comparison plugin/transform/RealTimePluginTransform.cpp @ 320:32e50b620a6c

* Move some things around to facilitate plundering libraries for other
  applications without needing to duplicate so much code.
    sv/osc -> data/osc
    sv/audioio -> audioio
    sv/transform -> plugin/transform
    sv/document -> document (will rename to framework in next commit)

+/* -*- 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;
+
+    long blockSize = m_plugin->getBufferSize();
+
+    float **inbufs = m_plugin->getAudioInputBuffers();
+
+    long startFrame = m_input->getStartFrame();
+    long   endFrame = m_input->getEndFrame();
+    
+    long contextStart = m_context.startFrame;
+    long contextDuration = m_context.duration;
+
+    if (contextStart == 0 || contextStart < startFrame) {
+        contextStart = startFrame;
+    }
+
+    if (contextDuration == 0) {
+        contextDuration = endFrame - contextStart;
+    }
+    if (contextStart + contextDuration > endFrame) {
+        contextDuration = endFrame - contextStart;
+    }
+
+    wwfm->setStartFrame(contextStart);
+
+    long blockFrame = contextStart;
+
+    long prevCompletion = 0;
+
+    long latency = m_plugin->getLatency();
+
+    while (blockFrame < contextStart + contextDuration + latency &&
+           !m_abandoned) {
+
+	long completion =
+	    (((blockFrame - contextStart) / blockSize) * 99) /
+	    ((contextDuration) / blockSize);
+
+	long got = 0;
+
+	if (channelCount == 1) {
+            if (inbufs && inbufs[0]) {
+                got = input->getData
+                    (m_context.channel, blockFrame, blockSize, inbufs[0]);
+                while (got < blockSize) {
+                    inbufs[0][got++] = 0.0;
+                }          
+            }
+            for (size_t ch = 1; ch < m_plugin->getAudioInputCount(); ++ch) {
+                for (long 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->getData
+                        (ch, blockFrame, blockSize, inbufs[ch]);
+                    while (got < blockSize) {
+                        inbufs[ch][got++] = 0.0;
+                    }
+                }
+	    }
+            for (size_t ch = channelCount; ch < m_plugin->getAudioInputCount(); ++ch) {
+                for (long 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);
+
+            long 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) {
+                    long writeSize = std::min
+                        (blockSize,
+                         contextStart + contextDuration + latency - blockFrame);
+                    wwfm->addSamples(outbufs, writeSize);
+                } else if (blockFrame + blockSize >= latency) {
+                    long offset = latency - blockFrame;
+                    long 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 == contextStart || 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);
+}
+