Mercurial > hg > svcore
view transform/RealTimeEffectModelTransformer.cpp @ 1849:00405d2cf0d1
Remove debug output
author | Chris Cannam |
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date | Wed, 29 Apr 2020 13:25:07 +0100 |
parents | 5f8fbbde08ff |
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 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 "TransformFactory.h" #include <iostream> RealTimeEffectModelTransformer::RealTimeEffectModelTransformer(Input in, const Transform &t) : ModelTransformer(in, t), m_plugin(nullptr) { Transform transform(t); if (!transform.getBlockSize()) { transform.setBlockSize(1024); m_transforms[0] = transform; } m_units = TransformFactory::getInstance()->getTransformUnits (transform.getIdentifier()); m_outputNo = (transform.getOutput() == "A") ? -1 : transform.getOutput().toInt(); QString pluginId = transform.getPluginIdentifier(); SVDEBUG << "RealTimeEffectModelTransformer::RealTimeEffectModelTransformer: plugin " << pluginId << ", output " << transform.getOutput() << endl; RealTimePluginFactory *factory = RealTimePluginFactory::instanceFor(pluginId); if (!factory) { SVCERR << "RealTimeEffectModelTransformer: No factory available for plugin id \"" << pluginId << "\"" << endl; return; } auto input = ModelById::getAs<DenseTimeValueModel>(getInputModel()); if (!input) { SVCERR << "RealTimeEffectModelTransformer: Input is absent or of wrong type" << endl; return; } m_plugin = factory->instantiatePlugin(pluginId, 0, 0, input->getSampleRate(), transform.getBlockSize(), input->getChannelCount()); if (!m_plugin) { SVCERR << "RealTimeEffectModelTransformer: Failed to instantiate plugin \"" << pluginId << "\"" << endl; return; } TransformFactory::getInstance()->setPluginParameters(transform, m_plugin); if (m_outputNo >= 0 && m_outputNo >= int(m_plugin->getControlOutputCount())) { cerr << "RealTimeEffectModelTransformer: Plugin has fewer than desired " << m_outputNo << " control outputs" << endl; return; } if (m_outputNo == -1) { int outputChannels = (int)m_plugin->getAudioOutputCount(); if (outputChannels > input->getChannelCount()) { outputChannels = input->getChannelCount(); } auto model = std::make_shared<WritableWaveFileModel> (input->getSampleRate(), outputChannels); m_outputs.push_back(ModelById::add(model)); } else { auto model = std::make_shared<SparseTimeValueModel> (input->getSampleRate(), transform.getBlockSize(), 0.0, 0.0, false); if (m_units != "") model->setScaleUnits(m_units); m_outputs.push_back(ModelById::add(model)); } } RealTimeEffectModelTransformer::~RealTimeEffectModelTransformer() { } void RealTimeEffectModelTransformer::run() { if (m_outputs.empty()) { abandon(); return; } bool ready = false; while (!ready && !m_abandoned) { { // scope so as to release input shared_ptr before sleeping auto input = ModelById::getAs<DenseTimeValueModel>(getInputModel()); if (!input) { abandon(); return; } ready = input->isReady(); } if (!ready) { SVDEBUG << "RealTimeEffectModelTransformer::run: Waiting for input model to be ready..." << endl; usleep(500000); } } if (m_abandoned) return; auto input = ModelById::getAs<DenseTimeValueModel>(getInputModel()); if (!input) { abandon(); return; } sv_samplerate_t sampleRate; int channelCount; sv_frame_t startFrame; sv_frame_t endFrame; { // scope so as not to have this borrowed pointer retained around // the edges of the process loop auto input = ModelById::getAs<DenseTimeValueModel>(getInputModel()); if (!input) { abandon(); return; } sampleRate = input->getSampleRate(); channelCount = input->getChannelCount(); startFrame = input->getStartFrame(); endFrame = input->getEndFrame(); } auto stvm = ModelById::getAs<SparseTimeValueModel>(m_outputs[0]); auto wwfm = ModelById::getAs<WritableWaveFileModel>(m_outputs[0]); if (!stvm && !wwfm) { return; } if (stvm && (m_outputNo >= int(m_plugin->getControlOutputCount()))) { return; } if (!wwfm && m_input.getChannel() != -1) channelCount = 1; sv_frame_t blockSize = m_plugin->getBufferSize(); float **inbufs = m_plugin->getAudioInputBuffers(); Transform transform = m_transforms[0]; RealTime contextStartRT = transform.getStartTime(); RealTime contextDurationRT = transform.getDuration(); sv_frame_t contextStart = RealTime::realTime2Frame(contextStartRT, sampleRate); sv_frame_t contextDuration = RealTime::realTime2Frame(contextDurationRT, sampleRate); if (contextStart == 0 || contextStart < startFrame) { contextStart = startFrame; } if (contextDuration == 0) { contextDuration = endFrame - contextStart; } if (contextStart + contextDuration > endFrame) { contextDuration = endFrame - contextStart; } if (wwfm) { wwfm->setStartFrame(contextStart); } sv_frame_t blockFrame = contextStart; int prevCompletion = 0; sv_frame_t latency = m_plugin->getLatency(); while (blockFrame < contextStart + contextDuration + latency && !m_abandoned) { int completion = int ((((blockFrame - contextStart) / blockSize) * 99) / (1 + ((contextDuration) / blockSize))); sv_frame_t got = 0; auto input = ModelById::getAs<DenseTimeValueModel>(getInputModel()); if (!input) { abandon(); return; } if (channelCount == 1) { if (inbufs && inbufs[0]) { auto data = input->getData (m_input.getChannel(), blockFrame, blockSize); got = data.size(); for (sv_frame_t i = 0; i < got; ++i) { inbufs[0][i] = data[i]; } while (got < blockSize) { inbufs[0][got++] = 0.f; } for (int ch = 1; ch < (int)m_plugin->getAudioInputCount(); ++ch) { for (sv_frame_t i = 0; i < blockSize; ++i) { inbufs[ch][i] = inbufs[0][i]; } } } } else { if (inbufs && inbufs[0]) { auto data = input->getMultiChannelData (0, channelCount - 1, blockFrame, blockSize); if (!data.empty()) got = data[0].size(); for (int ch = 0; ch < channelCount; ++ch) { for (sv_frame_t i = 0; i < got; ++i) { inbufs[ch][i] = data[ch][i]; } } while (got < blockSize) { for (int ch = 0; ch < channelCount; ++ch) { inbufs[ch][got] = 0.0; } ++got; } for (int ch = channelCount; ch < (int)m_plugin->getAudioInputCount(); ++ch) { for (sv_frame_t i = 0; i < blockSize; ++i) { inbufs[ch][i] = inbufs[ch % channelCount][i]; } } } } m_plugin->run(RealTime::frame2RealTime(blockFrame, sampleRate)); if (stvm) { float value = m_plugin->getControlOutputValue(m_outputNo); sv_frame_t pointFrame = blockFrame; if (pointFrame > latency) pointFrame -= latency; else pointFrame = 0; stvm->add(Event(pointFrame, value, "")); } else if (wwfm) { float **outbufs = m_plugin->getAudioOutputBuffers(); if (outbufs) { if (blockFrame >= latency) { sv_frame_t writeSize = std::min (blockSize, contextStart + contextDuration + latency - blockFrame); wwfm->addSamples(outbufs, writeSize); } else if (blockFrame + blockSize >= latency) { sv_frame_t offset = latency - blockFrame; sv_frame_t count = blockSize - offset; float **tmp = new float *[channelCount]; for (int c = 0; c < channelCount; ++c) { tmp[c] = outbufs[c] + offset; } wwfm->addSamples(tmp, count); delete[] tmp; } } } if (blockFrame == contextStart || completion > prevCompletion) { // This setCompletion is probably misusing the completion // terminology, just as it was for WritableWaveFileModel if (stvm) stvm->setCompletion(completion); if (wwfm) wwfm->setWriteProportion(completion); prevCompletion = completion; } blockFrame += blockSize; } if (m_abandoned) return; if (stvm) stvm->setCompletion(100); if (wwfm) wwfm->writeComplete(); }