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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 "RealTimePluginTransform.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 RealTimePluginTransform::RealTimePluginTransform(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 PluginTransform(inputModel, context),
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37 m_plugin(0),
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38 m_outputNo(output)
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39 {
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40 if (!m_context.blockSize) m_context.blockSize = 1024;
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41
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42 std::cerr << "RealTimePluginTransform::RealTimePluginTransform: plugin " << pluginId.toStdString() << ", output " << output << std::endl;
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43
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44 RealTimePluginFactory *factory =
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45 RealTimePluginFactory::instanceFor(pluginId);
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46
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47 if (!factory) {
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48 std::cerr << "RealTimePluginTransform: No factory available for plugin id \""
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49 << pluginId.toStdString() << "\"" << std::endl;
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50 return;
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51 }
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52
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53 DenseTimeValueModel *input = getInput();
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54 if (!input) return;
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55
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56 m_plugin = factory->instantiatePlugin(pluginId, 0, 0,
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57 m_input->getSampleRate(),
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58 m_context.blockSize,
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59 input->getChannelCount());
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60
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61 if (!m_plugin) {
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62 std::cerr << "RealTimePluginTransform: Failed to instantiate plugin \""
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63 << pluginId.toStdString() << "\"" << std::endl;
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64 return;
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65 }
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66
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67 if (configurationXml != "") {
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68 PluginXml(m_plugin).setParametersFromXml(configurationXml);
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69 }
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70
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71 if (m_outputNo >= 0 && m_outputNo >= m_plugin->getControlOutputCount()) {
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72 std::cerr << "RealTimePluginTransform: Plugin has fewer than desired " << m_outputNo << " control outputs" << std::endl;
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73 return;
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74 }
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75
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76 if (m_outputNo == -1) {
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77
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78 size_t outputChannels = m_plugin->getAudioOutputCount();
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79 if (outputChannels > input->getChannelCount()) {
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80 outputChannels = input->getChannelCount();
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81 }
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82
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83 WritableWaveFileModel *model = new WritableWaveFileModel
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84 (input->getSampleRate(), outputChannels);
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85
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86 m_output = model;
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87
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88 } else {
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89
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90 SparseTimeValueModel *model = new SparseTimeValueModel
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91 (input->getSampleRate(), m_context.blockSize, 0.0, 0.0, false);
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92
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93 if (units != "") model->setScaleUnits(units);
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94
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95 m_output = model;
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96 }
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97 }
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98
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99 RealTimePluginTransform::~RealTimePluginTransform()
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100 {
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101 delete m_plugin;
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102 }
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103
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104 DenseTimeValueModel *
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105 RealTimePluginTransform::getInput()
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106 {
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107 DenseTimeValueModel *dtvm =
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108 dynamic_cast<DenseTimeValueModel *>(getInputModel());
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109 if (!dtvm) {
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110 std::cerr << "RealTimePluginTransform::getInput: WARNING: Input model is not conformable to DenseTimeValueModel" << std::endl;
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111 }
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112 return dtvm;
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113 }
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114
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115 void
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116 RealTimePluginTransform::run()
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117 {
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118 DenseTimeValueModel *input = getInput();
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119 if (!input) return;
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120
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121 while (!input->isReady()) {
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122 if (dynamic_cast<WaveFileModel *>(input)) break; // no need to wait
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123 std::cerr << "FeatureExtractionPluginTransform::run: Waiting for input model to be ready..." << std::endl;
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124 sleep(1);
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125 }
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126
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127 SparseTimeValueModel *stvm = dynamic_cast<SparseTimeValueModel *>(m_output);
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128 WritableWaveFileModel *wwfm = dynamic_cast<WritableWaveFileModel *>(m_output);
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129 if (!stvm && !wwfm) return;
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130
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131 if (stvm && (m_outputNo >= m_plugin->getControlOutputCount())) return;
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132
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133 size_t sampleRate = input->getSampleRate();
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134 int channelCount = input->getChannelCount();
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135 if (!wwfm && m_context.channel != -1) channelCount = 1;
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136
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137 size_t blockSize = m_plugin->getBufferSize();
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138
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139 float **buffers = m_plugin->getAudioInputBuffers();
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140
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141 size_t startFrame = m_input->getStartFrame();
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142 size_t endFrame = m_input->getEndFrame();
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143 size_t blockFrame = startFrame;
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144
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145 size_t prevCompletion = 0;
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146
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147 size_t latency = m_plugin->getLatency();
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148
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149 int i = 0;
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150
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151 while (blockFrame < endFrame) {
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152
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153 size_t completion =
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154 (((blockFrame - startFrame) / blockSize) * 99) /
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155 ( (endFrame - startFrame) / blockSize);
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156
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157 size_t got = 0;
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158
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159 if (channelCount == 1) {
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160 if (buffers && buffers[0]) {
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161 got = input->getValues
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162 (m_context.channel, blockFrame, blockFrame + blockSize, buffers[0]);
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163 while (got < blockSize) {
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164 buffers[0][got++] = 0.0;
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165 }
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166 if (m_context.channel == -1 && channelCount > 1) {
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167 // use mean instead of sum, as plugin input
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168 for (size_t i = 0; i < got; ++i) {
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169 buffers[0][i] /= channelCount;
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170 }
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171 }
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172 }
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173 } else {
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174 for (size_t ch = 0; ch < channelCount; ++ch) {
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175 if (buffers && buffers[ch]) {
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176 got = input->getValues
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177 (ch, blockFrame, blockFrame + blockSize, buffers[ch]);
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178 while (got < blockSize) {
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179 buffers[ch][got++] = 0.0;
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180 }
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181 }
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182 }
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183 }
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184
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185 m_plugin->run(Vamp::RealTime::frame2RealTime(blockFrame, sampleRate));
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186
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187 if (stvm) {
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188
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189 float value = m_plugin->getControlOutputValue(m_outputNo);
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190
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191 size_t pointFrame = blockFrame;
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192 if (pointFrame > latency) pointFrame -= latency;
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193 else pointFrame = 0;
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194
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195 stvm->addPoint(SparseTimeValueModel::Point
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196 (pointFrame, value, ""));
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197
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198 } else if (wwfm) {
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199
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200 float **buffers = m_plugin->getAudioOutputBuffers();
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201
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202 if (buffers) {
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203
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204 if (blockFrame >= latency) {
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205 wwfm->addSamples(buffers, blockSize);
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206 } else if (blockFrame + blockSize >= latency) {
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207 size_t offset = latency - blockFrame;
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208 size_t count = blockSize - offset;
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209 float **tmp = new float *[channelCount];
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210 for (size_t c = 0; c < channelCount; ++c) {
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211 tmp[c] = buffers[c] + offset;
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212 }
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213 wwfm->addSamples(tmp, count);
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214 delete[] tmp;
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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 if (blockFrame == startFrame || completion > prevCompletion) {
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220 if (stvm) stvm->setCompletion(completion);
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221 if (wwfm) wwfm->setCompletion(completion);
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222 prevCompletion = completion;
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223 }
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224
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225 blockFrame += blockSize;
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226 }
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227
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228 if (stvm) stvm->setCompletion(100);
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229 if (wwfm) wwfm->setCompletion(100);
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230 }
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231
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