Mercurial > hg > svcore
view transform/RealTimePluginTransform.cpp @ 115:90ade4fa63be
* Fix serious failure to reload "imported" (i.e. all non-derived non-main)
models from .sv file
* Give a short playback duration to notes with formal duration of 0 or 1
* Show crosshairs on spectrogram even when there is another layer on top
(if it isn't opaque)
* Always paste to the same time in the layer as the cut/copy was from, rather
than to the playback pointer -- less flexible, but more predictable and
less annoying. We probably need a way to get the old behaviour if pasting
from somewhere else in the future (e.g. from a text file), but we can't do
that yet anyway
* Use a compound operation for dragging and resizing selections, so as to
ensure a single undo operation works
* Use a note model as the target for feature extraction plugins that output
variable samplerate data with more than one value per feature
* Avoid possible crashes in cut/paste if a layer proves to have no model
author | Chris Cannam |
---|---|
date | Thu, 11 May 2006 11:35:46 +0000 |
parents | 47fd14e29813 |
children |
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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 "base/Model.h" #include "model/SparseTimeValueModel.h" #include "model/DenseTimeValueModel.h" #include <iostream> RealTimePluginTransform::RealTimePluginTransform(Model *inputModel, QString pluginId, int channel, QString configurationXml, QString units, int output) : Transform(inputModel), m_plugin(0), m_channel(channel), m_outputNo(output) { 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(), 1024, //!!! wants to be configurable 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 >= m_plugin->getControlOutputCount()) { std::cerr << "RealTimePluginTransform: Plugin has fewer than desired " << m_outputNo << " control outputs" << std::endl; return; } SparseTimeValueModel *model = new SparseTimeValueModel (input->getSampleRate(), 1024, //!!! 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 *model = dynamic_cast<SparseTimeValueModel *>(m_output); if (!model) return; if (m_outputNo >= m_plugin->getControlOutputCount()) return; size_t sampleRate = input->getSampleRate(); int channelCount = input->getChannelCount(); if (m_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; int i = 0; while (blockFrame < endFrame) { size_t completion = (((blockFrame - startFrame) / blockSize) * 99) / ( (endFrame - startFrame) / blockSize); size_t got = 0; if (channelCount == 1) { got = input->getValues (m_channel, blockFrame, blockFrame + blockSize, buffers[0]); while (got < blockSize) { buffers[0][got++] = 0.0; } if (m_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) { 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)); float value = m_plugin->getControlOutputValue(m_outputNo); model->addPoint(SparseTimeValueModel::Point (blockFrame - m_plugin->getLatency(), value, "")); if (blockFrame == startFrame || completion > prevCompletion) { model->setCompletion(completion); prevCompletion = completion; } blockFrame += blockSize; } model->setCompletion(100); }