Mercurial > hg > svcore
view plugin/transform/RealTimeEffectModelTransformer.cpp @ 340:516819f2b97b
* Add Erase tool and mode
* Add icons for Normalize buttons in property boxes, and for Show Peaks
* Add support for velocity in notes -- not yet reflected in display or
editable in the note edit dialog, but they are imported from MIDI,
played, and exported
* Begin work on making pastes align pasted times (subtler than I thought)
author | Chris Cannam |
---|---|
date | Fri, 23 Nov 2007 16:48:23 +0000 |
parents | f620ce48c950 |
children | d7c41483af8f 94fc0591ea43 |
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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 and QMUL. 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 "RealTimeEffectModelTransformer.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 "data/model/WaveFileModel.h" #include <iostream> RealTimeEffectModelTransformer::RealTimeEffectModelTransformer(Model *inputModel, QString pluginId, const ExecutionContext &context, QString configurationXml, QString units, int output) : PluginTransformer(inputModel, context), m_pluginId(pluginId), m_configurationXml(configurationXml), m_units(units), m_plugin(0), m_outputNo(output) { if (!m_context.blockSize) m_context.blockSize = 1024; // std::cerr << "RealTimeEffectModelTransformer::RealTimeEffectModelTransformer: plugin " << pluginId.toStdString() << ", output " << output << std::endl; RealTimePluginFactory *factory = RealTimePluginFactory::instanceFor(pluginId); if (!factory) { std::cerr << "RealTimeEffectModelTransformer: 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 << "RealTimeEffectModelTransformer: Failed to instantiate plugin \"" << pluginId.toStdString() << "\"" << std::endl; return; } if (configurationXml != "") { PluginXml(m_plugin).setParametersFromXml(configurationXml); } if (m_outputNo >= 0 && m_outputNo >= int(m_plugin->getControlOutputCount())) { std::cerr << "RealTimeEffectModelTransformer: Plugin has fewer than desired " << m_outputNo << " control outputs" << std::endl; return; } if (m_outputNo == -1) { size_t outputChannels = m_plugin->getAudioOutputCount(); if (outputChannels > input->getChannelCount()) { outputChannels = input->getChannelCount(); } WritableWaveFileModel *model = new WritableWaveFileModel (input->getSampleRate(), outputChannels); 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; } } RealTimeEffectModelTransformer::~RealTimeEffectModelTransformer() { delete m_plugin; } DenseTimeValueModel * RealTimeEffectModelTransformer::getInput() { DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(getInputModel()); if (!dtvm) { std::cerr << "RealTimeEffectModelTransformer::getInput: WARNING: Input model is not conformable to DenseTimeValueModel" << std::endl; } return dtvm; } void RealTimeEffectModelTransformer::run() { DenseTimeValueModel *input = getInput(); if (!input) return; while (!input->isReady()) { if (dynamic_cast<WaveFileModel *>(input)) break; // no need to wait std::cerr << "RealTimeEffectModelTransformer::run: Waiting for input model to be ready..." << std::endl; sleep(1); } SparseTimeValueModel *stvm = dynamic_cast<SparseTimeValueModel *>(m_output); WritableWaveFileModel *wwfm = dynamic_cast<WritableWaveFileModel *>(m_output); if (!stvm && !wwfm) return; if (stvm && (m_outputNo >= int(m_plugin->getControlOutputCount()))) return; size_t sampleRate = input->getSampleRate(); size_t channelCount = input->getChannelCount(); if (!wwfm && m_context.channel != -1) channelCount = 1; long blockSize = m_plugin->getBufferSize(); float **inbufs = m_plugin->getAudioInputBuffers(); long startFrame = m_input->getStartFrame(); long endFrame = m_input->getEndFrame(); long contextStart = m_context.startFrame; long contextDuration = m_context.duration; if (contextStart == 0 || contextStart < startFrame) { contextStart = startFrame; } if (contextDuration == 0) { contextDuration = endFrame - contextStart; } if (contextStart + contextDuration > endFrame) { contextDuration = endFrame - contextStart; } wwfm->setStartFrame(contextStart); long blockFrame = contextStart; long prevCompletion = 0; long latency = m_plugin->getLatency(); while (blockFrame < contextStart + contextDuration + latency && !m_abandoned) { long completion = (((blockFrame - contextStart) / blockSize) * 99) / ((contextDuration) / blockSize); long got = 0; if (channelCount == 1) { if (inbufs && inbufs[0]) { got = input->getData (m_context.channel, blockFrame, blockSize, inbufs[0]); while (got < blockSize) { inbufs[0][got++] = 0.0; } } for (size_t ch = 1; ch < m_plugin->getAudioInputCount(); ++ch) { for (long i = 0; i < blockSize; ++i) { inbufs[ch][i] = inbufs[0][i]; } } } else { for (size_t ch = 0; ch < channelCount; ++ch) { if (inbufs && inbufs[ch]) { got = input->getData (ch, blockFrame, blockSize, inbufs[ch]); while (got < blockSize) { inbufs[ch][got++] = 0.0; } } } for (size_t ch = channelCount; ch < m_plugin->getAudioInputCount(); ++ch) { for (long i = 0; i < blockSize; ++i) { inbufs[ch][i] = inbufs[ch % channelCount][i]; } } } /* std::cerr << "Input for plugin: " << m_plugin->getAudioInputCount() << " channels "<< std::endl; for (size_t ch = 0; ch < m_plugin->getAudioInputCount(); ++ch) { std::cerr << "Input channel " << ch << std::endl; for (size_t i = 0; i < 100; ++i) { std::cerr << inbufs[ch][i] << " "; if (isnan(inbufs[ch][i])) { std::cerr << "\n\nWARNING: NaN in audio input" << std::endl; } } } */ m_plugin->run(Vamp::RealTime::frame2RealTime(blockFrame, sampleRate)); if (stvm) { float value = m_plugin->getControlOutputValue(m_outputNo); long pointFrame = blockFrame; if (pointFrame > latency) pointFrame -= latency; else pointFrame = 0; stvm->addPoint(SparseTimeValueModel::Point (pointFrame, value, "")); } else if (wwfm) { float **outbufs = m_plugin->getAudioOutputBuffers(); if (outbufs) { if (blockFrame >= latency) { long writeSize = std::min (blockSize, contextStart + contextDuration + latency - blockFrame); wwfm->addSamples(outbufs, writeSize); } else if (blockFrame + blockSize >= latency) { long offset = latency - blockFrame; long count = blockSize - offset; float **tmp = new float *[channelCount]; for (size_t c = 0; c < channelCount; ++c) { tmp[c] = outbufs[c] + offset; } wwfm->addSamples(tmp, count); delete[] tmp; } } } if (blockFrame == contextStart || completion > prevCompletion) { if (stvm) stvm->setCompletion(completion); if (wwfm) wwfm->setCompletion(completion); prevCompletion = completion; } blockFrame += blockSize; } if (m_abandoned) return; if (stvm) stvm->setCompletion(100); if (wwfm) wwfm->setCompletion(100); }