annotate audioio/AudioCallbackPlaySource.cpp @ 146:f90dcec6dd9a

* Permit viewing (though not editing) colour 3d plot layer data in
  the spreadsheet data viewer dialog
* Add somewhat simplistic RDF export for layers
* Fix display of peak frequencies in spectrum layer
* Fix (I hope) sizing of plugin parameter dialog

/* -*- 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 "AudioCallbackPlaySource.h"

#include "AudioGenerator.h"

#include "data/model/Model.h"
#include "base/ViewManagerBase.h"
#include "base/PlayParameterRepository.h"
#include "base/Preferences.h"
#include "data/model/DenseTimeValueModel.h"
#include "data/model/WaveFileModel.h"
#include "data/model/SparseOneDimensionalModel.h"
#include "plugin/RealTimePluginInstance.h"

#include "AudioCallbackPlayTarget.h"

#include <rubberband/RubberBandStretcher.h>
using namespace RubberBand;

#include <iostream>
#include <cassert>

//#define DEBUG_AUDIO_PLAY_SOURCE 1
//#define DEBUG_AUDIO_PLAY_SOURCE_PLAYING 1

const size_t AudioCallbackPlaySource::m_ringBufferSize = 131071;

AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManagerBase *manager,
                                                 QString clientName) :
    m_viewManager(manager),
    m_audioGenerator(new AudioGenerator()),
    m_clientName(clientName),
    m_readBuffers(0),
    m_writeBuffers(0),
    m_readBufferFill(0),
    m_writeBufferFill(0),
    m_bufferScavenger(1),
    m_sourceChannelCount(0),
    m_blockSize(1024),
    m_sourceSampleRate(0),
    m_targetSampleRate(0),
    m_playLatency(0),
    m_target(0),
    m_lastRetrievalTimestamp(0.0),
    m_lastRetrievedBlockSize(0),
    m_trustworthyTimestamps(true),
    m_lastCurrentFrame(0),
    m_playing(false),
    m_exiting(false),
    m_lastModelEndFrame(0),
    m_outputLeft(0.0),
    m_outputRight(0.0),
    m_auditioningPlugin(0),
    m_auditioningPluginBypassed(false),
    m_playStartFrame(0),
    m_playStartFramePassed(false),
    m_timeStretcher(0),
    m_monoStretcher(0),
    m_stretchRatio(1.0),
    m_stretcherInputCount(0),
    m_stretcherInputs(0),
    m_stretcherInputSizes(0),
    m_fillThread(0),
    m_converter(0),
    m_crapConverter(0),
    m_resampleQuality(Preferences::getInstance()->getResampleQuality())
{
    m_viewManager->setAudioPlaySource(this);

    connect(m_viewManager, SIGNAL(selectionChanged()),
	    this, SLOT(selectionChanged()));
    connect(m_viewManager, SIGNAL(playLoopModeChanged()),
	    this, SLOT(playLoopModeChanged()));
    connect(m_viewManager, SIGNAL(playSelectionModeChanged()),
	    this, SLOT(playSelectionModeChanged()));

    connect(PlayParameterRepository::getInstance(),
	    SIGNAL(playParametersChanged(PlayParameters *)),
	    this, SLOT(playParametersChanged(PlayParameters *)));

    connect(Preferences::getInstance(),
            SIGNAL(propertyChanged(PropertyContainer::PropertyName)),
            this, SLOT(preferenceChanged(PropertyContainer::PropertyName)));
}

AudioCallbackPlaySource::~AudioCallbackPlaySource()
{
    m_exiting = true;

    if (m_fillThread) {
	m_condition.wakeAll();
	m_fillThread->wait();
	delete m_fillThread;
    }

    clearModels();
    
    if (m_readBuffers != m_writeBuffers) {
	delete m_readBuffers;
    }

    delete m_writeBuffers;

    delete m_audioGenerator;

    for (size_t i = 0; i < m_stretcherInputCount; ++i) {
        delete[] m_stretcherInputs[i];
    }
    delete[] m_stretcherInputSizes;
    delete[] m_stretcherInputs;

    delete m_timeStretcher;
    delete m_monoStretcher;

    m_bufferScavenger.scavenge(true);
    m_pluginScavenger.scavenge(true);
}

void
AudioCallbackPlaySource::addModel(Model *model)
{
    if (m_models.find(model) != m_models.end()) return;

    bool canPlay = m_audioGenerator->addModel(model);

    m_mutex.lock();

    m_models.insert(model);
    if (model->getEndFrame() > m_lastModelEndFrame) {
	m_lastModelEndFrame = model->getEndFrame();
    }

    bool buffersChanged = false, srChanged = false;

    size_t modelChannels = 1;
    DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
    if (dtvm) modelChannels = dtvm->getChannelCount();
    if (modelChannels > m_sourceChannelCount) {
	m_sourceChannelCount = modelChannels;
    }

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << std::endl;
#endif

    if (m_sourceSampleRate == 0) {

	m_sourceSampleRate = model->getSampleRate();
	srChanged = true;

    } else if (model->getSampleRate() != m_sourceSampleRate) {

        // If this is a dense time-value model and we have no other, we
        // can just switch to this model's sample rate

        if (dtvm) {

            bool conflicting = false;

            for (std::set<Model *>::const_iterator i = m_models.begin();
                 i != m_models.end(); ++i) {
                // Only wave file models can be considered conflicting --
                // writable wave file models are derived and we shouldn't
                // take their rates into account.  Also, don't give any
                // particular weight to a file that's already playing at
                // the wrong rate anyway
                WaveFileModel *wfm = dynamic_cast<WaveFileModel *>(*i);
                if (wfm && wfm != dtvm &&
                    wfm->getSampleRate() != model->getSampleRate() &&
                    wfm->getSampleRate() == m_sourceSampleRate) {
                    std::cerr << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << std::endl;
                    conflicting = true;
                    break;
                }
            }

            if (conflicting) {

                std::cerr << "AudioCallbackPlaySource::addModel: ERROR: "
                          << "New model sample rate does not match" << std::endl
                          << "existing model(s) (new " << model->getSampleRate()
                          << " vs " << m_sourceSampleRate
                          << "), playback will be wrong"
                          << std::endl;
                
                emit sampleRateMismatch(model->getSampleRate(),
                                        m_sourceSampleRate,
                                        false);
            } else {
                m_sourceSampleRate = model->getSampleRate();
                srChanged = true;
            }
        }
    }

    if (!m_writeBuffers || (m_writeBuffers->size() < getTargetChannelCount())) {
	clearRingBuffers(true, getTargetChannelCount());
	buffersChanged = true;
    } else {
	if (canPlay) clearRingBuffers(true);
    }

    if (buffersChanged || srChanged) {
	if (m_converter) {
	    src_delete(m_converter);
            src_delete(m_crapConverter);
	    m_converter = 0;
            m_crapConverter = 0;
	}
    }

    m_mutex.unlock();

    m_audioGenerator->setTargetChannelCount(getTargetChannelCount());

    if (!m_fillThread) {
	m_fillThread = new FillThread(*this);
	m_fillThread->start();
    }

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << std::endl;
#endif

    if (buffersChanged || srChanged) {
	emit modelReplaced();
    }

    connect(model, SIGNAL(modelChanged(size_t, size_t)),
            this, SLOT(modelChanged(size_t, size_t)));

    m_condition.wakeAll();
}

void
AudioCallbackPlaySource::modelChanged(size_t startFrame, size_t endFrame)
{
#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cerr << "AudioCallbackPlaySource::modelChanged(" << startFrame << "," << endFrame << ")" << std::endl;
#endif
    if (endFrame > m_lastModelEndFrame) {
        m_lastModelEndFrame = endFrame;
        rebuildRangeLists();
    }
}

void
AudioCallbackPlaySource::removeModel(Model *model)
{
    m_mutex.lock();

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << std::endl;
#endif

    disconnect(model, SIGNAL(modelChanged(size_t, size_t)),
               this, SLOT(modelChanged(size_t, size_t)));

    m_models.erase(model);

    if (m_models.empty()) {
	if (m_converter) {
	    src_delete(m_converter);
            src_delete(m_crapConverter);
	    m_converter = 0;
            m_crapConverter = 0;
	}
	m_sourceSampleRate = 0;
    }

    size_t lastEnd = 0;
    for (std::set<Model *>::const_iterator i = m_models.begin();
	 i != m_models.end(); ++i) {
//	std::cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << std::endl;
	if ((*i)->getEndFrame() > lastEnd) lastEnd = (*i)->getEndFrame();
//	std::cout << "(done, lastEnd now " << lastEnd << ")" << std::endl;
    }
    m_lastModelEndFrame = lastEnd;

    m_mutex.unlock();

    m_audioGenerator->removeModel(model);

    clearRingBuffers();
}

void
AudioCallbackPlaySource::clearModels()
{
    m_mutex.lock();

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "AudioCallbackPlaySource::clearModels()" << std::endl;
#endif

    m_models.clear();

    if (m_converter) {
	src_delete(m_converter);
        src_delete(m_crapConverter);
	m_converter = 0;
        m_crapConverter = 0;
    }

    m_lastModelEndFrame = 0;

    m_sourceSampleRate = 0;

    m_mutex.unlock();

    m_audioGenerator->clearModels();

    clearRingBuffers();
}    

void
AudioCallbackPlaySource::clearRingBuffers(bool haveLock, size_t count)
{
    if (!haveLock) m_mutex.lock();

    rebuildRangeLists();

    if (count == 0) {
	if (m_writeBuffers) count = m_writeBuffers->size();
    }

    m_writeBufferFill = getCurrentBufferedFrame();

    if (m_readBuffers != m_writeBuffers) {
	delete m_writeBuffers;
    }

    m_writeBuffers = new RingBufferVector;

    for (size_t i = 0; i < count; ++i) {
	m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize));
    }

//    std::cout << "AudioCallbackPlaySource::clearRingBuffers: Created "
//	      << count << " write buffers" << std::endl;

    if (!haveLock) {
	m_mutex.unlock();
    }
}

void
AudioCallbackPlaySource::play(size_t startFrame)
{
    if (m_viewManager->getPlaySelectionMode() &&
	!m_viewManager->getSelections().empty()) {

        std::cerr << "AudioCallbackPlaySource::play: constraining frame " << startFrame << " to selection = ";

        startFrame = m_viewManager->constrainFrameToSelection(startFrame);

        std::cerr << startFrame << std::endl;

    } else {
	if (startFrame >= m_lastModelEndFrame) {
	    startFrame = 0;
	}
    }

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cerr << "play(" << startFrame << ") -> playback model ";
#endif

    startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame);

    std::cerr << startFrame << std::endl;

    // The fill thread will automatically empty its buffers before
    // starting again if we have not so far been playing, but not if
    // we're just re-seeking.
    // NO -- we can end up playing some first -- always reset here

    m_mutex.lock();

    if (m_timeStretcher) {
        m_timeStretcher->reset();
    }
    if (m_monoStretcher) {
        m_monoStretcher->reset();
    }

    m_readBufferFill = m_writeBufferFill = startFrame;
    if (m_readBuffers) {
        for (size_t c = 0; c < getTargetChannelCount(); ++c) {
            RingBuffer<float> *rb = getReadRingBuffer(c);
#ifdef DEBUG_AUDIO_PLAY_SOURCE
            std::cerr << "reset ring buffer for channel " << c << std::endl;
#endif
            if (rb) rb->reset();
        }
    }
    if (m_converter) src_reset(m_converter);
    if (m_crapConverter) src_reset(m_crapConverter);

    m_mutex.unlock();

    m_audioGenerator->reset();

    m_playStartFrame = startFrame;
    m_playStartFramePassed = false;
    m_playStartedAt = RealTime::zeroTime;
    if (m_target) {
        m_playStartedAt = RealTime::fromSeconds(m_target->getCurrentTime());
    }

    bool changed = !m_playing;
    m_lastRetrievalTimestamp = 0;
    m_lastCurrentFrame = 0;
    m_playing = true;
    m_condition.wakeAll();
    if (changed) emit playStatusChanged(m_playing);
}

void
AudioCallbackPlaySource::stop()
{
    bool changed = m_playing;
    m_playing = false;
    m_condition.wakeAll();
    m_lastRetrievalTimestamp = 0;
    m_lastCurrentFrame = 0;
    if (changed) emit playStatusChanged(m_playing);
}

void
AudioCallbackPlaySource::selectionChanged()
{
    if (m_viewManager->getPlaySelectionMode()) {
	clearRingBuffers();
    }
}

void
AudioCallbackPlaySource::playLoopModeChanged()
{
    clearRingBuffers();
}

void
AudioCallbackPlaySource::playSelectionModeChanged()
{
    if (!m_viewManager->getSelections().empty()) {
	clearRingBuffers();
    }
}

void
AudioCallbackPlaySource::playParametersChanged(PlayParameters *)
{
    clearRingBuffers();
}

void
AudioCallbackPlaySource::preferenceChanged(PropertyContainer::PropertyName n)
{
    if (n == "Resample Quality") {
        setResampleQuality(Preferences::getInstance()->getResampleQuality());
    }
}

void
AudioCallbackPlaySource::audioProcessingOverload()
{
    std::cerr << "Audio processing overload!" << std::endl;

    if (!m_playing) return;

    RealTimePluginInstance *ap = m_auditioningPlugin;
    if (ap && !m_auditioningPluginBypassed) {
        m_auditioningPluginBypassed = true;
        emit audioOverloadPluginDisabled();
        return;
    }

    if (m_timeStretcher &&
        m_timeStretcher->getTimeRatio() < 1.0 &&
        m_stretcherInputCount > 1 &&
        m_monoStretcher && !m_stretchMono) {
        m_stretchMono = true;
        emit audioTimeStretchMultiChannelDisabled();
        return;
    }
}

void
AudioCallbackPlaySource::setTarget(AudioCallbackPlayTarget *target, size_t size)
{
    m_target = target;
//    std::cout << "AudioCallbackPlaySource::setTargetBlockSize() -> " << size << std::endl;
    assert(size < m_ringBufferSize);
    m_blockSize = size;
}

size_t
AudioCallbackPlaySource::getTargetBlockSize() const
{
//    std::cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << std::endl;
    return m_blockSize;
}

void
AudioCallbackPlaySource::setTargetPlayLatency(size_t latency)
{
    m_playLatency = latency;
}

size_t
AudioCallbackPlaySource::getTargetPlayLatency() const
{
    return m_playLatency;
}

size_t
AudioCallbackPlaySource::getCurrentPlayingFrame()
{
    // This method attempts to estimate which audio sample frame is
    // "currently coming through the speakers".

    size_t targetRate = getTargetSampleRate();
    size_t latency = m_playLatency; // at target rate
    RealTime latency_t = RealTime::frame2RealTime(latency, targetRate);

    return getCurrentFrame(latency_t);
}

size_t
AudioCallbackPlaySource::getCurrentBufferedFrame()
{
    return getCurrentFrame(RealTime::zeroTime);
}

size_t
AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t)
{
    bool resample = false;
    double resampleRatio = 1.0;

    // We resample when filling the ring buffer, and time-stretch when
    // draining it.  The buffer contains data at the "target rate" and
    // the latency provided by the target is also at the target rate.
    // Because of the multiple rates involved, we do the actual
    // calculation using RealTime instead.

    size_t sourceRate = getSourceSampleRate();
    size_t targetRate = getTargetSampleRate();

    if (sourceRate == 0 || targetRate == 0) return 0;

    size_t inbuffer = 0; // at target rate

    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
	RingBuffer<float> *rb = getReadRingBuffer(c);
	if (rb) {
	    size_t here = rb->getReadSpace();
	    if (c == 0 || here < inbuffer) inbuffer = here;
	}
    }

    size_t readBufferFill = m_readBufferFill;
    size_t lastRetrievedBlockSize = m_lastRetrievedBlockSize;
    double lastRetrievalTimestamp = m_lastRetrievalTimestamp;
    double currentTime = 0.0;
    if (m_target) currentTime = m_target->getCurrentTime();

    bool looping = m_viewManager->getPlayLoopMode();

    RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, targetRate);

    size_t stretchlat = 0;
    double timeRatio = 1.0;

    if (m_timeStretcher) {
        stretchlat = m_timeStretcher->getLatency();
        timeRatio = m_timeStretcher->getTimeRatio();
    }

    RealTime stretchlat_t = RealTime::frame2RealTime(stretchlat, targetRate);

    // When the target has just requested a block from us, the last
    // sample it obtained was our buffer fill frame count minus the
    // amount of read space (converted back to source sample rate)
    // remaining now.  That sample is not expected to be played until
    // the target's play latency has elapsed.  By the time the
    // following block is requested, that sample will be at the
    // target's play latency minus the last requested block size away
    // from being played.

    RealTime sincerequest_t = RealTime::zeroTime;
    RealTime lastretrieved_t = RealTime::zeroTime;

    if (m_target &&
        m_trustworthyTimestamps &&
        lastRetrievalTimestamp != 0.0) {

        lastretrieved_t = RealTime::frame2RealTime
            (lastRetrievedBlockSize, targetRate);

        // calculate number of frames at target rate that have elapsed
        // since the end of the last call to getSourceSamples

        if (m_trustworthyTimestamps && !looping) {

            // this adjustment seems to cause more problems when looping
            double elapsed = currentTime - lastRetrievalTimestamp;

            if (elapsed > 0.0) {
                sincerequest_t = RealTime::fromSeconds(elapsed);
            }
        }

    } else {

        lastretrieved_t = RealTime::frame2RealTime
            (getTargetBlockSize(), targetRate);
    }

    RealTime bufferedto_t = RealTime::frame2RealTime(readBufferFill, sourceRate);

    if (timeRatio != 1.0) {
        lastretrieved_t = lastretrieved_t / timeRatio;
        sincerequest_t = sincerequest_t / timeRatio;
    }

#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
    std::cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n  stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n  since request: " << sincerequest_t << ", last retrieved: " << lastretrieved_t << std::endl;
#endif

    RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate);

    // Normally the range lists should contain at least one item each
    // -- if playback is unconstrained, that item should report the
    // entire source audio duration.

    if (m_rangeStarts.empty()) {
        rebuildRangeLists();
    }

    if (m_rangeStarts.empty()) {
        // this code is only used in case of error in rebuildRangeLists
        RealTime playing_t = bufferedto_t
            - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t
            + sincerequest_t;
        size_t frame = RealTime::realTime2Frame(playing_t, sourceRate);
        return m_viewManager->alignPlaybackFrameToReference(frame);
    }

    int inRange = 0;
    int index = 0;

    for (size_t i = 0; i < m_rangeStarts.size(); ++i) {
        if (bufferedto_t >= m_rangeStarts[i]) {
            inRange = index;
        } else {
            break;
        }
        ++index;
    }

    if (inRange >= m_rangeStarts.size()) inRange = m_rangeStarts.size()-1;

    RealTime playing_t = bufferedto_t;

    playing_t = playing_t
        - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t
        + sincerequest_t;

    // This rather gross little hack is used to ensure that latency
    // compensation doesn't result in the playback pointer appearing
    // to start earlier than the actual playback does.  It doesn't
    // work properly (hence the bail-out in the middle) because if we
    // are playing a relatively short looped region, the playing time
    // estimated from the buffer fill frame may have wrapped around
    // the region boundary and end up being much smaller than the
    // theoretical play start frame, perhaps even for the entire
    // duration of playback!

    if (!m_playStartFramePassed) {
        RealTime playstart_t = RealTime::frame2RealTime(m_playStartFrame,
                                                        sourceRate);
        if (playing_t < playstart_t) {
//            std::cerr << "playing_t " << playing_t << " < playstart_t " 
//                      << playstart_t << std::endl;
            if (/*!!! sincerequest_t > RealTime::zeroTime && */
                m_playStartedAt + latency_t + stretchlat_t <
                RealTime::fromSeconds(currentTime)) {
                std::cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << std::endl;
                m_playStartFramePassed = true;
            } else {
                playing_t = playstart_t;
            }
        } else {
            m_playStartFramePassed = true;
        }
    }

    playing_t = playing_t - m_rangeStarts[inRange];
 
#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
    std::cerr << "playing_t as offset into range " << inRange << " (with start = " << m_rangeStarts[inRange] << ") = " << playing_t << std::endl;
#endif

    while (playing_t < RealTime::zeroTime) {

        if (inRange == 0) {
            if (looping) {
                inRange = m_rangeStarts.size() - 1;
            } else {
                break;
            }
        } else {
            --inRange;
        }

        playing_t = playing_t + m_rangeDurations[inRange];
    }

    playing_t = playing_t + m_rangeStarts[inRange];

#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
    std::cerr << "  playing time: " << playing_t << std::endl;
#endif

    if (!looping) {
        if (inRange == m_rangeStarts.size()-1 &&
            playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) {
std::cerr << "Not looping, inRange " << inRange << " == rangeStarts.size()-1, playing_t " << playing_t << " >= m_rangeStarts[inRange] " << m_rangeStarts[inRange] << " + m_rangeDurations[inRange] " << m_rangeDurations[inRange] << " -- stopping" << std::endl;
            stop();
        }
    }

    if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;

    size_t frame = RealTime::realTime2Frame(playing_t, sourceRate);

    if (m_lastCurrentFrame > 0 && !looping) {
        if (frame < m_lastCurrentFrame) {
            frame = m_lastCurrentFrame;
        }
    }

    m_lastCurrentFrame = frame;

    return m_viewManager->alignPlaybackFrameToReference(frame);
}

void
AudioCallbackPlaySource::rebuildRangeLists()
{
    bool constrained = (m_viewManager->getPlaySelectionMode());

    m_rangeStarts.clear();
    m_rangeDurations.clear();

    size_t sourceRate = getSourceSampleRate();
    if (sourceRate == 0) return;

    RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate);
    if (end == RealTime::zeroTime) return;

    if (!constrained) {
        m_rangeStarts.push_back(RealTime::zeroTime);
        m_rangeDurations.push_back(end);
        return;
    }

    MultiSelection::SelectionList selections = m_viewManager->getSelections();
    MultiSelection::SelectionList::const_iterator i;

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cerr << "AudioCallbackPlaySource::rebuildRangeLists" << std::endl;
#endif

    if (!selections.empty()) {

        for (i = selections.begin(); i != selections.end(); ++i) {
            
            RealTime start =
                (RealTime::frame2RealTime
                 (m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()),
                  sourceRate));
            RealTime duration = 
                (RealTime::frame2RealTime
                 (m_viewManager->alignReferenceToPlaybackFrame(i->getEndFrame()) -
                  m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()),
                  sourceRate));
            
            m_rangeStarts.push_back(start);
            m_rangeDurations.push_back(duration);
        }
    } else {
        m_rangeStarts.push_back(RealTime::zeroTime);
        m_rangeDurations.push_back(end);
    }

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cerr << "Now have " << m_rangeStarts.size() << " play ranges" << std::endl;
#endif
}

void
AudioCallbackPlaySource::setOutputLevels(float left, float right)
{
    m_outputLeft = left;
    m_outputRight = right;
}

bool
AudioCallbackPlaySource::getOutputLevels(float &left, float &right)
{
    left = m_outputLeft;
    right = m_outputRight;
    return true;
}

void
AudioCallbackPlaySource::setTargetSampleRate(size_t sr)
{
    m_targetSampleRate = sr;
    initialiseConverter();
}

void
AudioCallbackPlaySource::initialiseConverter()
{
    m_mutex.lock();

    if (m_converter) {
        src_delete(m_converter);
        src_delete(m_crapConverter);
        m_converter = 0;
        m_crapConverter = 0;
    }

    if (getSourceSampleRate() != getTargetSampleRate()) {

	int err = 0;

	m_converter = src_new(m_resampleQuality == 2 ? SRC_SINC_BEST_QUALITY :
                              m_resampleQuality == 1 ? SRC_SINC_MEDIUM_QUALITY :
                              m_resampleQuality == 0 ? SRC_SINC_FASTEST :
                                                       SRC_SINC_MEDIUM_QUALITY,
			      getTargetChannelCount(), &err);

        if (m_converter) {
            m_crapConverter = src_new(SRC_LINEAR,
                                      getTargetChannelCount(),
                                      &err);
        }

	if (!m_converter || !m_crapConverter) {
	    std::cerr
		<< "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: "
		<< src_strerror(err) << std::endl;

            if (m_converter) {
                src_delete(m_converter);
                m_converter = 0;
            } 

            if (m_crapConverter) {
                src_delete(m_crapConverter);
                m_crapConverter = 0;
            }

            m_mutex.unlock();

            emit sampleRateMismatch(getSourceSampleRate(),
                                    getTargetSampleRate(),
                                    false);
	} else {

            m_mutex.unlock();

            emit sampleRateMismatch(getSourceSampleRate(),
                                    getTargetSampleRate(),
                                    true);
        }
    } else {
        m_mutex.unlock();
    }
}

void
AudioCallbackPlaySource::setResampleQuality(int q)
{
    if (q == m_resampleQuality) return;
    m_resampleQuality = q;

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cerr << "AudioCallbackPlaySource::setResampleQuality: setting to "
              << m_resampleQuality << std::endl;
#endif

    initialiseConverter();
}

void
AudioCallbackPlaySource::setAuditioningEffect(Auditionable *a)
{
    RealTimePluginInstance *plugin = dynamic_cast<RealTimePluginInstance *>(a);
    if (a && !plugin) {
        std::cerr << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << std::endl;
    }
    RealTimePluginInstance *formerPlugin = m_auditioningPlugin;
    m_auditioningPlugin = plugin;
    m_auditioningPluginBypassed = false;
    if (formerPlugin) m_pluginScavenger.claim(formerPlugin);
}

void
AudioCallbackPlaySource::setSoloModelSet(std::set<Model *> s)
{
    m_audioGenerator->setSoloModelSet(s);
    clearRingBuffers();
}

void
AudioCallbackPlaySource::clearSoloModelSet()
{
    m_audioGenerator->clearSoloModelSet();
    clearRingBuffers();
}

size_t
AudioCallbackPlaySource::getTargetSampleRate() const
{
    if (m_targetSampleRate) return m_targetSampleRate;
    else return getSourceSampleRate();
}

size_t
AudioCallbackPlaySource::getSourceChannelCount() const
{
    return m_sourceChannelCount;
}

size_t
AudioCallbackPlaySource::getTargetChannelCount() const
{
    if (m_sourceChannelCount < 2) return 2;
    return m_sourceChannelCount;
}

size_t
AudioCallbackPlaySource::getSourceSampleRate() const
{
    return m_sourceSampleRate;
}

void
AudioCallbackPlaySource::setTimeStretch(float factor)
{
    m_stretchRatio = factor;

    if (m_timeStretcher || (factor == 1.f)) {
        // stretch ratio will be set in next process call if appropriate
        return;
    } else {
        m_stretcherInputCount = getTargetChannelCount();
        RubberBandStretcher *stretcher = new RubberBandStretcher
            (getTargetSampleRate(),
             m_stretcherInputCount,
             RubberBandStretcher::OptionProcessRealTime,
             factor);
        RubberBandStretcher *monoStretcher = new RubberBandStretcher
            (getTargetSampleRate(),
             1,
             RubberBandStretcher::OptionProcessRealTime,
             factor);
        m_stretcherInputs = new float *[m_stretcherInputCount];
        m_stretcherInputSizes = new size_t[m_stretcherInputCount];
        for (size_t c = 0; c < m_stretcherInputCount; ++c) {
            m_stretcherInputSizes[c] = 16384;
            m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
        }
        m_monoStretcher = monoStretcher;
        m_timeStretcher = stretcher;
        return;
    }
}

size_t
AudioCallbackPlaySource::getSourceSamples(size_t ucount, float **buffer)
{
    int count = ucount;

    if (!m_playing) {
	for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
	    for (int i = 0; i < count; ++i) {
		buffer[ch][i] = 0.0;
	    }
	}
	return 0;
    }

    // Ensure that all buffers have at least the amount of data we
    // need -- else reduce the size of our requests correspondingly

    for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {

        RingBuffer<float> *rb = getReadRingBuffer(ch);
        
        if (!rb) {
            std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
                      << "No ring buffer available for channel " << ch
                      << ", returning no data here" << std::endl;
            count = 0;
            break;
        }

        size_t rs = rb->getReadSpace();
        if (rs < count) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
            std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
                      << "Ring buffer for channel " << ch << " has only "
                      << rs << " (of " << count << ") samples available, "
                      << "reducing request size" << std::endl;
#endif
            count = rs;
        }
    }

    if (count == 0) return 0;

    RubberBandStretcher *ts = m_timeStretcher;
    RubberBandStretcher *ms = m_monoStretcher;

    float ratio = ts ? ts->getTimeRatio() : 1.f;

    if (ratio != m_stretchRatio) {
        if (!ts) {
            std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << std::endl;
            m_stretchRatio = 1.f;
        } else {
            ts->setTimeRatio(m_stretchRatio);
            if (ms) ms->setTimeRatio(m_stretchRatio);
            if (m_stretchRatio >= 1.0) m_stretchMono = false;
        }
    }

    int stretchChannels = m_stretcherInputCount;
    if (m_stretchMono) {
        if (ms) {
            ts = ms;
            stretchChannels = 1;
        } else {
            m_stretchMono = false;
        }
    }

    if (m_target) {
        m_lastRetrievedBlockSize = count;
        m_lastRetrievalTimestamp = m_target->getCurrentTime();
    }

    if (!ts || ratio == 1.f) {

	int got = 0;

	for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {

	    RingBuffer<float> *rb = getReadRingBuffer(ch);

	    if (rb) {

		// this is marginally more likely to leave our channels in
		// sync after a processing failure than just passing "count":
		size_t request = count;
		if (ch > 0) request = got;

		got = rb->read(buffer[ch], request);
	    
#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
		std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << std::endl;
#endif
	    }

	    for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
		for (int i = got; i < count; ++i) {
		    buffer[ch][i] = 0.0;
		}
	    }
	}

        applyAuditioningEffect(count, buffer);

        m_condition.wakeAll();

	return got;
    }

    size_t channels = getTargetChannelCount();
    size_t available;
    int warned = 0;
    size_t fedToStretcher = 0;

    // The input block for a given output is approx output / ratio,
    // but we can't predict it exactly, for an adaptive timestretcher.

    while ((available = ts->available()) < count) {

        size_t reqd = lrintf((count - available) / ratio);
        reqd = std::max(reqd, ts->getSamplesRequired());
        if (reqd == 0) reqd = 1;
                
        size_t got = reqd;

#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
        std::cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << std::endl;
#endif

        for (size_t c = 0; c < channels; ++c) {
            if (c >= m_stretcherInputCount) continue;
            if (reqd > m_stretcherInputSizes[c]) {
                if (c == 0) {
                    std::cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << std::endl;
                }
                delete[] m_stretcherInputs[c];
                m_stretcherInputSizes[c] = reqd * 2;
                m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
            }
        }

        for (size_t c = 0; c < channels; ++c) {
            if (c >= m_stretcherInputCount) continue;
            RingBuffer<float> *rb = getReadRingBuffer(c);
            if (rb) {
                size_t gotHere;
                if (stretchChannels == 1 && c > 0) {
                    gotHere = rb->readAdding(m_stretcherInputs[0], got);
                } else {
                    gotHere = rb->read(m_stretcherInputs[c], got);
                }
                if (gotHere < got) got = gotHere;
                
#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
                if (c == 0) {
                    std::cerr << "feeding stretcher: got " << gotHere
                              << ", " << rb->getReadSpace() << " remain" << std::endl;
                }
#endif
                
            } else {
                std::cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << std::endl;
            }
        }

        if (got < reqd) {
            std::cerr << "WARNING: Read underrun in playback ("
                      << got << " < " << reqd << ")" << std::endl;
        }

        ts->process(m_stretcherInputs, got, false);

        fedToStretcher += got;

        if (got == 0) break;

        if (ts->available() == available) {
            std::cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << std::endl;
            if (++warned == 5) break;
        }
    }

    ts->retrieve(buffer, count);

    for (int c = stretchChannels; c < getTargetChannelCount(); ++c) {
        for (int i = 0; i < count; ++i) {
            buffer[c][i] = buffer[0][i];
        }
    }

    applyAuditioningEffect(count, buffer);

    m_condition.wakeAll();

    return count;
}

void
AudioCallbackPlaySource::applyAuditioningEffect(size_t count, float **buffers)
{
    if (m_auditioningPluginBypassed) return;
    RealTimePluginInstance *plugin = m_auditioningPlugin;
    if (!plugin) return;

    if (plugin->getAudioInputCount() != getTargetChannelCount()) {
//        std::cerr << "plugin input count " << plugin->getAudioInputCount() 
//                  << " != our channel count " << getTargetChannelCount()
//                  << std::endl;
        return;
    }
    if (plugin->getAudioOutputCount() != getTargetChannelCount()) {
//        std::cerr << "plugin output count " << plugin->getAudioOutputCount() 
//                  << " != our channel count " << getTargetChannelCount()
//                  << std::endl;
        return;
    }
    if (plugin->getBufferSize() < count) {
//        std::cerr << "plugin buffer size " << plugin->getBufferSize() 
//                  << " < our block size " << count
//                  << std::endl;
        return;
    }

    float **ib = plugin->getAudioInputBuffers();
    float **ob = plugin->getAudioOutputBuffers();

    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
        for (size_t i = 0; i < count; ++i) {
            ib[c][i] = buffers[c][i];
        }
    }

    plugin->run(Vamp::RealTime::zeroTime, count);
    
    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
        for (size_t i = 0; i < count; ++i) {
            buffers[c][i] = ob[c][i];
        }
    }
}    

// Called from fill thread, m_playing true, mutex held
bool
AudioCallbackPlaySource::fillBuffers()
{
    static float *tmp = 0;
    static size_t tmpSize = 0;

    size_t space = 0;
    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
	RingBuffer<float> *wb = getWriteRingBuffer(c);
	if (wb) {
	    size_t spaceHere = wb->getWriteSpace();
	    if (c == 0 || spaceHere < space) space = spaceHere;
	}
    }
    
    if (space == 0) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
        std::cout << "AudioCallbackPlaySourceFillThread: no space to fill" << std::endl;
#endif
        return false;
    }

    size_t f = m_writeBufferFill;
	
    bool readWriteEqual = (m_readBuffers == m_writeBuffers);

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << std::endl;
#endif

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "buffered to " << f << " already" << std::endl;
#endif

    bool resample = (getSourceSampleRate() != getTargetSampleRate());

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << std::endl;
#endif

    size_t channels = getTargetChannelCount();

    size_t orig = space;
    size_t got = 0;

    static float **bufferPtrs = 0;
    static size_t bufferPtrCount = 0;

    if (bufferPtrCount < channels) {
	if (bufferPtrs) delete[] bufferPtrs;
	bufferPtrs = new float *[channels];
	bufferPtrCount = channels;
    }

    size_t generatorBlockSize = m_audioGenerator->getBlockSize();

    if (resample && !m_converter) {
	static bool warned = false;
	if (!warned) {
	    std::cerr << "WARNING: sample rates differ, but no converter available!" << std::endl;
	    warned = true;
	}
    }

    if (resample && m_converter) {

	double ratio =
	    double(getTargetSampleRate()) / double(getSourceSampleRate());
	orig = size_t(orig / ratio + 0.1);

	// orig must be a multiple of generatorBlockSize
	orig = (orig / generatorBlockSize) * generatorBlockSize;
	if (orig == 0) return false;

	size_t work = std::max(orig, space);

	// We only allocate one buffer, but we use it in two halves.
	// We place the non-interleaved values in the second half of
	// the buffer (orig samples for channel 0, orig samples for
	// channel 1 etc), and then interleave them into the first
	// half of the buffer.  Then we resample back into the second
	// half (interleaved) and de-interleave the results back to
	// the start of the buffer for insertion into the ringbuffers.
	// What a faff -- especially as we've already de-interleaved
	// the audio data from the source file elsewhere before we
	// even reach this point.
	
	if (tmpSize < channels * work * 2) {
	    delete[] tmp;
	    tmp = new float[channels * work * 2];
	    tmpSize = channels * work * 2;
	}

	float *nonintlv = tmp + channels * work;
	float *intlv = tmp;
	float *srcout = tmp + channels * work;
	
	for (size_t c = 0; c < channels; ++c) {
	    for (size_t i = 0; i < orig; ++i) {
		nonintlv[channels * i + c] = 0.0f;
	    }
	}

	for (size_t c = 0; c < channels; ++c) {
	    bufferPtrs[c] = nonintlv + c * orig;
	}

	got = mixModels(f, orig, bufferPtrs);

	// and interleave into first half
	for (size_t c = 0; c < channels; ++c) {
	    for (size_t i = 0; i < got; ++i) {
		float sample = nonintlv[c * got + i];
		intlv[channels * i + c] = sample;
	    }
	}
		
	SRC_DATA data;
	data.data_in = intlv;
	data.data_out = srcout;
	data.input_frames = got;
	data.output_frames = work;
	data.src_ratio = ratio;
	data.end_of_input = 0;
	
	int err = 0;

        if (m_timeStretcher && m_timeStretcher->getTimeRatio() < 0.4) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
            std::cout << "Using crappy converter" << std::endl;
#endif
            err = src_process(m_crapConverter, &data);
        } else {
            err = src_process(m_converter, &data);
        }

	size_t toCopy = size_t(got * ratio + 0.1);

	if (err) {
	    std::cerr
		<< "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: "
		<< src_strerror(err) << std::endl;
	    //!!! Then what?
	} else {
	    got = data.input_frames_used;
	    toCopy = data.output_frames_gen;
#ifdef DEBUG_AUDIO_PLAY_SOURCE
	    std::cout << "Resampled " << got << " frames to " << toCopy << " frames" << std::endl;
#endif
	}
	
	for (size_t c = 0; c < channels; ++c) {
	    for (size_t i = 0; i < toCopy; ++i) {
		tmp[i] = srcout[channels * i + c];
	    }
	    RingBuffer<float> *wb = getWriteRingBuffer(c);
	    if (wb) wb->write(tmp, toCopy);
	}

	m_writeBufferFill = f;
	if (readWriteEqual) m_readBufferFill = f;

    } else {

	// space must be a multiple of generatorBlockSize
	space = (space / generatorBlockSize) * generatorBlockSize;
	if (space == 0) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
            std::cout << "requested fill is less than generator block size of "
                      << generatorBlockSize << ", leaving it" << std::endl;
#endif
            return false;
        }

	if (tmpSize < channels * space) {
	    delete[] tmp;
	    tmp = new float[channels * space];
	    tmpSize = channels * space;
	}

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

	    bufferPtrs[c] = tmp + c * space;
	    
	    for (size_t i = 0; i < space; ++i) {
		tmp[c * space + i] = 0.0f;
	    }
	}

	size_t got = mixModels(f, space, bufferPtrs);

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

	    RingBuffer<float> *wb = getWriteRingBuffer(c);
	    if (wb) {
                size_t actual = wb->write(bufferPtrs[c], got);
#ifdef DEBUG_AUDIO_PLAY_SOURCE
		std::cout << "Wrote " << actual << " samples for ch " << c << ", now "
			  << wb->getReadSpace() << " to read" 
			  << std::endl;
#endif
                if (actual < got) {
                    std::cerr << "WARNING: Buffer overrun in channel " << c
                              << ": wrote " << actual << " of " << got
                              << " samples" << std::endl;
                }
            }
	}

	m_writeBufferFill = f;
	if (readWriteEqual) m_readBufferFill = f;

	//!!! how do we know when ended? need to mark up a fully-buffered flag and check this if we find the buffers empty in getSourceSamples
    }

    return true;
}    

size_t
AudioCallbackPlaySource::mixModels(size_t &frame, size_t count, float **buffers)
{
    size_t processed = 0;
    size_t chunkStart = frame;
    size_t chunkSize = count;
    size_t selectionSize = 0;
    size_t nextChunkStart = chunkStart + chunkSize;
    
    bool looping = m_viewManager->getPlayLoopMode();
    bool constrained = (m_viewManager->getPlaySelectionMode() &&
			!m_viewManager->getSelections().empty());

    static float **chunkBufferPtrs = 0;
    static size_t chunkBufferPtrCount = 0;
    size_t channels = getTargetChannelCount();

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << std::endl;
#endif

    if (chunkBufferPtrCount < channels) {
	if (chunkBufferPtrs) delete[] chunkBufferPtrs;
	chunkBufferPtrs = new float *[channels];
	chunkBufferPtrCount = channels;
    }

    for (size_t c = 0; c < channels; ++c) {
	chunkBufferPtrs[c] = buffers[c];
    }

    while (processed < count) {
	
	chunkSize = count - processed;
	nextChunkStart = chunkStart + chunkSize;
	selectionSize = 0;

	size_t fadeIn = 0, fadeOut = 0;

	if (constrained) {

            size_t rChunkStart =
                m_viewManager->alignPlaybackFrameToReference(chunkStart);
	    
	    Selection selection =
		m_viewManager->getContainingSelection(rChunkStart, true);
	    
	    if (selection.isEmpty()) {
		if (looping) {
		    selection = *m_viewManager->getSelections().begin();
		    chunkStart = m_viewManager->alignReferenceToPlaybackFrame
                        (selection.getStartFrame());
		    fadeIn = 50;
		}
	    }

	    if (selection.isEmpty()) {

		chunkSize = 0;
		nextChunkStart = chunkStart;

	    } else {

                size_t sf = m_viewManager->alignReferenceToPlaybackFrame
                    (selection.getStartFrame());
                size_t ef = m_viewManager->alignReferenceToPlaybackFrame
                    (selection.getEndFrame());

		selectionSize = ef - sf;

		if (chunkStart < sf) {
		    chunkStart = sf;
		    fadeIn = 50;
		}

		nextChunkStart = chunkStart + chunkSize;

		if (nextChunkStart >= ef) {
		    nextChunkStart = ef;
		    fadeOut = 50;
		}

		chunkSize = nextChunkStart - chunkStart;
	    }
	
	} else if (looping && m_lastModelEndFrame > 0) {

	    if (chunkStart >= m_lastModelEndFrame) {
		chunkStart = 0;
	    }
	    if (chunkSize > m_lastModelEndFrame - chunkStart) {
		chunkSize = m_lastModelEndFrame - chunkStart;
	    }
	    nextChunkStart = chunkStart + chunkSize;
	}
	
//	std::cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << std::endl;

	if (!chunkSize) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
	    std::cout << "Ending selection playback at " << nextChunkStart << std::endl;
#endif
	    // We need to maintain full buffers so that the other
	    // thread can tell where it's got to in the playback -- so
	    // return the full amount here
	    frame = frame + count;
	    return count;
	}

#ifdef DEBUG_AUDIO_PLAY_SOURCE
	std::cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << std::endl;
#endif

	size_t got = 0;

	if (selectionSize < 100) {
	    fadeIn = 0;
	    fadeOut = 0;
	} else if (selectionSize < 300) {
	    if (fadeIn > 0) fadeIn = 10;
	    if (fadeOut > 0) fadeOut = 10;
	}

	if (fadeIn > 0) {
	    if (processed * 2 < fadeIn) {
		fadeIn = processed * 2;
	    }
	}

	if (fadeOut > 0) {
	    if ((count - processed - chunkSize) * 2 < fadeOut) {
		fadeOut = (count - processed - chunkSize) * 2;
	    }
	}

	for (std::set<Model *>::iterator mi = m_models.begin();
	     mi != m_models.end(); ++mi) {
	    
	    got = m_audioGenerator->mixModel(*mi, chunkStart, 
					     chunkSize, chunkBufferPtrs,
					     fadeIn, fadeOut);
	}

	for (size_t c = 0; c < channels; ++c) {
	    chunkBufferPtrs[c] += chunkSize;
	}

	processed += chunkSize;
	chunkStart = nextChunkStart;
    }

#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << std::endl;
#endif

    frame = nextChunkStart;
    return processed;
}

void
AudioCallbackPlaySource::unifyRingBuffers()
{
    if (m_readBuffers == m_writeBuffers) return;

    // only unify if there will be something to read
    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
	RingBuffer<float> *wb = getWriteRingBuffer(c);
	if (wb) {
	    if (wb->getReadSpace() < m_blockSize * 2) {
		if ((m_writeBufferFill + m_blockSize * 2) < 
		    m_lastModelEndFrame) {
		    // OK, we don't have enough and there's more to
		    // read -- don't unify until we can do better
		    return;
		}
	    }
	    break;
	}
    }

    size_t rf = m_readBufferFill;
    RingBuffer<float> *rb = getReadRingBuffer(0);
    if (rb) {
	size_t rs = rb->getReadSpace();
	//!!! incorrect when in non-contiguous selection, see comments elsewhere
//	std::cout << "rs = " << rs << std::endl;
	if (rs < rf) rf -= rs;
	else rf = 0;
    }
    
    //std::cout << "m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << std::endl;

    size_t wf = m_writeBufferFill;
    size_t skip = 0;
    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
	RingBuffer<float> *wb = getWriteRingBuffer(c);
	if (wb) {
	    if (c == 0) {
		
		size_t wrs = wb->getReadSpace();
//		std::cout << "wrs = " << wrs << std::endl;

		if (wrs < wf) wf -= wrs;
		else wf = 0;
//		std::cout << "wf = " << wf << std::endl;
		
		if (wf < rf) skip = rf - wf;
		if (skip == 0) break;
	    }

//	    std::cout << "skipping " << skip << std::endl;
	    wb->skip(skip);
	}
    }
		    
    m_bufferScavenger.claim(m_readBuffers);
    m_readBuffers = m_writeBuffers;
    m_readBufferFill = m_writeBufferFill;
//    std::cout << "unified" << std::endl;
}

void
AudioCallbackPlaySource::FillThread::run()
{
    AudioCallbackPlaySource &s(m_source);
    
#ifdef DEBUG_AUDIO_PLAY_SOURCE
    std::cout << "AudioCallbackPlaySourceFillThread starting" << std::endl;
#endif

    s.m_mutex.lock();

    bool previouslyPlaying = s.m_playing;
    bool work = false;

    while (!s.m_exiting) {

	s.unifyRingBuffers();
	s.m_bufferScavenger.scavenge();
        s.m_pluginScavenger.scavenge();

	if (work && s.m_playing && s.getSourceSampleRate()) {
	    
#ifdef DEBUG_AUDIO_PLAY_SOURCE
	    std::cout << "AudioCallbackPlaySourceFillThread: not waiting" << std::endl;
#endif

	    s.m_mutex.unlock();
	    s.m_mutex.lock();

	} else {
	    
	    float ms = 100;
	    if (s.getSourceSampleRate() > 0) {
		ms = float(m_ringBufferSize) / float(s.getSourceSampleRate()) * 1000.0;
	    }
	    
	    if (s.m_playing) ms /= 10;

#ifdef DEBUG_AUDIO_PLAY_SOURCE
            if (!s.m_playing) std::cout << std::endl;
	    std::cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << std::endl;
#endif
	    
	    s.m_condition.wait(&s.m_mutex, size_t(ms));
	}

#ifdef DEBUG_AUDIO_PLAY_SOURCE
	std::cout << "AudioCallbackPlaySourceFillThread: awoken" << std::endl;
#endif

	work = false;

	if (!s.getSourceSampleRate()) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
            std::cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << std::endl;
#endif
            continue;
        }

	bool playing = s.m_playing;

	if (playing && !previouslyPlaying) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
	    std::cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << std::endl;
#endif
	    for (size_t c = 0; c < s.getTargetChannelCount(); ++c) {
		RingBuffer<float> *rb = s.getReadRingBuffer(c);
		if (rb) rb->reset();
	    }
	}
	previouslyPlaying = playing;

	work = s.fillBuffers();
    }

    s.m_mutex.unlock();
}