Mercurial > hg > sonic-visualiser
view transform/RealTimePluginTransform.cpp @ 114:a46d68ae3c3e sv1-1.0pre1
* Make getPropertyRangeAndValue return the default separately from the current
value. Previously some contexts were incorrectly treating the current value
as a default.
author | Chris Cannam |
---|---|
date | Fri, 02 Mar 2007 14:00:12 +0000 |
parents | d25ea0c2af5c |
children | b4110b17bca8 |
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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 >= 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 >= 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 **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(); int i = 0; while (blockFrame < endFrame) { 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 (stvm) stvm->setCompletion(100); if (wwfm) wwfm->setCompletion(100); }