view audioio/AudioGenerator.cpp @ 88:51be0daa1386

Several changes related to referring to remote URLs for sessions and files:
* Pull file dialog wrapper functions out from MainWindow into FileFinder
* If a file referred to in a session is not found at its expected location,
  try a few other alternatives (same location as the session file or same
  location as the last audio file) before asking the user to locate it
* Allow user to give a URL when locating an audio file, not just locate on
  the filesystem
* Make wave file models remember the "original" location (e.g. URL) of
  the audio file, not just the actual location from which the data was
  loaded (e.g. local copy of that URL) -- when saving a session, use the
  original location so as not to refer to a temporary file
* Clean up incompletely-downloaded local copies of files

/* -*- 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)
{
    connect(PlayParameterRepository::getInstance(),
            SIGNAL(playPluginIdChanged(const Model *, QString)),
            this,
            SLOT(playPluginIdChanged(const Model *, QString)));

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

AudioGenerator::~AudioGenerator()
{
}

bool
AudioGenerator::canPlay(const Model *model)
{
    if (dynamic_cast<const DenseTimeValueModel *>(model) ||
	dynamic_cast<const SparseOneDimensionalModel *>(model) ||
	dynamic_cast<const NoteModel *>(model)) {
	return true;
    } else {
	return false;
    }
}

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 Model *model, QString)
{
    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 Model *model,
                                               QString configurationXml)
{
//    std::cerr << "AudioGenerator::playPluginConfigurationChanged" << std::endl;

    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);
    }
}

QString
AudioGenerator::getDefaultPlayPluginId(const Model *model)
{
    const SparseOneDimensionalModel *sodm =
        dynamic_cast<const SparseOneDimensionalModel *>(model);
    if (sodm) {
        return QString("dssi:%1:sample_player").
            arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
    }

    const NoteModel *nm = dynamic_cast<const NoteModel *>(model);
    if (nm) {
        return QString("dssi:%1:sample_player").
            arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
    }  
    
    return "";
}

QString
AudioGenerator::getDefaultPlayPluginConfiguration(const Model *model)
{
    QString program = "";

    const SparseOneDimensionalModel *sodm =
        dynamic_cast<const SparseOneDimensionalModel *>(model);
    if (sodm) {
        program = "tap";
    }

    const NoteModel *nm = dynamic_cast<const NoteModel *>(model);
    if (nm) {
        program = "piano";
    }

    if (program == "") return "";

    return
        QString("<plugin configuration=\"%1\" program=\"%2\"/>")
        .arg(XmlExportable::encodeEntities
             (QString("sampledir=%1")
              .arg(PluginXml::encodeConfigurationChars(getSampleDir()))))
        .arg(XmlExportable::encodeEntities(program));
}    

QString
AudioGenerator::getSampleDir()
{
    if (m_sampleDir != "") return m_sampleDir;

    try {
        m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples");
    } catch (DirectoryCreationFailed f) {
        std::cerr << "WARNING: AudioGenerator::getSampleDir: 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;
        }
    }

    return m_sampleDir;
}

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

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

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

    if (pluginId == "") {
        pluginId = getDefaultPlayPluginId(model);
        configurationXml = getDefaultPlayPluginConfiguration(model);
    }

    if (pluginId == "") return 0;

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

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

    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;
}

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);

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

    bool playing = !parameters->isPlayMuted();
    if (!playing) 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;

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

    if (channelBufSiz < totalFrames) {
	delete[] channelBuffer;
	channelBuffer = new float[totalFrames];
	channelBufSiz = totalFrames;
    }
    
    size_t got = 0;
    size_t prevChannel = 999;

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

	size_t sourceChannel = (c % dtvm->getChannelCount());

//	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;
	    }
	}

	if (prevChannel != sourceChannel) {
	    if (startFrame >= fadeIn/2) {
		got = dtvm->getValues
		    (sourceChannel,
		     startFrame - fadeIn/2, startFrame + frames + fadeOut/2,
		     channelBuffer);
	    } else {
		size_t missing = fadeIn/2 - startFrame;
		got = dtvm->getValues
		    (sourceChannel,
		     0, startFrame + frames + fadeOut/2,
		     channelBuffer + missing);
	    }	    
	}
	prevChannel = sourceChannel;

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

    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 = 127;

    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 = 127;

    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);
            }

	    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;
}