Mercurial > hg > sonic-visualiser
view transform/RealTimePluginTransform.cpp @ 88:51be0daa1386
Several changes related to referring to remote URLs for sessions and files:
* Pull file dialog wrapper functions out from MainWindow into FileFinder
* If a file referred to in a session is not found at its expected location,
try a few other alternatives (same location as the session file or same
location as the last audio file) before asking the user to locate it
* Allow user to give a URL when locating an audio file, not just locate on
the filesystem
* Make wave file models remember the "original" location (e.g. URL) of
the audio file, not just the actual location from which the data was
loaded (e.g. local copy of that URL) -- when saving a session, use the
original location so as not to refer to a temporary file
* Clean up incompletely-downloaded local copies of files
author | Chris Cannam |
---|---|
date | Thu, 11 Jan 2007 13:29:58 +0000 |
parents | bedc7517b6e8 |
children | d25ea0c2af5c |
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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_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 << "FeatureExtractionPluginTransform::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 **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) { 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); if (wwfm) wwfm->setCompletion(completion); prevCompletion = completion; } blockFrame += blockSize; } if (stvm) stvm->setCompletion(100); if (wwfm) wwfm->setCompletion(100); }