 } else {
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286 QCOMPARE(read, refFrames);
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287 }
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288
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289 bool resampled = readRate != fileRate;
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290 double maxLimit, rmsLimit;
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291 getExpectedThresholds(format,
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292 audiofile,
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293 resampled,
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294 gapless,
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295 normalised,
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296 maxLimit, rmsLimit);
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297
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298 double edgeLimit = maxLimit * 3; // in first or final edgeSize frames
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299 if (resampled && edgeLimit < 0.1) edgeLimit = 0.1;
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300 int edgeSize = 100;
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301
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302 // And we ignore completely the last few frames when upsampling
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303 int discard = 1 + int(round(readRate / fileRate));
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304
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305 int offset = 0;
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306
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307 if (perceptual) {
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308
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309 // Look for an initial offset.
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310 //
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311 // We know the first channel has a sinusoid in it. It
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312 // should have a peak at 0.4ms (see AudioTestData.h) but
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313 // that might have been clipped, which would make it
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314 // imprecise. We can tell if it's clipped, though, as
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315 // there will be samples having exactly identical
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316 // values. So what we look for is the peak if it's not
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317 // clipped and, if it is, the first zero crossing after
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318 // the peak, which should be at 0.8ms.
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319
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320 int expectedPeak = int(0.0004 * readRate);
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321 int expectedZC = int(0.0008 * readRate);
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322 bool foundPeak = false;
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cannam@1314
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323 for (int i = 1; i+1 < read; ++i) {
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324 float prevSample = test[(i-1) * channels];
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325 float thisSample = test[i * channels];
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cannam@1314
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326 float nextSample = test[(i+1) * channels];
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cannam@1314
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327 if (thisSample > 0.8 && nextSample < thisSample) {
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328 foundPeak = true;
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329 if (thisSample > prevSample) {
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330 // not clipped
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331 offset = i - expectedPeak - 1;
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cannam@1314
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332 break;
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333 }
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cannam@1314
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334 }
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cannam@1314
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335 if (foundPeak && (thisSample >= 0.0 && nextSample < 0.0)) {
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cannam@1315
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336 // cerr << "thisSample = " << thisSample << ", nextSample = "
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337 // << nextSample << endl;
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338 offset = i - expectedZC - 1;
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339 break;
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340 }
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341 }
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342
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cannam@1315
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343 // int fileRateEquivalent = int((offset / readRate) * fileRate);
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cannam@1315
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344 // std::cerr << "offset = " << offset << std::endl;
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345 // std::cerr << "at file rate would be " << fileRateEquivalent << std::endl;
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346
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347 // Previously our m4a test file had a fixed offset of 1024
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348 // at the file sample rate -- this may be because it was
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349 // produced by FAAC which did not write in the delay as
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350 // metadata? We now have an m4a produced by Core Audio
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351 // which gives a 0 offset. What to do...
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352
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353 // Anyway, mp3s should have 0 offset in gapless mode and
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354 // "something else" otherwise.
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355
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356 if (gapless) {
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cannam@1315
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357 if (format == "aac") {
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cannam@1315
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358 // ouch!
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cannam@1315
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359 if (offset == -1) offset = 0;
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360 }
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361 QCOMPARE(offset, 0);
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362 }
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363 }
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364
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cannam@1315
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365 {
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cannam@1315
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366 // Write the diff file now, so that it's already been written
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cannam@1315
|
367 // even if the comparison fails. We aren't checking anything
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cannam@1315
|
368 // here except as necessary to avoid buffer overruns etc
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cannam@1315
|
369
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cannam@1315
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370 QString diffFile =
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cannam@1315
|
371 testName(format, audiofile, rate, normalised, gapless);
|
cannam@1315
|
372 diffFile.replace("/", "_");
|
cannam@1315
|
373 diffFile.replace(".", "_");
|
cannam@1315
|
374 diffFile.replace(" ", "_");
|
cannam@1315
|
375 diffFile += ".wav";
|
cannam@1315
|
376 diffFile = QDir(diffDir).filePath(diffFile);
|
cannam@1315
|
377 WavFileWriter diffWriter(diffFile, readRate, channels,
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cannam@1315
|
378 WavFileWriter::WriteToTarget); //!!! NB WriteToTemporary not working, why?
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cannam@1315
|
379 QVERIFY(diffWriter.isOK());
|
cannam@1315
|
380
|
cannam@1315
|
381 vector<vector<float>> diffs(channels);
|
cannam@1315
|
382 for (int c = 0; c < channels; ++c) {
|
cannam@1315
|
383 for (int i = 0; i < refFrames; ++i) {
|
cannam@1315
|
384 int ix = i + offset;
|
cannam@1315
|
385 if (ix < read) {
|
cannam@1315
|
386 float signeddiff =
|
cannam@1315
|
387 test[ix * channels + c] -
|
cannam@1315
|
388 reference[i * channels + c];
|
cannam@1315
|
389 diffs[c].push_back(signeddiff);
|
cannam@1315
|
390 }
|
cannam@1315
|
391 }
|
cannam@1315
|
392 }
|
cannam@1315
|
393 float **ptrs = new float*[channels];
|
cannam@1315
|
394 for (int c = 0; c < channels; ++c) {
|
cannam@1315
|
395 ptrs[c] = diffs[c].data();
|
cannam@1315
|
396 }
|
cannam@1315
|
397 diffWriter.writeSamples(ptrs, refFrames);
|
cannam@1315
|
398 delete[] ptrs;
|
cannam@1315
|
399 }
|
Chris@1313
|
400
|
Chris@756
|
401 for (int c = 0; c < channels; ++c) {
|
Chris@1313
|
402
|
Chris@1313
|
403 double maxDiff = 0.0;
|
Chris@1313
|
404 double totalDiff = 0.0;
|
Chris@1313
|
405 double totalSqrDiff = 0.0;
|
Chris@1313
|
406 int maxIndex = 0;
|
Chris@1313
|
407
|
Chris@1296
|
408 for (int i = 0; i < refFrames; ++i) {
|
Chris@1296
|
409 int ix = i + offset;
|
Chris@1296
|
410 if (ix >= read) {
|
cannam@1308
|
411 cerr << "ERROR: audiofile " << audiofile << " reads truncated (read-rate reference frames " << i << " onward, of " << refFrames << ", are lost)" << endl;
|
Chris@1296
|
412 QVERIFY(ix < read);
|
Chris@1296
|
413 }
|
Chris@1313
|
414
|
Chris@1296
|
415 if (ix + discard >= read) {
|
Chris@1296
|
416 // we forgive the very edge samples when
|
Chris@1296
|
417 // resampling (discard > 0)
|
Chris@1296
|
418 continue;
|
Chris@1296
|
419 }
|
Chris@1313
|
420
|
cannam@1315
|
421 double diff = fabs(test[ix * channels + c] -
|
cannam@1315
|
422 reference[i * channels + c]);
|
Chris@1313
|
423
|
Chris@1313
|
424 totalDiff += diff;
|
Chris@1313
|
425 totalSqrDiff += diff * diff;
|
Chris@1313
|
426
|
Chris@757
|
427 // in edge areas, record this only if it exceeds edgeLimit
|
Chris@1313
|
428 if (i < edgeSize || i + edgeSize >= refFrames) {
|
Chris@1313
|
429 if (diff > edgeLimit && diff > maxDiff) {
|
Chris@1313
|
430 maxDiff = diff;
|
Chris@1313
|
431 maxIndex = i;
|
Chris@757
|
432 }
|
Chris@757
|
433 } else {
|
Chris@1313
|
434 if (diff > maxDiff) {
|
Chris@1313
|
435 maxDiff = diff;
|
Chris@1313
|
436 maxIndex = i;
|
Chris@757
|
437 }
|
Chris@756
|
438 }
|
Chris@756
|
439 }
|
Chris@1313
|
440
|
Chris@1313
|
441 double meanDiff = totalDiff / double(refFrames);
|
Chris@1313
|
442 double rmsDiff = sqrt(totalSqrDiff / double(refFrames));
|
cannam@1308
|
443
|
cannam@1314
|
444 /*
|
Chris@1313
|
445 cerr << "channel " << c << ": mean diff " << meanDiff << endl;
|
Chris@1313
|
446 cerr << "channel " << c << ": rms diff " << rmsDiff << endl;
|
Chris@1313
|
447 cerr << "channel " << c << ": max diff " << maxDiff << " at " << maxIndex << endl;
|
cannam@1314
|
448 */
|
Chris@1313
|
449 if (rmsDiff >= rmsLimit) {
|
Chris@1313
|
450 cerr << "ERROR: for audiofile " << audiofile << ": RMS diff = " << rmsDiff << " for channel " << c << " (limit = " << rmsLimit << ")" << endl;
|
Chris@1313
|
451 QVERIFY(rmsDiff < rmsLimit);
|
Chris@1313
|
452 }
|
Chris@1313
|
453 if (maxDiff >= maxLimit) {
|
Chris@1313
|
454 cerr << "ERROR: for audiofile " << audiofile << ": max diff = " << maxDiff << " at frame " << maxIndex << " of " << read << " on channel " << c << " (limit = " << maxLimit << ", edge limit = " << edgeLimit << ", mean diff = " << meanDiff << ", rms = " << rmsDiff << ")" << endl;
|
Chris@1313
|
455 QVERIFY(maxDiff < maxLimit);
|
Chris@1313
|
456 }
|
Chris@1313
|
457
|
Chris@1313
|
458 // and check for spurious material at end
|
Chris@1313
|
459
|
Chris@1309
|
460 for (sv_frame_t i = refFrames; i + offset < read; ++i) {
|
Chris@1309
|
461 sv_frame_t ix = i + offset;
|
Chris@1323
|
462 float quiet = 0.1f; //!!! allow some ringing - but let's come back to this, it should tail off
|
cannam@1308
|
463 float mag = fabsf(test[ix * channels + c]);
|
cannam@1308
|
464 if (mag > quiet) {
|
Chris@1313
|
465 cerr << "ERROR: audiofile " << audiofile << " contains spurious data after end of reference (found sample " << test[ix * channels + c] << " at index " << ix << " of channel " << c << " after reference+offset ended at " << refFrames+offset << ")" << endl;
|
cannam@1308
|
466 QVERIFY(mag < quiet);
|
cannam@1308
|
467 }
|
cannam@1308
|
468 }
|
Chris@756
|
469 }
|
Chris@756
|
470 }
|
Chris@756
|
471 };
|
Chris@756
|
472
|
Chris@756
|
473 #endif
|