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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 2006-2007 Chris Cannam and QMUL.
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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 #include "CodedAudioFileReader.h"
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17
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18 #include "WavFileReader.h"
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19 #include "base/TempDirectory.h"
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20 #include "base/Exceptions.h"
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21 #include "base/Profiler.h"
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22 #include "base/Serialiser.h"
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23 #include "base/Resampler.h"
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24 #include "base/StorageAdviser.h"
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25
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26 #include <stdint.h>
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27 #include <iostream>
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28 #include <QDir>
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29 #include <QMutexLocker>
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30
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31 using namespace std;
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32
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33 CodedAudioFileReader::CodedAudioFileReader(CacheMode cacheMode,
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34 sv_samplerate_t targetRate,
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35 bool normalised) :
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36 m_cacheMode(cacheMode),
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37 m_initialised(false),
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38 m_serialiser(0),
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39 m_fileRate(0),
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40 m_cacheFileWritePtr(0),
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41 m_cacheFileReader(0),
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42 m_cacheWriteBuffer(0),
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43 m_cacheWriteBufferIndex(0),
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44 m_cacheWriteBufferSize(16384),
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45 m_resampler(0),
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46 m_resampleBuffer(0),
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47 m_fileFrameCount(0),
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48 m_normalised(normalised),
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49 m_max(0.f),
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50 m_gain(1.f),
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51 m_clippedCount(0),
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52 m_firstNonzero(0),
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53 m_lastNonzero(0)
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54 {
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55 SVDEBUG << "CodedAudioFileReader:: cache mode: " << cacheMode
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56 << " (" << (cacheMode == CacheInTemporaryFile
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57 ? "CacheInTemporaryFile" : "CacheInMemory") << ")"
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58 << ", rate: " << targetRate
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59 << (targetRate == 0 ? " (use source rate)" : "")
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60 << ", normalised: " << normalised << endl;
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61
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62 m_frameCount = 0;
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63 m_sampleRate = targetRate;
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64 }
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65
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66 CodedAudioFileReader::~CodedAudioFileReader()
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67 {
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68 QMutexLocker locker(&m_cacheMutex);
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69
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70 if (m_serialiser) endSerialised();
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71
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72 if (m_cacheFileWritePtr) sf_close(m_cacheFileWritePtr);
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73
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74 SVDEBUG << "CodedAudioFileReader::~CodedAudioFileReader: deleting cache file reader" << endl;
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75
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76 delete m_cacheFileReader;
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77 delete[] m_cacheWriteBuffer;
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78
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79 if (m_cacheFileName != "") {
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80 SVDEBUG << "CodedAudioFileReader::~CodedAudioFileReader: deleting cache file " << m_cacheFileName << endl;
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81 if (!QFile(m_cacheFileName).remove()) {
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82 SVDEBUG << "WARNING: CodedAudioFileReader::~CodedAudioFileReader: Failed to delete cache file \"" << m_cacheFileName << "\"" << endl;
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83 }
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84 }
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85
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86 delete m_resampler;
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87 delete[] m_resampleBuffer;
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88
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89 if (!m_data.empty()) {
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90 StorageAdviser::notifyDoneAllocation
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91 (StorageAdviser::MemoryAllocation,
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92 (m_data.size() * sizeof(float)) / 1024);
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93 }
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94 }
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95
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96 void
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97 CodedAudioFileReader::startSerialised(QString id)
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98 {
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99 SVDEBUG << "CodedAudioFileReader(" << this << ")::startSerialised: id = " << id << endl;
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100
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101 delete m_serialiser;
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102 m_serialiser = new Serialiser(id);
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103 }
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104
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105 void
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106 CodedAudioFileReader::endSerialised()
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107 {
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108 SVDEBUG << "CodedAudioFileReader(" << this << ")::endSerialised: id = " << (m_serialiser ? m_serialiser->getId() : "(none)") << endl;
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109
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110 delete m_serialiser;
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111 m_serialiser = 0;
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112 }
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113
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114 void
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115 CodedAudioFileReader::initialiseDecodeCache()
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116 {
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117 QMutexLocker locker(&m_cacheMutex);
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118
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119 SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: file rate = " << m_fileRate << endl;
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120
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121 if (m_fileRate == 0) {
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122 SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: ERROR: File sample rate unknown (bug in subclass implementation?)" << endl;
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123 throw FileOperationFailed("(coded file)", "File sample rate unknown (bug in subclass implementation?)");
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124 }
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125 if (m_sampleRate == 0) {
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126 m_sampleRate = m_fileRate;
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127 SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: rate (from file) = " << m_fileRate << endl;
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128 }
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129 if (m_fileRate != m_sampleRate) {
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130 SVDEBUG << "CodedAudioFileReader: resampling " << m_fileRate << " -> " << m_sampleRate << endl;
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131 m_resampler = new Resampler(Resampler::FastestTolerable,
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132 m_channelCount,
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133 m_cacheWriteBufferSize);
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134 double ratio = m_sampleRate / m_fileRate;
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135 m_resampleBuffer = new float
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136 [lrint(ceil(double(m_cacheWriteBufferSize) * m_channelCount * ratio + 1))];
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137 }
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138
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139 m_cacheWriteBuffer = new float[m_cacheWriteBufferSize * m_channelCount];
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140 m_cacheWriteBufferIndex = 0;
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141
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142 if (m_cacheMode == CacheInTemporaryFile) {
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143
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144 try {
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145 QDir dir(TempDirectory::getInstance()->getPath());
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146 m_cacheFileName = dir.filePath(QString("decoded_%1.wav")
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147 .arg((intptr_t)this));
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148
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149 SF_INFO fileInfo;
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150 int fileRate = int(round(m_sampleRate));
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151 if (m_sampleRate != sv_samplerate_t(fileRate)) {
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152 SVDEBUG << "CodedAudioFileReader: WARNING: Non-integer sample rate "
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153 << m_sampleRate << " presented for writing, rounding to " << fileRate
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154 << endl;
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155 }
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156 fileInfo.samplerate = fileRate;
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157 fileInfo.channels = m_channelCount;
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158
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159 // Previously we were writing SF_FORMAT_PCM_16 and in a
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160 // comment I wrote: "No point in writing 24-bit or float;
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161 // generally this class is used for decoding files that
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162 // have come from a 16 bit source or that decode to only
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163 // 16 bits anyway." That was naive -- we want to preserve
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164 // the original values to the same float precision that we
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165 // use internally. Saving PCM_16 obviously doesn't
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166 // preserve values for sources at bit depths greater than
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167 // 16, but it also doesn't always do so for sources at bit
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168 // depths less than 16.
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169 //
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170 // (This came to light with a bug in libsndfile 1.0.26,
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171 // which always reports every file as non-seekable, so
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172 // that coded readers were being used even for WAV
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173 // files. This changed the values that came from PCM_8 WAV
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174 // sources, breaking Sonic Annotator's output comparison
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175 // tests.)
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176 //
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177 // So: now we write floats.
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178 fileInfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT;
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179
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180 m_cacheFileWritePtr = sf_open(m_cacheFileName.toLocal8Bit(),
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181 SFM_WRITE, &fileInfo);
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182
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183 if (m_cacheFileWritePtr) {
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184
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185 // Ideally we would do this now only if we were in a
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186 // threaded mode -- creating the reader later if we're
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187 // not threaded -- but we don't have access to that
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188 // information here
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189
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190 m_cacheFileReader = new WavFileReader(m_cacheFileName);
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191
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192 if (!m_cacheFileReader->isOK()) {
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193 SVDEBUG << "ERROR: CodedAudioFileReader::initialiseDecodeCache: Failed to construct WAV file reader for temporary file: " << m_cacheFileReader->getError() << endl;
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194 delete m_cacheFileReader;
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195 m_cacheFileReader = 0;
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196 m_cacheMode = CacheInMemory;
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197 sf_close(m_cacheFileWritePtr);
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198 }
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199
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200 } else {
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201 SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: failed to open cache file \"" << m_cacheFileName << "\" (" << m_channelCount << " channels, sample rate " << m_sampleRate << " for writing, falling back to in-memory cache" << endl;
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202 m_cacheMode = CacheInMemory;
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203 }
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204
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205 } catch (DirectoryCreationFailed f) {
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206 SVDEBUG << "CodedAudioFileReader::initialiseDecodeCache: failed to create temporary directory! Falling back to in-memory cache" << endl;
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207 m_cacheMode = CacheInMemory;
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208 }
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209 }
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210
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211 if (m_cacheMode == CacheInMemory) {
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212 m_data.clear();
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213 }
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214
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215 m_initialised = true;
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216 }
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217
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218 void
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219 CodedAudioFileReader::addSamplesToDecodeCache(float **samples, sv_frame_t nframes)
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220 {
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221 QMutexLocker locker(&m_cacheMutex);
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222
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223 if (!m_initialised) return;
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224
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225 for (sv_frame_t i = 0; i < nframes; ++i) {
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226
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227 for (int c = 0; c < m_channelCount; ++c) {
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228
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229 float sample = samples[c][i];
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230
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231 m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
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232
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233 if (m_cacheWriteBufferIndex ==
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234 m_cacheWriteBufferSize * m_channelCount) {
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235
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236 pushBuffer(m_cacheWriteBuffer, m_cacheWriteBufferSize, false);
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237 m_cacheWriteBufferIndex = 0;
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238 }
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239
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240 if (m_cacheWriteBufferIndex % 10240 == 0 &&
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241 m_cacheFileReader) {
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242 m_cacheFileReader->updateFrameCount();
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243 }
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244 }
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245 }
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246 }
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247
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248 void
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249 CodedAudioFileReader::addSamplesToDecodeCache(float *samples, sv_frame_t nframes)
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250 {
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251 QMutexLocker locker(&m_cacheMutex);
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252
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253 if (!m_initialised) return;
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254
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255 for (sv_frame_t i = 0; i < nframes; ++i) {
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256
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257 for (int c = 0; c < m_channelCount; ++c) {
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258
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259 float sample = samples[i * m_channelCount + c];
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260
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261 m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
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262
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263 if (m_cacheWriteBufferIndex ==
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264 m_cacheWriteBufferSize * m_channelCount) {
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265
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266 pushBuffer(m_cacheWriteBuffer, m_cacheWriteBufferSize, false);
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267 m_cacheWriteBufferIndex = 0;
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268 }
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269
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270 if (m_cacheWriteBufferIndex % 10240 == 0 &&
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271 m_cacheFileReader) {
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272 m_cacheFileReader->updateFrameCount();
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273 }
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274 }
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275 }
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276 }
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277
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278 void
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279 CodedAudioFileReader::addSamplesToDecodeCache(const vector<float> &samples)
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280 {
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281 QMutexLocker locker(&m_cacheMutex);
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282
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283 if (!m_initialised) return;
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284
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285 for (float sample: samples) {
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286
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287 m_cacheWriteBuffer[m_cacheWriteBufferIndex++] = sample;
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288
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289 if (m_cacheWriteBufferIndex ==
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290 m_cacheWriteBufferSize * m_channelCount) {
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291
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292 pushBuffer(m_cacheWriteBuffer, m_cacheWriteBufferSize, false);
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293 m_cacheWriteBufferIndex = 0;
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294 }
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295
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296 if (m_cacheWriteBufferIndex % 10240 == 0 &&
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297 m_cacheFileReader) {
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298 m_cacheFileReader->updateFrameCount();
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299 }
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300 }
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301 }
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302
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303 void
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304 CodedAudioFileReader::finishDecodeCache()
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305 {
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306 QMutexLocker locker(&m_cacheMutex);
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307
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308 Profiler profiler("CodedAudioFileReader::finishDecodeCache");
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309
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310 if (!m_initialised) {
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311 SVDEBUG << "WARNING: CodedAudioFileReader::finishDecodeCache: Cache was never initialised!" << endl;
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312 return;
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313 }
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314
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315 pushBuffer(m_cacheWriteBuffer,
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316 m_cacheWriteBufferIndex / m_channelCount,
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317 true);
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318
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319 delete[] m_cacheWriteBuffer;
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320 m_cacheWriteBuffer = 0;
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321
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322 delete[] m_resampleBuffer;
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323 m_resampleBuffer = 0;
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324
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325 delete m_resampler;
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326 m_resampler = 0;
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327
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328 if (m_cacheMode == CacheInTemporaryFile) {
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329
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330 sf_close(m_cacheFileWritePtr);
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331 m_cacheFileWritePtr = 0;
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332 if (m_cacheFileReader) m_cacheFileReader->updateFrameCount();
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333
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334 } else {
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335 // I know, I know, we already allocated it...
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336 StorageAdviser::notifyPlannedAllocation
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337 (StorageAdviser::MemoryAllocation,
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338 (m_data.size() * sizeof(float)) / 1024);
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339 }
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340
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341 SVDEBUG << "CodedAudioFileReader: File decodes to " << m_fileFrameCount
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342 << " frames" << endl;
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343 if (m_fileFrameCount != m_frameCount) {
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344 SVDEBUG << "CodedAudioFileReader: Resampled to " << m_frameCount
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345 << " frames" << endl;
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346 }
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347 SVDEBUG << "CodedAudioFileReader: Signal abs max is " << m_max
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Chris@1285
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348 << ", " << m_clippedCount
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Chris@1285
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349 << " samples clipped, first non-zero frame is at "
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Chris@1286
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350 << m_firstNonzero << ", last at " << m_lastNonzero << endl;
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Chris@1285
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351 if (m_normalised) {
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352 SVDEBUG << "CodedAudioFileReader: Normalising, gain is " << m_gain << endl;
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Chris@1285
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353 }
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Chris@297
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354 }
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Chris@297
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355
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Chris@297
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356 void
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Chris@1038
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357 CodedAudioFileReader::pushBuffer(float *buffer, sv_frame_t sz, bool final)
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Chris@297
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358 {
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Chris@757
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359 m_fileFrameCount += sz;
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360
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Chris@1040
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361 double ratio = 1.0;
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Chris@758
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362 if (m_resampler && m_fileRate != 0) {
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363 ratio = m_sampleRate / m_fileRate;
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Chris@758
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364 }
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Chris@758
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365
|
Chris@1040
|
366 if (ratio != 1.0) {
|
Chris@758
|
367 pushBufferResampling(buffer, sz, ratio, final);
|
Chris@758
|
368 } else {
|
Chris@758
|
369 pushBufferNonResampling(buffer, sz);
|
Chris@758
|
370 }
|
Chris@758
|
371 }
|
Chris@757
|
372
|
Chris@758
|
373 void
|
Chris@1038
|
374 CodedAudioFileReader::pushBufferNonResampling(float *buffer, sv_frame_t sz)
|
Chris@758
|
375 {
|
Chris@920
|
376 float clip = 1.0;
|
Chris@1038
|
377 sv_frame_t count = sz * m_channelCount;
|
Chris@318
|
378
|
Chris@1286
|
379 for (sv_frame_t j = 0; j < sz; ++j) {
|
Chris@1286
|
380 for (int c = 0; c < m_channelCount; ++c) {
|
Chris@1286
|
381 sv_frame_t i = j * m_channelCount + c;
|
Chris@1286
|
382 float v = buffer[i];
|
Chris@1286
|
383 if (!m_normalised) {
|
Chris@1286
|
384 if (v > clip) {
|
Chris@1286
|
385 buffer[i] = clip;
|
Chris@1286
|
386 ++m_clippedCount;
|
Chris@1286
|
387 } else if (v < -clip) {
|
Chris@1286
|
388 buffer[i] = -clip;
|
Chris@1286
|
389 ++m_clippedCount;
|
Chris@1286
|
390 }
|
Chris@1285
|
391 }
|
Chris@1286
|
392 v = fabsf(v);
|
Chris@1286
|
393 if (v != 0.f) {
|
Chris@1286
|
394 if (m_firstNonzero == 0) {
|
Chris@1286
|
395 m_firstNonzero = m_frameCount;
|
Chris@1286
|
396 }
|
Chris@1286
|
397 m_lastNonzero = m_frameCount;
|
Chris@1286
|
398 if (v > m_max) {
|
Chris@1286
|
399 m_max = v;
|
Chris@1286
|
400 }
|
Chris@920
|
401 }
|
Chris@920
|
402 }
|
Chris@1286
|
403 ++m_frameCount;
|
Chris@297
|
404 }
|
Chris@297
|
405
|
Chris@1286
|
406 if (m_max > 0.f) {
|
Chris@1286
|
407 m_gain = 1.f / m_max; // used when normalising only
|
Chris@1286
|
408 }
|
Chris@297
|
409
|
Chris@148
|
410 switch (m_cacheMode) {
|
Chris@148
|
411
|
Chris@148
|
412 case CacheInTemporaryFile:
|
Chris@1038
|
413 if (sf_writef_float(m_cacheFileWritePtr, buffer, sz) < sz) {
|
Chris@544
|
414 sf_close(m_cacheFileWritePtr);
|
Chris@544
|
415 m_cacheFileWritePtr = 0;
|
Chris@544
|
416 throw InsufficientDiscSpace(TempDirectory::getInstance()->getPath());
|
Chris@544
|
417 }
|
Chris@148
|
418 break;
|
Chris@148
|
419
|
Chris@148
|
420 case CacheInMemory:
|
Chris@1100
|
421 m_dataLock.lock();
|
Chris@1285
|
422 /*
|
Chris@1285
|
423 if (m_data.size() < 5120) {
|
Chris@1285
|
424 for (int i = 0; i < count && i < 5120; ++i) {
|
Chris@1285
|
425 if (i % 8 == 0) cerr << i << ": ";
|
Chris@1285
|
426 cerr << buffer[i] << " ";
|
Chris@1285
|
427 if (i % 8 == 7) cerr << endl;
|
Chris@1285
|
428 }
|
Chris@1285
|
429 }
|
Chris@1285
|
430 cerr << endl;
|
Chris@1285
|
431 */
|
Chris@1096
|
432 m_data.insert(m_data.end(), buffer, buffer + count);
|
Chris@543
|
433 m_dataLock.unlock();
|
Chris@148
|
434 break;
|
Chris@148
|
435 }
|
Chris@758
|
436 }
|
Chris@757
|
437
|
Chris@758
|
438 void
|
Chris@1038
|
439 CodedAudioFileReader::pushBufferResampling(float *buffer, sv_frame_t sz,
|
Chris@1038
|
440 double ratio, bool final)
|
Chris@758
|
441 {
|
Chris@759
|
442 SVDEBUG << "pushBufferResampling: ratio = " << ratio << ", sz = " << sz << ", final = " << final << endl;
|
Chris@757
|
443
|
Chris@759
|
444 if (sz > 0) {
|
Chris@759
|
445
|
Chris@1038
|
446 sv_frame_t out = m_resampler->resampleInterleaved
|
Chris@759
|
447 (buffer,
|
Chris@759
|
448 m_resampleBuffer,
|
Chris@759
|
449 sz,
|
Chris@759
|
450 ratio,
|
Chris@759
|
451 false);
|
Chris@759
|
452
|
Chris@759
|
453 pushBufferNonResampling(m_resampleBuffer, out);
|
Chris@759
|
454 }
|
Chris@757
|
455
|
Chris@758
|
456 if (final) {
|
Chris@758
|
457
|
Chris@1038
|
458 sv_frame_t padFrames = 1;
|
Chris@1038
|
459 if (double(m_frameCount) / ratio < double(m_fileFrameCount)) {
|
Chris@1038
|
460 padFrames = m_fileFrameCount - sv_frame_t(double(m_frameCount) / ratio) + 1;
|
Chris@757
|
461 }
|
Chris@758
|
462
|
Chris@1038
|
463 sv_frame_t padSamples = padFrames * m_channelCount;
|
Chris@758
|
464
|
Chris@1038
|
465 SVDEBUG << "frameCount = " << m_frameCount << ", equivFileFrames = " << double(m_frameCount) / ratio << ", m_fileFrameCount = " << m_fileFrameCount << ", padFrames= " << padFrames << ", padSamples = " << padSamples << endl;
|
Chris@758
|
466
|
Chris@758
|
467 float *padding = new float[padSamples];
|
Chris@1038
|
468 for (sv_frame_t i = 0; i < padSamples; ++i) padding[i] = 0.f;
|
Chris@758
|
469
|
Chris@1038
|
470 sv_frame_t out = m_resampler->resampleInterleaved
|
Chris@758
|
471 (padding,
|
Chris@758
|
472 m_resampleBuffer,
|
Chris@758
|
473 padFrames,
|
Chris@758
|
474 ratio,
|
Chris@758
|
475 true);
|
Chris@758
|
476
|
Chris@1038
|
477 if (m_frameCount + out > sv_frame_t(double(m_fileFrameCount) * ratio)) {
|
Chris@1038
|
478 out = sv_frame_t(double(m_fileFrameCount) * ratio) - m_frameCount;
|
Chris@759
|
479 }
|
Chris@759
|
480
|
Chris@758
|
481 pushBufferNonResampling(m_resampleBuffer, out);
|
Chris@758
|
482 delete[] padding;
|
Chris@757
|
483 }
|
Chris@148
|
484 }
|
Chris@148
|
485
|
Chris@1096
|
486 vector<float>
|
Chris@1041
|
487 CodedAudioFileReader::getInterleavedFrames(sv_frame_t start, sv_frame_t count) const
|
Chris@148
|
488 {
|
Chris@543
|
489 // Lock is only required in CacheInMemory mode (the cache file
|
Chris@543
|
490 // reader is expected to be thread safe and manage its own
|
Chris@543
|
491 // locking)
|
Chris@263
|
492
|
Chris@265
|
493 if (!m_initialised) {
|
Chris@690
|
494 SVDEBUG << "CodedAudioFileReader::getInterleavedFrames: not initialised" << endl;
|
Chris@1096
|
495 return {};
|
Chris@265
|
496 }
|
Chris@148
|
497
|
Chris@1096
|
498 vector<float> frames;
|
Chris@1041
|
499
|
Chris@148
|
500 switch (m_cacheMode) {
|
Chris@148
|
501
|
Chris@148
|
502 case CacheInTemporaryFile:
|
Chris@148
|
503 if (m_cacheFileReader) {
|
Chris@1041
|
504 frames = m_cacheFileReader->getInterleavedFrames(start, count);
|
Chris@148
|
505 }
|
Chris@148
|
506 break;
|
Chris@148
|
507
|
Chris@148
|
508 case CacheInMemory:
|
Chris@148
|
509 {
|
Chris@1096
|
510 if (!isOK()) return {};
|
Chris@1096
|
511 if (count == 0) return {};
|
Chris@148
|
512
|
Chris@1100
|
513 sv_frame_t ix0 = start * m_channelCount;
|
Chris@1100
|
514 sv_frame_t ix1 = ix0 + (count * m_channelCount);
|
Chris@148
|
515
|
Chris@1100
|
516 // This lock used to be a QReadWriteLock, but it appears that
|
Chris@1100
|
517 // its lock mechanism is significantly slower than QMutex so
|
Chris@1100
|
518 // it's not a good idea in cases like this where we don't
|
Chris@1100
|
519 // really have threads taking a long time to read concurrently
|
Chris@1100
|
520 m_dataLock.lock();
|
Chris@1100
|
521 sv_frame_t n = sv_frame_t(m_data.size());
|
Chris@1282
|
522 if (ix0 > n) ix0 = n;
|
Chris@1100
|
523 if (ix1 > n) ix1 = n;
|
Chris@1100
|
524 frames = vector<float>(m_data.begin() + ix0, m_data.begin() + ix1);
|
Chris@543
|
525 m_dataLock.unlock();
|
Chris@1282
|
526 break;
|
Chris@148
|
527 }
|
Chris@148
|
528 }
|
Chris@920
|
529
|
Chris@920
|
530 if (m_normalised) {
|
Chris@1052
|
531 for (auto &f: frames) f *= m_gain;
|
Chris@920
|
532 }
|
Chris@1041
|
533
|
Chris@1041
|
534 return frames;
|
Chris@148
|
535 }
|
Chris@148
|
536
|