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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 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 "RealTimePluginTransformer.h"
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17
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18 #include "plugin/RealTimePluginFactory.h"
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19 #include "plugin/RealTimePluginInstance.h"
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20 #include "plugin/PluginXml.h"
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21
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22 #include "data/model/Model.h"
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23 #include "data/model/SparseTimeValueModel.h"
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24 #include "data/model/DenseTimeValueModel.h"
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25 #include "data/model/WritableWaveFileModel.h"
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26 #include "data/model/WaveFileModel.h"
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27
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28 #include <iostream>
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29
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30 RealTimePluginTransformer::RealTimePluginTransformer(Model *inputModel,
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31 QString pluginId,
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32 const ExecutionContext &context,
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33 QString configurationXml,
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34 QString units,
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35 int output) :
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36 PluginTransformer(inputModel, context),
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37 m_pluginId(pluginId),
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38 m_configurationXml(configurationXml),
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39 m_units(units),
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40 m_plugin(0),
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41 m_outputNo(output)
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42 {
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43 if (!m_context.blockSize) m_context.blockSize = 1024;
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44
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45 // std::cerr << "RealTimePluginTransformer::RealTimePluginTransformer: plugin " << pluginId.toStdString() << ", output " << output << std::endl;
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46
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47 RealTimePluginFactory *factory =
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48 RealTimePluginFactory::instanceFor(pluginId);
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49
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50 if (!factory) {
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51 std::cerr << "RealTimePluginTransformer: No factory available for plugin id \""
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52 << pluginId.toStdString() << "\"" << std::endl;
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53 return;
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54 }
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55
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56 DenseTimeValueModel *input = getInput();
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57 if (!input) return;
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58
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59 m_plugin = factory->instantiatePlugin(pluginId, 0, 0,
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60 m_input->getSampleRate(),
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61 m_context.blockSize,
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62 input->getChannelCount());
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63
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64 if (!m_plugin) {
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65 std::cerr << "RealTimePluginTransformer: Failed to instantiate plugin \""
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66 << pluginId.toStdString() << "\"" << std::endl;
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67 return;
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68 }
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69
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70 if (configurationXml != "") {
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71 PluginXml(m_plugin).setParametersFromXml(configurationXml);
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72 }
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73
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74 if (m_outputNo >= 0 &&
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75 m_outputNo >= int(m_plugin->getControlOutputCount())) {
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76 std::cerr << "RealTimePluginTransformer: Plugin has fewer than desired " << m_outputNo << " control outputs" << std::endl;
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77 return;
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78 }
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79
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80 if (m_outputNo == -1) {
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81
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82 size_t outputChannels = m_plugin->getAudioOutputCount();
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83 if (outputChannels > input->getChannelCount()) {
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84 outputChannels = input->getChannelCount();
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85 }
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86
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87 WritableWaveFileModel *model = new WritableWaveFileModel
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88 (input->getSampleRate(), outputChannels);
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89
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90 m_output = model;
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91
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92 } else {
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93
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94 SparseTimeValueModel *model = new SparseTimeValueModel
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95 (input->getSampleRate(), m_context.blockSize, 0.0, 0.0, false);
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96
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97 if (units != "") model->setScaleUnits(units);
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98
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99 m_output = model;
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100 }
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101 }
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102
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103 RealTimePluginTransformer::~RealTimePluginTransformer()
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104 {
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105 delete m_plugin;
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106 }
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107
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108 DenseTimeValueModel *
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109 RealTimePluginTransformer::getInput()
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110 {
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111 DenseTimeValueModel *dtvm =
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112 dynamic_cast<DenseTimeValueModel *>(getInputModel());
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113 if (!dtvm) {
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114 std::cerr << "RealTimePluginTransformer::getInput: WARNING: Input model is not conformable to DenseTimeValueModel" << std::endl;
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115 }
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116 return dtvm;
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117 }
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118
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119 void
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120 RealTimePluginTransformer::run()
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121 {
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122 DenseTimeValueModel *input = getInput();
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123 if (!input) return;
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124
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125 while (!input->isReady()) {
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126 if (dynamic_cast<WaveFileModel *>(input)) break; // no need to wait
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127 std::cerr << "RealTimePluginTransformer::run: Waiting for input model to be ready..." << std::endl;
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128 sleep(1);
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129 }
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130
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131 SparseTimeValueModel *stvm = dynamic_cast<SparseTimeValueModel *>(m_output);
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132 WritableWaveFileModel *wwfm = dynamic_cast<WritableWaveFileModel *>(m_output);
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133 if (!stvm && !wwfm) return;
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134
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135 if (stvm && (m_outputNo >= int(m_plugin->getControlOutputCount()))) return;
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136
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137 size_t sampleRate = input->getSampleRate();
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138 size_t channelCount = input->getChannelCount();
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139 if (!wwfm && m_context.channel != -1) channelCount = 1;
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140
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141 long blockSize = m_plugin->getBufferSize();
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142
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143 float **inbufs = m_plugin->getAudioInputBuffers();
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144
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145 long startFrame = m_input->getStartFrame();
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146 long endFrame = m_input->getEndFrame();
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147
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148 long contextStart = m_context.startFrame;
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149 long contextDuration = m_context.duration;
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150
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151 if (contextStart == 0 || contextStart < startFrame) {
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152 contextStart = startFrame;
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153 }
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154
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155 if (contextDuration == 0) {
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156 contextDuration = endFrame - contextStart;
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157 }
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158 if (contextStart + contextDuration > endFrame) {
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159 contextDuration = endFrame - contextStart;
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160 }
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161
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162 wwfm->setStartFrame(contextStart);
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163
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164 long blockFrame = contextStart;
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165
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166 long prevCompletion = 0;
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167
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168 long latency = m_plugin->getLatency();
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169
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170 while (blockFrame < contextStart + contextDuration + latency &&
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171 !m_abandoned) {
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172
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173 long completion =
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174 (((blockFrame - contextStart) / blockSize) * 99) /
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175 ((contextDuration) / blockSize);
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176
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177 long got = 0;
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178
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179 if (channelCount == 1) {
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180 if (inbufs && inbufs[0]) {
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181 got = input->getData
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182 (m_context.channel, blockFrame, blockSize, inbufs[0]);
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183 while (got < blockSize) {
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184 inbufs[0][got++] = 0.0;
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185 }
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186 }
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187 for (size_t ch = 1; ch < m_plugin->getAudioInputCount(); ++ch) {
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188 for (long i = 0; i < blockSize; ++i) {
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189 inbufs[ch][i] = inbufs[0][i];
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190 }
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191 }
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192 } else {
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193 for (size_t ch = 0; ch < channelCount; ++ch) {
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194 if (inbufs && inbufs[ch]) {
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195 got = input->getData
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196 (ch, blockFrame, blockSize, inbufs[ch]);
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197 while (got < blockSize) {
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198 inbufs[ch][got++] = 0.0;
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199 }
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200 }
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201 }
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202 for (size_t ch = channelCount; ch < m_plugin->getAudioInputCount(); ++ch) {
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203 for (long i = 0; i < blockSize; ++i) {
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204 inbufs[ch][i] = inbufs[ch % channelCount][i];
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205 }
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206 }
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207 }
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208
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209 /*
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210 std::cerr << "Input for plugin: " << m_plugin->getAudioInputCount() << " channels "<< std::endl;
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211
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212 for (size_t ch = 0; ch < m_plugin->getAudioInputCount(); ++ch) {
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213 std::cerr << "Input channel " << ch << std::endl;
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214 for (size_t i = 0; i < 100; ++i) {
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215 std::cerr << inbufs[ch][i] << " ";
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216 if (isnan(inbufs[ch][i])) {
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217 std::cerr << "\n\nWARNING: NaN in audio input" << std::endl;
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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
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223 m_plugin->run(Vamp::RealTime::frame2RealTime(blockFrame, sampleRate));
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224
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225 if (stvm) {
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226
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227 float value = m_plugin->getControlOutputValue(m_outputNo);
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228
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229 long pointFrame = blockFrame;
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230 if (pointFrame > latency) pointFrame -= latency;
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231 else pointFrame = 0;
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232
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233 stvm->addPoint(SparseTimeValueModel::Point
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234 (pointFrame, value, ""));
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235
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236 } else if (wwfm) {
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237
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238 float **outbufs = m_plugin->getAudioOutputBuffers();
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239
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240 if (outbufs) {
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241
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242 if (blockFrame >= latency) {
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243 long writeSize = std::min
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244 (blockSize,
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245 contextStart + contextDuration + latency - blockFrame);
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246 wwfm->addSamples(outbufs, writeSize);
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247 } else if (blockFrame + blockSize >= latency) {
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248 long offset = latency - blockFrame;
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249 long count = blockSize - offset;
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250 float **tmp = new float *[channelCount];
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251 for (size_t c = 0; c < channelCount; ++c) {
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252 tmp[c] = outbufs[c] + offset;
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253 }
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254 wwfm->addSamples(tmp, count);
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255 delete[] tmp;
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256 }
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257 }
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258 }
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259
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260 if (blockFrame == contextStart || completion > prevCompletion) {
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261 if (stvm) stvm->setCompletion(completion);
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262 if (wwfm) wwfm->setCompletion(completion);
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263 prevCompletion = completion;
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264 }
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265
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266 blockFrame += blockSize;
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267 }
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268
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269 if (m_abandoned) return;
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270
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271 if (stvm) stvm->setCompletion(100);
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272 if (wwfm) wwfm->setCompletion(100);
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273 }
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274
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