Mercurial > hg > sonic-visualiser
view transform/RealTimePluginTransform.cpp @ 172:c1980ed39d2e
* continue to pick "new" colours for coloured layers even when all colours
have been used at least once, rather than sticking on the last one
* some messing about with application palette settings
* when replacing an audio file, retain the previous playback settings for
any layers that depended on the old file
* re-check plugin program setting when a parameter changes -- so a plugin
can decide to reset the program if the parameters no longer match those
for the current program
* fix failure to update check-boxes for toggled plugin parameters when their
parameters are changed by program changes
author | Chris Cannam |
---|---|
date | Thu, 09 Aug 2007 14:40:03 +0000 |
parents | 107ca17594c8 |
children | 3e5a32a2acf4 |
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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 "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 "data/model/WaveFileModel.h" #include <iostream> RealTimePluginTransform::RealTimePluginTransform(Model *inputModel, QString pluginId, const ExecutionContext &context, QString configurationXml, QString units, int output) : PluginTransform(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 << "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 >= int(m_plugin->getControlOutputCount())) { std::cerr << "RealTimePluginTransform: 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; } } 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; while (!input->isReady()) { if (dynamic_cast<WaveFileModel *>(input)) break; // no need to wait std::cerr << "RealTimePluginTransform::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; size_t blockSize = m_plugin->getBufferSize(); float **inbufs = 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(); while (blockFrame < endFrame && !m_abandoned) { size_t completion = (((blockFrame - startFrame) / blockSize) * 99) / ( (endFrame - startFrame) / blockSize); size_t got = 0; if (channelCount == 1) { if (inbufs && inbufs[0]) { got = input->getValues (m_context.channel, blockFrame, blockFrame + blockSize, inbufs[0]); while (got < blockSize) { inbufs[0][got++] = 0.0; } } for (size_t ch = 1; ch < m_plugin->getAudioInputCount(); ++ch) { for (size_t 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->getValues (ch, blockFrame, blockFrame + blockSize, inbufs[ch]); while (got < blockSize) { inbufs[ch][got++] = 0.0; } } } for (size_t ch = channelCount; ch < m_plugin->getAudioInputCount(); ++ch) { for (size_t 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); size_t 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) { wwfm->addSamples(outbufs, 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] = outbufs[c] + offset; } wwfm->addSamples(tmp, count); delete[] tmp; } } } if (blockFrame == startFrame || 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); }