Mercurial > hg > sonic-visualiser
view transform/RealTimePluginTransform.cpp @ 46:47bff9f88882
* Add --no-audio option to avoid opening any audio output device
author | Chris Cannam |
---|---|
date | Thu, 05 Oct 2006 11:01:36 +0000 |
parents | 5a72bf7490ae |
children | 527598e2fa10 |
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/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ /* Sonic Visualiser An audio file viewer and annotation editor. Centre for Digital Music, Queen Mary, University of London. This file copyright 2006 Chris Cannam. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. See the file COPYING included with this distribution for more information. */ #include "RealTimePluginTransform.h" #include "plugin/RealTimePluginFactory.h" #include "plugin/RealTimePluginInstance.h" #include "plugin/PluginXml.h" #include "data/model/Model.h" #include "data/model/SparseTimeValueModel.h" #include "data/model/DenseTimeValueModel.h" #include "data/model/WritableWaveFileModel.h" #include <iostream> RealTimePluginTransform::RealTimePluginTransform(Model *inputModel, QString pluginId, const ExecutionContext &context, QString configurationXml, QString units, int output) : PluginTransform(inputModel, context), m_plugin(0), m_outputNo(output) { if (!m_context.blockSize) m_context.blockSize = 1024; std::cerr << "RealTimePluginTransform::RealTimePluginTransform: plugin " << pluginId.toStdString() << ", output " << output << std::endl; RealTimePluginFactory *factory = RealTimePluginFactory::instanceFor(pluginId); if (!factory) { std::cerr << "RealTimePluginTransform: No factory available for plugin id \"" << pluginId.toStdString() << "\"" << std::endl; return; } DenseTimeValueModel *input = getInput(); if (!input) return; m_plugin = factory->instantiatePlugin(pluginId, 0, 0, m_input->getSampleRate(), m_context.blockSize, input->getChannelCount()); if (!m_plugin) { std::cerr << "RealTimePluginTransform: Failed to instantiate plugin \"" << pluginId.toStdString() << "\"" << std::endl; return; } if (configurationXml != "") { PluginXml(m_plugin).setParametersFromXml(configurationXml); } if (m_outputNo >= 0 && m_outputNo >= m_plugin->getControlOutputCount()) { std::cerr << "RealTimePluginTransform: Plugin has fewer than desired " << m_outputNo << " control outputs" << std::endl; return; } if (m_outputNo == -1) { WritableWaveFileModel *model = new WritableWaveFileModel (input->getSampleRate(), input->getChannelCount()); //!!! m_output = model; } else { SparseTimeValueModel *model = new SparseTimeValueModel (input->getSampleRate(), m_context.blockSize, 0.0, 0.0, false); if (units != "") model->setScaleUnits(units); m_output = model; } } RealTimePluginTransform::~RealTimePluginTransform() { delete m_plugin; } DenseTimeValueModel * RealTimePluginTransform::getInput() { DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(getInputModel()); if (!dtvm) { std::cerr << "RealTimePluginTransform::getInput: WARNING: Input model is not conformable to DenseTimeValueModel" << std::endl; } return dtvm; } void RealTimePluginTransform::run() { DenseTimeValueModel *input = getInput(); if (!input) return; SparseTimeValueModel *stvm = dynamic_cast<SparseTimeValueModel *>(m_output); WritableWaveFileModel *wwfm = dynamic_cast<WritableWaveFileModel *>(m_output); if (!stvm && !wwfm) return; if (stvm && (m_outputNo >= m_plugin->getControlOutputCount())) return; size_t sampleRate = input->getSampleRate(); int channelCount = input->getChannelCount(); if (!wwfm && m_context.channel != -1) channelCount = 1; size_t blockSize = m_plugin->getBufferSize(); float **buffers = m_plugin->getAudioInputBuffers(); size_t startFrame = m_input->getStartFrame(); size_t endFrame = m_input->getEndFrame(); size_t blockFrame = startFrame; size_t prevCompletion = 0; size_t latency = m_plugin->getLatency(); int i = 0; while (blockFrame < endFrame) { size_t completion = (((blockFrame - startFrame) / blockSize) * 99) / ( (endFrame - startFrame) / blockSize); size_t got = 0; if (channelCount == 1) { if (buffers && buffers[0]) { got = input->getValues (m_context.channel, blockFrame, blockFrame + blockSize, buffers[0]); while (got < blockSize) { buffers[0][got++] = 0.0; } if (m_context.channel == -1 && channelCount > 1) { // use mean instead of sum, as plugin input for (size_t i = 0; i < got; ++i) { buffers[0][i] /= channelCount; } } } } else { for (size_t ch = 0; ch < channelCount; ++ch) { if (buffers && buffers[ch]) { got = input->getValues (ch, blockFrame, blockFrame + blockSize, buffers[ch]); while (got < blockSize) { buffers[ch][got++] = 0.0; } } } } m_plugin->run(Vamp::RealTime::frame2RealTime(blockFrame, sampleRate)); if (stvm) { float value = m_plugin->getControlOutputValue(m_outputNo); size_t pointFrame = blockFrame; if (pointFrame > latency) pointFrame -= latency; else pointFrame = 0; stvm->addPoint(SparseTimeValueModel::Point (pointFrame, value, "")); } else if (wwfm) { float **buffers = m_plugin->getAudioOutputBuffers(); if (buffers) { //!!! This will fail if any buffers[c] is null or //uninitialised. The plugin instance should ensure //that that can't happen -- but it doesn't if (blockFrame >= latency) { wwfm->addSamples(buffers, blockSize); } else if (blockFrame + blockSize >= latency) { size_t offset = latency - blockFrame; size_t count = blockSize - offset; float **tmp = new float *[channelCount]; for (size_t c = 0; c < channelCount; ++c) { tmp[c] = buffers[c] + offset; } wwfm->addSamples(tmp, count); delete[] tmp; } } } if (blockFrame == startFrame || completion > prevCompletion) { if (stvm) stvm->setCompletion(completion); prevCompletion = completion; } blockFrame += blockSize; } if (stvm) stvm->setCompletion(100); if (wwfm) wwfm->sync(); }