Chris@0: /* -*- c-basic-offset: 4 -*-  vi:set ts=8 sts=4 sw=4: */
Chris@0: 
Chris@0: /*
Chris@0:     A waveform viewer and audio annotation editor.
Chris@2:     Chris Cannam, Queen Mary University of London, 2005-2006
Chris@0:     
Chris@0:     This is experimental software.  Not for distribution.
Chris@0: */
Chris@0: 
Chris@0: #include "AudioGenerator.h"
Chris@0: 
Chris@0: #include "base/ViewManager.h"
Chris@0: #include "base/PlayParameters.h"
Chris@11: #include "base/PlayParameterRepository.h"
Chris@0: 
Chris@11: #include "model/NoteModel.h"
Chris@0: #include "model/DenseTimeValueModel.h"
Chris@0: #include "model/SparseOneDimensionalModel.h"
Chris@0: 
Chris@0: #include "plugin/RealTimePluginFactory.h"
Chris@0: #include "plugin/RealTimePluginInstance.h"
Chris@0: #include "plugin/PluginIdentifier.h"
Chris@0: #include "plugin/api/alsa/seq_event.h"
Chris@0: 
Chris@0: #include <iostream>
Martin@14: #include <math.h>
Chris@0: 
Chris@0: const size_t
Chris@0: AudioGenerator::m_pluginBlockSize = 2048;
Chris@0: 
Chris@5: //#define DEBUG_AUDIO_GENERATOR 1
Chris@0: 
Chris@0: AudioGenerator::AudioGenerator(ViewManager *manager) :
Chris@0:     m_viewManager(manager),
Chris@0:     m_sourceSampleRate(0),
Chris@0:     m_targetChannelCount(1)
Chris@0: {
Chris@0: }
Chris@0: 
Chris@0: AudioGenerator::~AudioGenerator()
Chris@0: {
Chris@0: }
Chris@0: 
Chris@10: bool
Chris@13: AudioGenerator::canPlay(const Model *model)
Chris@13: {
Chris@13:     if (dynamic_cast<const DenseTimeValueModel *>(model) ||
Chris@13: 	dynamic_cast<const SparseOneDimensionalModel *>(model) ||
Chris@13: 	dynamic_cast<const NoteModel *>(model)) {
Chris@13: 	return true;
Chris@13:     } else {
Chris@13: 	return false;
Chris@13:     }
Chris@13: }
Chris@13: 
Chris@13: bool
Chris@0: AudioGenerator::addModel(Model *model)
Chris@0: {
Chris@0:     if (m_sourceSampleRate == 0) {
Chris@0: 
Chris@0: 	m_sourceSampleRate = model->getSampleRate();
Chris@0: 
Chris@0:     } else {
Chris@0: 
Chris@0: 	DenseTimeValueModel *dtvm =
Chris@0: 	    dynamic_cast<DenseTimeValueModel *>(model);
Chris@0: 
Chris@0: 	if (dtvm) {
Chris@0: 	    m_sourceSampleRate = model->getSampleRate();
Chris@10: 	    return true;
Chris@0: 	}
Chris@0:     }
Chris@0: 
Chris@0:     SparseOneDimensionalModel *sodm =
Chris@0: 	dynamic_cast<SparseOneDimensionalModel *>(model);
Chris@11:     if (sodm) {
Chris@11: 	QString pluginId = QString("dssi:%1:sample_player").
Chris@11: 	    arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
Chris@11: 	RealTimePluginInstance *plugin = loadPlugin(pluginId, "cowbell");
Chris@11: 	if (plugin) {
Chris@11: 	    QMutexLocker locker(&m_mutex);
Chris@11: 	    m_synthMap[sodm] = plugin;
Chris@11: 	    return true;
Chris@11: 	} else {
Chris@11: 	    return false;
Chris@11: 	}
Chris@11:     }
Chris@0: 
Chris@11:     NoteModel *nm = dynamic_cast<NoteModel *>(model);
Chris@11:     if (nm) {
Chris@11: 	QString pluginId = QString("dssi:%1:sample_player").
Chris@11: 	    arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
Chris@11: 	RealTimePluginInstance *plugin = loadPlugin(pluginId, "piano");
Chris@11: 	if (plugin) {
Chris@11: 	    QMutexLocker locker(&m_mutex);
Chris@11: 	    m_synthMap[nm] = plugin;
Chris@11: 	    return true;
Chris@11: 	} else {
Chris@11: 	    return false;
Chris@11: 	}
Chris@11:     }
Chris@11: 	
Chris@11: 
Chris@11:     return false;
Chris@11: }
Chris@11: 
Chris@11: RealTimePluginInstance *
Chris@11: AudioGenerator::loadPlugin(QString pluginId, QString program)
Chris@11: {
Chris@0: //	QString pluginId = "dssi:/usr/lib/dssi/dssi-vst.so:FEARkILLERrev1.dll";
Chris@0: //	QString pluginId = "dssi:/usr/lib/dssi/hexter.so:hexter";
Chris@0: //	QString pluginId = "dssi:/usr/lib/dssi/sineshaper.so:sineshaper";
Chris@0: //	QString pluginId = "dssi:/usr/local/lib/dssi/xsynth-dssi.so:Xsynth";
Chris@0: //	QString pluginId = "dssi:/usr/local/lib/dssi/trivial_synth.so:TS";
Chris@0:     RealTimePluginFactory *factory =
Chris@0: 	RealTimePluginFactory::instanceFor(pluginId);
Chris@0:     
Chris@0:     if (!factory) {
Chris@0: 	std::cerr << "Failed to get plugin factory" << std::endl;
Chris@10: 	return false;
Chris@0:     }
Chris@0: 	
Chris@0:     RealTimePluginInstance *instance =
Chris@0: 	factory->instantiatePlugin
Chris@0: 	(pluginId, 0, 0, m_sourceSampleRate, m_pluginBlockSize, m_targetChannelCount);
Chris@0: 
Chris@0:     if (instance) {
Chris@0: 	for (unsigned int i = 0; i < instance->getParameterCount(); ++i) {
Chris@0: 	    instance->setParameterValue(i, instance->getParameterDefault(i));
Chris@0: 	}
Chris@11: 	QString defaultProgram = instance->getProgram(0, 0);
Chris@11: 	if (defaultProgram != "") {
Chris@11: 	    std::cerr << "first selecting default program " << defaultProgram.toLocal8Bit().data() << std::endl;
Chris@11: 	    instance->selectProgram(defaultProgram);
Chris@11: 	}
Chris@0: 	if (program != "") {
Chris@11: 	    std::cerr << "now selecting desired program " << program.toLocal8Bit().data() << std::endl;
Chris@0: 	    instance->selectProgram(program);
Chris@0: 	}
Chris@0: 	instance->setIdealChannelCount(m_targetChannelCount); // reset!
Chris@0:     } else {
Chris@0: 	std::cerr << "Failed to instantiate plugin" << std::endl;
Chris@0:     }
Chris@10: 
Chris@11:     return instance;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioGenerator::removeModel(Model *model)
Chris@0: {
Chris@0:     SparseOneDimensionalModel *sodm =
Chris@0: 	dynamic_cast<SparseOneDimensionalModel *>(model);
Chris@0:     if (!sodm) return; // nothing to do
Chris@0: 
Chris@8:     QMutexLocker locker(&m_mutex);
Chris@8: 
Chris@0:     if (m_synthMap.find(sodm) == m_synthMap.end()) return;
Chris@0: 
Chris@0:     RealTimePluginInstance *instance = m_synthMap[sodm];
Chris@0:     m_synthMap.erase(sodm);
Chris@0:     delete instance;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioGenerator::clearModels()
Chris@0: {
Chris@8:     QMutexLocker locker(&m_mutex);
Chris@0:     while (!m_synthMap.empty()) {
Chris@0: 	RealTimePluginInstance *instance = m_synthMap.begin()->second;
Chris@0: 	m_synthMap.erase(m_synthMap.begin());
Chris@0: 	delete instance;
Chris@0:     }
Chris@0: }    
Chris@0: 
Chris@0: void
Chris@0: AudioGenerator::reset()
Chris@0: {
Chris@8:     QMutexLocker locker(&m_mutex);
Chris@0:     for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
Chris@0: 	if (i->second) {
Chris@0: 	    i->second->silence();
Chris@0: 	    i->second->discardEvents();
Chris@0: 	}
Chris@0:     }
Chris@0: 
Chris@0:     m_noteOffs.clear();
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioGenerator::setTargetChannelCount(size_t targetChannelCount)
Chris@0: {
Chris@13:     if (m_targetChannelCount == targetChannelCount) return;
Chris@13: 
Chris@13:     std::cerr << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << std::endl;
Chris@13: 
Chris@8:     QMutexLocker locker(&m_mutex);
Chris@0:     m_targetChannelCount = targetChannelCount;
Chris@0: 
Chris@0:     for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
Chris@0: 	if (i->second) i->second->setIdealChannelCount(targetChannelCount);
Chris@0:     }
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioGenerator::getBlockSize() const
Chris@0: {
Chris@0:     return m_pluginBlockSize;
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioGenerator::mixModel(Model *model, size_t startFrame, size_t frameCount,
Chris@4: 			 float **buffer, size_t fadeIn, size_t fadeOut)
Chris@0: {
Chris@0:     if (m_sourceSampleRate == 0) {
Chris@0: 	std::cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << std::endl;
Chris@0: 	return frameCount;
Chris@0:     }
Chris@0: 
Chris@8:     QMutexLocker locker(&m_mutex);
Chris@8: 
Chris@11:     PlayParameters *parameters =
Chris@11: 	PlayParameterRepository::instance()->getPlayParameters(model);
Chris@0:     if (!parameters) return frameCount;
Chris@0: 
Chris@0:     bool playing = !parameters->isPlayMuted();
Chris@0:     if (!playing) return frameCount;
Chris@0: 
Chris@0:     float gain = parameters->getPlayGain();
Chris@0:     float pan = parameters->getPlayPan();
Chris@0: 
Chris@0:     DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
Chris@0:     if (dtvm) {
Chris@0: 	return mixDenseTimeValueModel(dtvm, startFrame, frameCount,
Chris@4: 				      buffer, gain, pan, fadeIn, fadeOut);
Chris@0:     }
Chris@0: 
Chris@0:     SparseOneDimensionalModel *sodm = dynamic_cast<SparseOneDimensionalModel *>
Chris@0: 	(model);
Chris@0:     if (sodm) {
Chris@0: 	return mixSparseOneDimensionalModel(sodm, startFrame, frameCount,
Chris@4: 					    buffer, gain, pan, fadeIn, fadeOut);
Chris@0:     }
Chris@0: 
Chris@11:     NoteModel *nm = dynamic_cast<NoteModel *>(model);
Chris@11:     if (nm) {
Chris@11: 	return mixNoteModel(nm, startFrame, frameCount,
Chris@11: 			    buffer, gain, pan, fadeIn, fadeOut);
Chris@11:     }
Chris@11: 
Chris@0:     return frameCount;
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm,
Chris@0: 				       size_t startFrame, size_t frames,
Chris@4: 				       float **buffer, float gain, float pan,
Chris@4: 				       size_t fadeIn, size_t fadeOut)
Chris@0: {
Chris@0:     static float *channelBuffer = 0;
Chris@0:     static size_t channelBufSiz = 0;
Chris@5: 
Chris@4:     size_t totalFrames = frames + fadeIn/2 + fadeOut/2;
Chris@4: 
Chris@4:     if (channelBufSiz < totalFrames) {
Chris@0: 	delete[] channelBuffer;
Chris@4: 	channelBuffer = new float[totalFrames];
Chris@4: 	channelBufSiz = totalFrames;
Chris@0:     }
Chris@0:     
Chris@0:     size_t got = 0;
Chris@13:     size_t prevChannel = 999;
Chris@0: 
Chris@13:     for (size_t c = 0; c < m_targetChannelCount; ++c) {
Chris@4: 
Chris@13: 	size_t sourceChannel = (c % dtvm->getChannelCount());
Chris@13: 
Chris@13: //	std::cerr << "mixing channel " << c << " from source channel " << sourceChannel << std::endl;
Chris@13: 
Chris@13: 	float channelGain = gain;
Chris@13: 	if (pan != 0.0) {
Chris@13: 	    if (c == 0) {
Chris@13: 		if (pan > 0.0) channelGain *= 1.0 - pan;
Chris@13: 	    } else {
Chris@13: 		if (pan < 0.0) channelGain *= pan + 1.0;
Chris@13: 	    }
Chris@13: 	}
Chris@13: 
Chris@13: 	if (prevChannel != sourceChannel) {
Chris@13: 	    if (startFrame >= fadeIn/2) {
Chris@13: 		got = dtvm->getValues
Chris@13: 		    (sourceChannel,
Chris@13: 		     startFrame - fadeIn/2, startFrame + frames + fadeOut/2,
Chris@13: 		     channelBuffer);
Chris@13: 	    } else {
Chris@13: 		size_t missing = fadeIn/2 - startFrame;
Chris@13: 		got = dtvm->getValues
Chris@13: 		    (sourceChannel,
Chris@13: 		     0, startFrame + frames + fadeOut/2,
Chris@13: 		     channelBuffer + missing);
Chris@13: 	    }	    
Chris@13: 	}
Chris@13: 	prevChannel = sourceChannel;
Chris@4: 
Chris@4: 	for (size_t i = 0; i < fadeIn/2; ++i) {
Chris@4: 	    float *back = buffer[c];
Chris@4: 	    back -= fadeIn/2;
Chris@13: 	    back[i] += (channelGain * channelBuffer[i] * i) / fadeIn;
Chris@4: 	}
Chris@4: 
Chris@4: 	for (size_t i = 0; i < frames + fadeOut/2; ++i) {
Chris@13: 	    float mult = channelGain;
Chris@4: 	    if (i < fadeIn/2) {
Chris@4: 		mult = (mult * i) / fadeIn;
Chris@4: 	    }
Chris@4: 	    if (i > frames - fadeOut/2) {
Chris@4: 		mult = (mult * ((frames + fadeOut/2) - i)) / fadeOut;
Chris@4: 	    }
Chris@4: 	    buffer[c][i] += mult * channelBuffer[i];
Chris@0: 	}
Chris@0:     }
Chris@0: 
Chris@0:     return got;
Chris@0: }
Chris@11:   
Chris@0: size_t
Chris@0: AudioGenerator::mixSparseOneDimensionalModel(SparseOneDimensionalModel *sodm,
Chris@0: 					     size_t startFrame, size_t frames,
Chris@4: 					     float **buffer, float gain, float pan,
Chris@4: 					     size_t /* fadeIn */,
Chris@4: 					     size_t /* fadeOut */)
Chris@0: {
Chris@0:     RealTimePluginInstance *plugin = m_synthMap[sodm];
Chris@0:     if (!plugin) return 0;
Chris@0: 
Chris@0:     size_t latency = plugin->getLatency();
Chris@0:     size_t blocks = frames / m_pluginBlockSize;
Chris@0:     
Chris@0:     //!!! hang on -- the fact that the audio callback play source's
Chris@0:     //buffer is a multiple of the plugin's buffer size doesn't mean
Chris@0:     //that we always get called for a multiple of it here (because it
Chris@0:     //also depends on the JACK block size).  how should we ensure that
Chris@0:     //all models write the same amount in to the mix, and that we
Chris@0:     //always have a multiple of the plugin buffer size?  I guess this
Chris@0:     //class has to be queryable for the plugin buffer size & the
Chris@0:     //callback play source has to use that as a multiple for all the
Chris@0:     //calls to mixModel
Chris@0: 
Chris@0:     size_t got = blocks * m_pluginBlockSize;
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_GENERATOR
Chris@0:     std::cout << "mixModel [sparse]: frames " << frames
Chris@0: 	      << ", blocks " << blocks << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0:     snd_seq_event_t onEv;
Chris@0:     onEv.type = SND_SEQ_EVENT_NOTEON;
Chris@0:     onEv.data.note.channel = 0;
Chris@0:     onEv.data.note.note = 64;
Chris@0:     onEv.data.note.velocity = 127;
Chris@0: 
Chris@0:     snd_seq_event_t offEv;
Chris@0:     offEv.type = SND_SEQ_EVENT_NOTEOFF;
Chris@0:     offEv.data.note.channel = 0;
Chris@0:     offEv.data.note.velocity = 0;
Chris@0:     
Chris@0:     NoteOffSet &noteOffs = m_noteOffs[sodm];
Chris@0: 
Chris@0:     for (size_t i = 0; i < blocks; ++i) {
Chris@0: 
Chris@0: 	size_t reqStart = startFrame + i * m_pluginBlockSize;
Chris@0: 
Chris@0: 	SparseOneDimensionalModel::PointList points =
Chris@5: 	    sodm->getPoints(reqStart + latency,
Chris@0: 			    reqStart + latency + m_pluginBlockSize);
Chris@0: 
Chris@0: 	RealTime blockTime = RealTime::frame2RealTime
Chris@0: 	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);
Chris@0: 
Chris@0: 	for (SparseOneDimensionalModel::PointList::iterator pli =
Chris@0: 		 points.begin(); pli != points.end(); ++pli) {
Chris@0: 
Chris@0: 	    size_t pliFrame = pli->frame;
Chris@5: 
Chris@0: 	    if (pliFrame >= latency) pliFrame -= latency;
Chris@0: 
Chris@5: 	    if (pliFrame < reqStart ||
Chris@5: 		pliFrame >= reqStart + m_pluginBlockSize) continue;
Chris@5: 
Chris@0: 	    while (noteOffs.begin() != noteOffs.end() &&
Chris@0: 		   noteOffs.begin()->frame <= pliFrame) {
Chris@0: 
Chris@0: 		RealTime eventTime = RealTime::frame2RealTime
Chris@0: 		    (noteOffs.begin()->frame, m_sourceSampleRate);
Chris@0: 
Chris@0: 		offEv.data.note.note = noteOffs.begin()->pitch;
Chris@5: 
Chris@5: #ifdef DEBUG_AUDIO_GENERATOR
Chris@5: 		std::cerr << "mixModel [sparse]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
Chris@5: #endif
Chris@5: 
Chris@0: 		plugin->sendEvent(eventTime, &offEv);
Chris@0: 		noteOffs.erase(noteOffs.begin());
Chris@0: 	    }
Chris@0: 
Chris@0: 	    RealTime eventTime = RealTime::frame2RealTime
Chris@0: 		(pliFrame, m_sourceSampleRate);
Chris@0: 	    
Chris@0: 	    plugin->sendEvent(eventTime, &onEv);
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_GENERATOR
Chris@0: 	    std::cout << "mixModel [sparse]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
Chris@0: #endif
Chris@0: 	    
Chris@0: 	    size_t duration = 7000; // frames [for now]
Chris@0: 	    NoteOff noff;
Chris@0: 	    noff.pitch = onEv.data.note.note;
Chris@0: 	    noff.frame = pliFrame + duration;
Chris@0: 	    noteOffs.insert(noff);
Chris@0: 	}
Chris@0: 
Chris@0: 	while (noteOffs.begin() != noteOffs.end() &&
Chris@0: 	       noteOffs.begin()->frame <=
Chris@0: 	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {
Chris@0: 
Chris@0: 	    RealTime eventTime = RealTime::frame2RealTime
Chris@0: 		(noteOffs.begin()->frame, m_sourceSampleRate);
Chris@0: 
Chris@0: 	    offEv.data.note.note = noteOffs.begin()->pitch;
Chris@5: 
Chris@5: #ifdef DEBUG_AUDIO_GENERATOR
Chris@5: 		std::cerr << "mixModel [sparse]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
Chris@5: #endif
Chris@5: 
Chris@0: 	    plugin->sendEvent(eventTime, &offEv);
Chris@0: 	    noteOffs.erase(noteOffs.begin());
Chris@0: 	}
Chris@0: 	
Chris@0: 	plugin->run(blockTime);
Chris@0: 	float **outs = plugin->getAudioOutputBuffers();
Chris@0: 
Chris@13: 	for (size_t c = 0; c < m_targetChannelCount; ++c) {
Chris@0: #ifdef DEBUG_AUDIO_GENERATOR
Chris@0: 	    std::cout << "mixModel [sparse]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
Chris@0: #endif
Chris@0: 
Chris@13: 	    size_t sourceChannel = (c % plugin->getAudioOutputCount());
Chris@13: 
Chris@13: 	    float channelGain = gain;
Chris@13: 	    if (pan != 0.0) {
Chris@13: 		if (c == 0) {
Chris@13: 		    if (pan > 0.0) channelGain *= 1.0 - pan;
Chris@13: 		} else {
Chris@13: 		    if (pan < 0.0) channelGain *= pan + 1.0;
Chris@13: 		}
Chris@13: 	    }
Chris@13: 
Chris@0: 	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
Chris@13: 		buffer[c][i * m_pluginBlockSize + j] +=
Chris@13: 		    channelGain * outs[sourceChannel][j];
Chris@0: 	    }
Chris@0: 	}
Chris@0:     }
Chris@0: 
Chris@0:     return got;
Chris@0: }
Chris@0: 
Chris@11:     
Chris@11: //!!! mucho duplication with above -- refactor
Chris@11: size_t
Chris@11: AudioGenerator::mixNoteModel(NoteModel *nm,
Chris@11: 			     size_t startFrame, size_t frames,
Chris@11: 			     float **buffer, float gain, float pan,
Chris@11: 			     size_t /* fadeIn */,
Chris@11: 			     size_t /* fadeOut */)
Chris@11: {
Chris@11:     RealTimePluginInstance *plugin = m_synthMap[nm];
Chris@11:     if (!plugin) return 0;
Chris@11: 
Chris@11:     size_t latency = plugin->getLatency();
Chris@11:     size_t blocks = frames / m_pluginBlockSize;
Chris@11:     
Chris@11:     //!!! hang on -- the fact that the audio callback play source's
Chris@11:     //buffer is a multiple of the plugin's buffer size doesn't mean
Chris@11:     //that we always get called for a multiple of it here (because it
Chris@11:     //also depends on the JACK block size).  how should we ensure that
Chris@11:     //all models write the same amount in to the mix, and that we
Chris@11:     //always have a multiple of the plugin buffer size?  I guess this
Chris@11:     //class has to be queryable for the plugin buffer size & the
Chris@11:     //callback play source has to use that as a multiple for all the
Chris@11:     //calls to mixModel
Chris@11: 
Chris@11:     size_t got = blocks * m_pluginBlockSize;
Chris@11: 
Chris@11: #ifdef DEBUG_AUDIO_GENERATOR
Chris@11:     std::cout << "mixModel [note]: frames " << frames
Chris@11: 	      << ", blocks " << blocks << std::endl;
Chris@11: #endif
Chris@11: 
Chris@11:     snd_seq_event_t onEv;
Chris@11:     onEv.type = SND_SEQ_EVENT_NOTEON;
Chris@11:     onEv.data.note.channel = 0;
Chris@11:     onEv.data.note.note = 64;
Chris@11:     onEv.data.note.velocity = 127;
Chris@11: 
Chris@11:     snd_seq_event_t offEv;
Chris@11:     offEv.type = SND_SEQ_EVENT_NOTEOFF;
Chris@11:     offEv.data.note.channel = 0;
Chris@11:     offEv.data.note.velocity = 0;
Chris@11:     
Chris@11:     NoteOffSet &noteOffs = m_noteOffs[nm];
Chris@11: 
Chris@11:     for (size_t i = 0; i < blocks; ++i) {
Chris@11: 
Chris@11: 	size_t reqStart = startFrame + i * m_pluginBlockSize;
Chris@11: 
Chris@11: 	NoteModel::PointList points =
Chris@11: 	    nm->getPoints(reqStart + latency,
Chris@11: 			    reqStart + latency + m_pluginBlockSize);
Chris@11: 
Chris@11: 	RealTime blockTime = RealTime::frame2RealTime
Chris@11: 	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);
Chris@11: 
Chris@11: 	for (NoteModel::PointList::iterator pli =
Chris@11: 		 points.begin(); pli != points.end(); ++pli) {
Chris@11: 
Chris@11: 	    size_t pliFrame = pli->frame;
Chris@11: 
Chris@11: 	    if (pliFrame >= latency) pliFrame -= latency;
Chris@11: 
Chris@11: 	    if (pliFrame < reqStart ||
Chris@11: 		pliFrame >= reqStart + m_pluginBlockSize) continue;
Chris@11: 
Chris@11: 	    while (noteOffs.begin() != noteOffs.end() &&
Chris@11: 		   noteOffs.begin()->frame <= pliFrame) {
Chris@11: 
Chris@11: 		RealTime eventTime = RealTime::frame2RealTime
Chris@11: 		    (noteOffs.begin()->frame, m_sourceSampleRate);
Chris@11: 
Chris@11: 		offEv.data.note.note = noteOffs.begin()->pitch;
Chris@11: 
Chris@11: #ifdef DEBUG_AUDIO_GENERATOR
Chris@11: 		std::cerr << "mixModel [note]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
Chris@11: #endif
Chris@11: 
Chris@11: 		plugin->sendEvent(eventTime, &offEv);
Chris@11: 		noteOffs.erase(noteOffs.begin());
Chris@11: 	    }
Chris@11: 
Chris@11: 	    RealTime eventTime = RealTime::frame2RealTime
Chris@11: 		(pliFrame, m_sourceSampleRate);
Chris@11: 	    
Chris@11: 	    onEv.data.note.note = lrintf(pli->value);
Chris@11: 
Chris@11: 	    plugin->sendEvent(eventTime, &onEv);
Chris@11: 
Chris@11: #ifdef DEBUG_AUDIO_GENERATOR
Chris@11: 	    std::cout << "mixModel [note]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
Chris@11: #endif
Chris@11: 	    
Chris@11: 	    size_t duration = pli->duration;
Chris@11: 	    NoteOff noff;
Chris@11: 	    noff.pitch = onEv.data.note.note;
Chris@11: 	    noff.frame = pliFrame + duration;
Chris@11: 	    noteOffs.insert(noff);
Chris@11: 	}
Chris@11: 
Chris@11: 	while (noteOffs.begin() != noteOffs.end() &&
Chris@11: 	       noteOffs.begin()->frame <=
Chris@11: 	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {
Chris@11: 
Chris@11: 	    RealTime eventTime = RealTime::frame2RealTime
Chris@11: 		(noteOffs.begin()->frame, m_sourceSampleRate);
Chris@11: 
Chris@11: 	    offEv.data.note.note = noteOffs.begin()->pitch;
Chris@11: 
Chris@11: #ifdef DEBUG_AUDIO_GENERATOR
Chris@11: 		std::cerr << "mixModel [note]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
Chris@11: #endif
Chris@11: 
Chris@11: 	    plugin->sendEvent(eventTime, &offEv);
Chris@11: 	    noteOffs.erase(noteOffs.begin());
Chris@11: 	}
Chris@11: 	
Chris@11: 	plugin->run(blockTime);
Chris@11: 	float **outs = plugin->getAudioOutputBuffers();
Chris@11: 
Chris@13: 	for (size_t c = 0; c < m_targetChannelCount; ++c) {
Chris@11: #ifdef DEBUG_AUDIO_GENERATOR
Chris@11: 	    std::cout << "mixModel [note]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
Chris@11: #endif
Chris@11: 
Chris@13: 	    size_t sourceChannel = (c % plugin->getAudioOutputCount());
Chris@13: 
Chris@13: 	    float channelGain = gain;
Chris@13: 	    if (pan != 0.0) {
Chris@13: 		if (c == 0) {
Chris@13: 		    if (pan > 0.0) channelGain *= 1.0 - pan;
Chris@13: 		} else {
Chris@13: 		    if (pan < 0.0) channelGain *= pan + 1.0;
Chris@13: 		}
Chris@13: 	    }
Chris@13: 
Chris@11: 	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
Chris@13: 		buffer[c][i * m_pluginBlockSize + j] += 
Chris@13: 		    channelGain * outs[sourceChannel][j];
Chris@11: 	    }
Chris@11: 	}
Chris@11:     }
Chris@11: 
Chris@11:     return got;
Chris@11: }
Chris@11: