annotate audioio/AudioGenerator.cpp @ 140:9ccaa8fd9b9f

* Add code to identify (usually) the type of an XML file that contains
  either SV session or layer data, and use it to permit loading files
  with plain .xml extension that contain complete uncompression sessions

/* -*- 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 "AudioGenerator.h"

#include "base/TempDirectory.h"
#include "base/PlayParameters.h"
#include "base/PlayParameterRepository.h"
#include "base/Pitch.h"
#include "base/Exceptions.h"

#include "data/model/NoteModel.h"
#include "data/model/DenseTimeValueModel.h"
#include "data/model/SparseOneDimensionalModel.h"

#include "plugin/RealTimePluginFactory.h"
#include "plugin/RealTimePluginInstance.h"
#include "plugin/PluginIdentifier.h"
#include "plugin/PluginXml.h"
#include "plugin/api/alsa/seq_event.h"

#include <iostream>
#include <math.h>

#include <QDir>
#include <QFile>

const size_t
AudioGenerator::m_pluginBlockSize = 2048;

QString
AudioGenerator::m_sampleDir = "";

//#define DEBUG_AUDIO_GENERATOR 1

AudioGenerator::AudioGenerator() :
    m_sourceSampleRate(0),
    m_targetChannelCount(1),
    m_soloing(false)
{
    initialiseSampleDir();

    connect(PlayParameterRepository::getInstance(),
            SIGNAL(playPluginIdChanged(const Playable *, QString)),
            this,
            SLOT(playPluginIdChanged(const Playable *, QString)));

    connect(PlayParameterRepository::getInstance(),
            SIGNAL(playPluginConfigurationChanged(const Playable *, QString)),
            this,
            SLOT(playPluginConfigurationChanged(const Playable *, QString)));
}

AudioGenerator::~AudioGenerator()
{
}

void
AudioGenerator::initialiseSampleDir()
{
    if (m_sampleDir != "") return;

    try {
        m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples");
    } catch (DirectoryCreationFailed f) {
        std::cerr << "WARNING: AudioGenerator::initialiseSampleDir:"
                  << " Failed to create temporary sample directory"
                  << std::endl;
        m_sampleDir = "";
        return;
    }

    QDir sampleResourceDir(":/samples", "*.wav");

    for (unsigned int i = 0; i < sampleResourceDir.count(); ++i) {

        QString fileName(sampleResourceDir[i]);
        QFile file(sampleResourceDir.filePath(fileName));

        if (!file.copy(QDir(m_sampleDir).filePath(fileName))) {
            std::cerr << "WARNING: AudioGenerator::getSampleDir: "
                      << "Unable to copy " << fileName.toStdString()
                      << " into temporary directory \""
                      << m_sampleDir.toStdString() << "\"" << std::endl;
        }
    }
}

bool
AudioGenerator::addModel(Model *model)
{
    if (m_sourceSampleRate == 0) {

	m_sourceSampleRate = model->getSampleRate();

    } else {

	DenseTimeValueModel *dtvm =
	    dynamic_cast<DenseTimeValueModel *>(model);

	if (dtvm) {
	    m_sourceSampleRate = model->getSampleRate();
	    return true;
	}
    }

    RealTimePluginInstance *plugin = loadPluginFor(model);
    if (plugin) {
        QMutexLocker locker(&m_mutex);
        m_synthMap[model] = plugin;
        return true;
    }

    return false;
}

void
AudioGenerator::playPluginIdChanged(const Playable *playable, QString)
{
    const Model *model = dynamic_cast<const Model *>(playable);
    if (!model) {
        std::cerr << "WARNING: AudioGenerator::playPluginIdChanged: playable "
                  << playable << " is not a supported model type"
                  << std::endl;
        return;
    }

    if (m_synthMap.find(model) == m_synthMap.end()) return;
    
    RealTimePluginInstance *plugin = loadPluginFor(model);
    if (plugin) {
        QMutexLocker locker(&m_mutex);
        delete m_synthMap[model];
        m_synthMap[model] = plugin;
    }
}

void
AudioGenerator::playPluginConfigurationChanged(const Playable *playable,
                                               QString configurationXml)
{
//    std::cerr << "AudioGenerator::playPluginConfigurationChanged" << std::endl;

    const Model *model = dynamic_cast<const Model *>(playable);
    if (!model) {
        std::cerr << "WARNING: AudioGenerator::playPluginIdChanged: playable "
                  << playable << " is not a supported model type"
                  << std::endl;
        return;
    }

    if (m_synthMap.find(model) == m_synthMap.end()) {
        std::cerr << "AudioGenerator::playPluginConfigurationChanged: We don't know about this plugin" << std::endl;
        return;
    }

    RealTimePluginInstance *plugin = m_synthMap[model];
    if (plugin) {
        PluginXml(plugin).setParametersFromXml(configurationXml);
    }
}

void
AudioGenerator::setSampleDir(RealTimePluginInstance *plugin)
{
    if (m_sampleDir != "") {
        plugin->configure("sampledir", m_sampleDir.toStdString());
    }
} 

RealTimePluginInstance *
AudioGenerator::loadPluginFor(const Model *model)
{
    QString pluginId, configurationXml;

    const Playable *playable = model;
    if (!playable || !playable->canPlay()) return 0;

    PlayParameters *parameters =
	PlayParameterRepository::getInstance()->getPlayParameters(playable);
    if (parameters) {
        pluginId = parameters->getPlayPluginId();
        configurationXml = parameters->getPlayPluginConfiguration();
    }

    if (pluginId == "") return 0;

    RealTimePluginInstance *plugin = loadPlugin(pluginId, "");
    if (!plugin) return 0;

    if (configurationXml != "") {
        PluginXml(plugin).setParametersFromXml(configurationXml);
    }

    configurationXml = PluginXml(plugin).toXmlString();

    if (parameters) {
        parameters->setPlayPluginId(pluginId);
        parameters->setPlayPluginConfiguration(configurationXml);
    }

    return plugin;
}

RealTimePluginInstance *
AudioGenerator::loadPlugin(QString pluginId, QString program)
{
    RealTimePluginFactory *factory =
	RealTimePluginFactory::instanceFor(pluginId);
    
    if (!factory) {
	std::cerr << "Failed to get plugin factory" << std::endl;
	return false;
    }
	
    RealTimePluginInstance *instance =
	factory->instantiatePlugin
	(pluginId, 0, 0, m_sourceSampleRate, m_pluginBlockSize, m_targetChannelCount);

    if (!instance) {
	std::cerr << "Failed to instantiate plugin " << pluginId.toStdString() << std::endl;
        return 0;
    }

    setSampleDir(instance);

    for (unsigned int i = 0; i < instance->getParameterCount(); ++i) {
        instance->setParameterValue(i, instance->getParameterDefault(i));
    }
    std::string defaultProgram = instance->getProgram(0, 0);
    if (defaultProgram != "") {
//        std::cerr << "first selecting default program " << defaultProgram << std::endl;
        instance->selectProgram(defaultProgram);
    }
    if (program != "") {
//        std::cerr << "now selecting desired program " << program.toStdString() << std::endl;
        instance->selectProgram(program.toStdString());
    }
    instance->setIdealChannelCount(m_targetChannelCount); // reset!

    return instance;
}

void
AudioGenerator::removeModel(Model *model)
{
    SparseOneDimensionalModel *sodm =
	dynamic_cast<SparseOneDimensionalModel *>(model);
    if (!sodm) return; // nothing to do

    QMutexLocker locker(&m_mutex);

    if (m_synthMap.find(sodm) == m_synthMap.end()) return;

    RealTimePluginInstance *instance = m_synthMap[sodm];
    m_synthMap.erase(sodm);
    delete instance;
}

void
AudioGenerator::clearModels()
{
    QMutexLocker locker(&m_mutex);
    while (!m_synthMap.empty()) {
	RealTimePluginInstance *instance = m_synthMap.begin()->second;
	m_synthMap.erase(m_synthMap.begin());
	delete instance;
    }
}    

void
AudioGenerator::reset()
{
    QMutexLocker locker(&m_mutex);
    for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
	if (i->second) {
	    i->second->silence();
	    i->second->discardEvents();
	}
    }

    m_noteOffs.clear();
}

void
AudioGenerator::setTargetChannelCount(size_t targetChannelCount)
{
    if (m_targetChannelCount == targetChannelCount) return;

//    std::cerr << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << std::endl;

    QMutexLocker locker(&m_mutex);
    m_targetChannelCount = targetChannelCount;

    for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
	if (i->second) i->second->setIdealChannelCount(targetChannelCount);
    }
}

size_t
AudioGenerator::getBlockSize() const
{
    return m_pluginBlockSize;
}

void
AudioGenerator::setSoloModelSet(std::set<Model *> s)
{
    QMutexLocker locker(&m_mutex);

    m_soloModelSet = s;
    m_soloing = true;
}

void
AudioGenerator::clearSoloModelSet()
{
    QMutexLocker locker(&m_mutex);

    m_soloModelSet.clear();
    m_soloing = false;
}

size_t
AudioGenerator::mixModel(Model *model, size_t startFrame, size_t frameCount,
			 float **buffer, size_t fadeIn, size_t fadeOut)
{
    if (m_sourceSampleRate == 0) {
	std::cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << std::endl;
	return frameCount;
    }

    QMutexLocker locker(&m_mutex);

    Playable *playable = model;
    if (!playable || !playable->canPlay()) return frameCount;

    PlayParameters *parameters =
	PlayParameterRepository::getInstance()->getPlayParameters(playable);
    if (!parameters) return frameCount;

    bool playing = !parameters->isPlayMuted();
    if (!playing) {
#ifdef DEBUG_AUDIO_GENERATOR
        std::cout << "AudioGenerator::mixModel(" << model << "): muted" << std::endl;
#endif
        return frameCount;
    }

    if (m_soloing) {
        if (m_soloModelSet.find(model) == m_soloModelSet.end()) {
#ifdef DEBUG_AUDIO_GENERATOR
            std::cout << "AudioGenerator::mixModel(" << model << "): not one of the solo'd models" << std::endl;
#endif
            return frameCount;
        }
    }

    float gain = parameters->getPlayGain();
    float pan = parameters->getPlayPan();

    DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
    if (dtvm) {
	return mixDenseTimeValueModel(dtvm, startFrame, frameCount,
				      buffer, gain, pan, fadeIn, fadeOut);
    }

    SparseOneDimensionalModel *sodm = dynamic_cast<SparseOneDimensionalModel *>
	(model);
    if (sodm) {
	return mixSparseOneDimensionalModel(sodm, startFrame, frameCount,
					    buffer, gain, pan, fadeIn, fadeOut);
    }

    NoteModel *nm = dynamic_cast<NoteModel *>(model);
    if (nm) {
	return mixNoteModel(nm, startFrame, frameCount,
			    buffer, gain, pan, fadeIn, fadeOut);
    }

    return frameCount;
}

size_t
AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm,
				       size_t startFrame, size_t frames,
				       float **buffer, float gain, float pan,
				       size_t fadeIn, size_t fadeOut)
{
    static float **channelBuffer = 0;
    static size_t  channelBufSiz = 0;
    static size_t  channelBufCount = 0;

    size_t totalFrames = frames + fadeIn/2 + fadeOut/2;

    size_t modelChannels = dtvm->getChannelCount();

    if (channelBufSiz < totalFrames || channelBufCount < modelChannels) {

        for (size_t c = 0; c < channelBufCount; ++c) {
            delete[] channelBuffer[c];
        }

	delete[] channelBuffer;
        channelBuffer = new float *[modelChannels];

        for (size_t c = 0; c < modelChannels; ++c) {
            channelBuffer[c] = new float[totalFrames];
        }

        channelBufCount = modelChannels;
	channelBufSiz = totalFrames;
    }

    size_t got = 0;

    if (startFrame >= fadeIn/2) {
        got = dtvm->getData(0, modelChannels - 1,
                            startFrame - fadeIn/2,
                            frames + fadeOut/2 + fadeIn/2,
                            channelBuffer);
    } else {
        size_t missing = fadeIn/2 - startFrame;

        for (size_t c = 0; c < modelChannels; ++c) {
            channelBuffer[c] += missing;
        }

        got = dtvm->getData(0, modelChannels - 1,
                            startFrame,
                            frames + fadeOut/2,
                            channelBuffer);

        for (size_t c = 0; c < modelChannels; ++c) {
            channelBuffer[c] -= missing;
        }

        got += missing;
    }	    

    for (size_t c = 0; c < m_targetChannelCount; ++c) {

	size_t sourceChannel = (c % modelChannels);

//	std::cerr << "mixing channel " << c << " from source channel " << sourceChannel << std::endl;

	float channelGain = gain;
	if (pan != 0.0) {
	    if (c == 0) {
		if (pan > 0.0) channelGain *= 1.0 - pan;
	    } else {
		if (pan < 0.0) channelGain *= pan + 1.0;
	    }
	}

	for (size_t i = 0; i < fadeIn/2; ++i) {
	    float *back = buffer[c];
	    back -= fadeIn/2;
	    back[i] += (channelGain * channelBuffer[sourceChannel][i] * i) / fadeIn;
	}

	for (size_t i = 0; i < frames + fadeOut/2; ++i) {
	    float mult = channelGain;
	    if (i < fadeIn/2) {
		mult = (mult * i) / fadeIn;
	    }
	    if (i > frames - fadeOut/2) {
		mult = (mult * ((frames + fadeOut/2) - i)) / fadeOut;
	    }
            float val = channelBuffer[sourceChannel][i];
            if (i >= got) val = 0.f;
	    buffer[c][i] += mult * val;
	}
    }

    return got;
}
  
size_t
AudioGenerator::mixSparseOneDimensionalModel(SparseOneDimensionalModel *sodm,
					     size_t startFrame, size_t frames,
					     float **buffer, float gain, float pan,
					     size_t /* fadeIn */,
					     size_t /* fadeOut */)
{
    RealTimePluginInstance *plugin = m_synthMap[sodm];
    if (!plugin) return 0;

    size_t latency = plugin->getLatency();
    size_t blocks = frames / m_pluginBlockSize;
    
    //!!! hang on -- the fact that the audio callback play source's
    //buffer is a multiple of the plugin's buffer size doesn't mean
    //that we always get called for a multiple of it here (because it
    //also depends on the JACK block size).  how should we ensure that
    //all models write the same amount in to the mix, and that we
    //always have a multiple of the plugin buffer size?  I guess this
    //class has to be queryable for the plugin buffer size & the
    //callback play source has to use that as a multiple for all the
    //calls to mixModel

    size_t got = blocks * m_pluginBlockSize;

#ifdef DEBUG_AUDIO_GENERATOR
    std::cout << "mixModel [sparse]: frames " << frames
	      << ", blocks " << blocks << std::endl;
#endif

    snd_seq_event_t onEv;
    onEv.type = SND_SEQ_EVENT_NOTEON;
    onEv.data.note.channel = 0;
    onEv.data.note.note = 64;
    onEv.data.note.velocity = 100;

    snd_seq_event_t offEv;
    offEv.type = SND_SEQ_EVENT_NOTEOFF;
    offEv.data.note.channel = 0;
    offEv.data.note.velocity = 0;
    
    NoteOffSet &noteOffs = m_noteOffs[sodm];

    for (size_t i = 0; i < blocks; ++i) {

	size_t reqStart = startFrame + i * m_pluginBlockSize;

	SparseOneDimensionalModel::PointList points =
	    sodm->getPoints(reqStart + latency,
			    reqStart + latency + m_pluginBlockSize);

        Vamp::RealTime blockTime = Vamp::RealTime::frame2RealTime
	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);

	for (SparseOneDimensionalModel::PointList::iterator pli =
		 points.begin(); pli != points.end(); ++pli) {

	    size_t pliFrame = pli->frame;

	    if (pliFrame >= latency) pliFrame -= latency;

	    if (pliFrame < reqStart ||
		pliFrame >= reqStart + m_pluginBlockSize) continue;

	    while (noteOffs.begin() != noteOffs.end() &&
		   noteOffs.begin()->frame <= pliFrame) {

                Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
		    (noteOffs.begin()->frame, m_sourceSampleRate);

		offEv.data.note.note = noteOffs.begin()->pitch;

#ifdef DEBUG_AUDIO_GENERATOR
		std::cerr << "mixModel [sparse]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
#endif

		plugin->sendEvent(eventTime, &offEv);
		noteOffs.erase(noteOffs.begin());
	    }

            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
		(pliFrame, m_sourceSampleRate);
	    
	    plugin->sendEvent(eventTime, &onEv);

#ifdef DEBUG_AUDIO_GENERATOR
	    std::cout << "mixModel [sparse]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
#endif
	    
	    size_t duration = 7000; // frames [for now]
	    NoteOff noff;
	    noff.pitch = onEv.data.note.note;
	    noff.frame = pliFrame + duration;
	    noteOffs.insert(noff);
	}

	while (noteOffs.begin() != noteOffs.end() &&
	       noteOffs.begin()->frame <=
	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {

            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
		(noteOffs.begin()->frame, m_sourceSampleRate);

	    offEv.data.note.note = noteOffs.begin()->pitch;

#ifdef DEBUG_AUDIO_GENERATOR
		std::cerr << "mixModel [sparse]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
#endif

	    plugin->sendEvent(eventTime, &offEv);
	    noteOffs.erase(noteOffs.begin());
	}
	
	plugin->run(blockTime);
	float **outs = plugin->getAudioOutputBuffers();

	for (size_t c = 0; c < m_targetChannelCount; ++c) {
#ifdef DEBUG_AUDIO_GENERATOR
	    std::cout << "mixModel [sparse]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
#endif

	    size_t sourceChannel = (c % plugin->getAudioOutputCount());

	    float channelGain = gain;
	    if (pan != 0.0) {
		if (c == 0) {
		    if (pan > 0.0) channelGain *= 1.0 - pan;
		} else {
		    if (pan < 0.0) channelGain *= pan + 1.0;
		}
	    }

	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
		buffer[c][i * m_pluginBlockSize + j] +=
		    channelGain * outs[sourceChannel][j];
	    }
	}
    }

    return got;
}

    
//!!! mucho duplication with above -- refactor
size_t
AudioGenerator::mixNoteModel(NoteModel *nm,
			     size_t startFrame, size_t frames,
			     float **buffer, float gain, float pan,
			     size_t /* fadeIn */,
			     size_t /* fadeOut */)
{
    RealTimePluginInstance *plugin = m_synthMap[nm];
    if (!plugin) return 0;

    size_t latency = plugin->getLatency();
    size_t blocks = frames / m_pluginBlockSize;
    
    //!!! hang on -- the fact that the audio callback play source's
    //buffer is a multiple of the plugin's buffer size doesn't mean
    //that we always get called for a multiple of it here (because it
    //also depends on the JACK block size).  how should we ensure that
    //all models write the same amount in to the mix, and that we
    //always have a multiple of the plugin buffer size?  I guess this
    //class has to be queryable for the plugin buffer size & the
    //callback play source has to use that as a multiple for all the
    //calls to mixModel

    size_t got = blocks * m_pluginBlockSize;

#ifdef DEBUG_AUDIO_GENERATOR
    std::cout << "mixModel [note]: frames " << frames
	      << ", blocks " << blocks << std::endl;
#endif

    snd_seq_event_t onEv;
    onEv.type = SND_SEQ_EVENT_NOTEON;
    onEv.data.note.channel = 0;
    onEv.data.note.note = 64;
    onEv.data.note.velocity = 100;

    snd_seq_event_t offEv;
    offEv.type = SND_SEQ_EVENT_NOTEOFF;
    offEv.data.note.channel = 0;
    offEv.data.note.velocity = 0;
    
    NoteOffSet &noteOffs = m_noteOffs[nm];

    for (size_t i = 0; i < blocks; ++i) {

	size_t reqStart = startFrame + i * m_pluginBlockSize;

	NoteModel::PointList points =
	    nm->getPoints(reqStart + latency,
			    reqStart + latency + m_pluginBlockSize);

        Vamp::RealTime blockTime = Vamp::RealTime::frame2RealTime
	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);

	for (NoteModel::PointList::iterator pli =
		 points.begin(); pli != points.end(); ++pli) {

	    size_t pliFrame = pli->frame;

	    if (pliFrame >= latency) pliFrame -= latency;

	    if (pliFrame < reqStart ||
		pliFrame >= reqStart + m_pluginBlockSize) continue;

	    while (noteOffs.begin() != noteOffs.end() &&
		   noteOffs.begin()->frame <= pliFrame) {

                Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
		    (noteOffs.begin()->frame, m_sourceSampleRate);

		offEv.data.note.note = noteOffs.begin()->pitch;

#ifdef DEBUG_AUDIO_GENERATOR
		std::cerr << "mixModel [note]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
#endif

		plugin->sendEvent(eventTime, &offEv);
		noteOffs.erase(noteOffs.begin());
	    }

            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
		(pliFrame, m_sourceSampleRate);
	    
            if (nm->getScaleUnits() == "Hz") {
                onEv.data.note.note = Pitch::getPitchForFrequency(pli->value);
            } else {
                onEv.data.note.note = lrintf(pli->value);
            }

            if (pli->level > 0.f && pli->level <= 1.f) {
                onEv.data.note.velocity = lrintf(pli->level * 127);
            } else {
                onEv.data.note.velocity = 100;
            }

	    plugin->sendEvent(eventTime, &onEv);

#ifdef DEBUG_AUDIO_GENERATOR
	    std::cout << "mixModel [note]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
#endif
	    
	    size_t duration = pli->duration;
            if (duration == 0 || duration == 1) {
                duration = m_sourceSampleRate / 20;
            }
	    NoteOff noff;
	    noff.pitch = onEv.data.note.note;
	    noff.frame = pliFrame + duration;
	    noteOffs.insert(noff);
	}

	while (noteOffs.begin() != noteOffs.end() &&
	       noteOffs.begin()->frame <=
	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {

            Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime
		(noteOffs.begin()->frame, m_sourceSampleRate);

	    offEv.data.note.note = noteOffs.begin()->pitch;

#ifdef DEBUG_AUDIO_GENERATOR
		std::cerr << "mixModel [note]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl;
#endif

	    plugin->sendEvent(eventTime, &offEv);
	    noteOffs.erase(noteOffs.begin());
	}
	
	plugin->run(blockTime);
	float **outs = plugin->getAudioOutputBuffers();

	for (size_t c = 0; c < m_targetChannelCount; ++c) {
#ifdef DEBUG_AUDIO_GENERATOR
	    std::cout << "mixModel [note]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
#endif

	    size_t sourceChannel = (c % plugin->getAudioOutputCount());

	    float channelGain = gain;
	    if (pan != 0.0) {
		if (c == 0) {
		    if (pan > 0.0) channelGain *= 1.0 - pan;
		} else {
		    if (pan < 0.0) channelGain *= pan + 1.0;
		}
	    }

	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
		buffer[c][i * m_pluginBlockSize + j] += 
		    channelGain * outs[sourceChannel][j];
	    }
	}
    }

    return got;
}