annotate sv/audioio/AudioCallbackPlaySource.cpp @ 82:8ebc85f6ce4e

integrate Pitch-Time scaling filter from DIT

/* -*- 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 "view/ViewManager.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 "PhaseVocoderTimeStretcher.h"

#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(ViewManager *manager) :
    m_viewManager(manager),
    m_audioGenerator(new AudioGenerator()),
    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_playing(false),
    m_exiting(false),
    m_lastModelEndFrame(0),
    m_outputLeft(0.0),
    m_outputRight(0.0),
    m_auditioningPlugin(0),
    m_auditioningPluginBypassed(false),
    m_timeStretcher(0),
    m_fillThread(0),
    m_converter(0),
    m_crapConverter(0),
    m_resampleQuality(Preferences::getInstance()->getResampleQuality()),
	m_filterStack(0)
{
    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;

    m_bufferScavenger.scavenge(true);
    m_pluginScavenger.scavenge(true);
    m_timeStretcherScavenger.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 " << 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();
    }

    m_condition.wakeAll();
}

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

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

    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();
}    

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

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

    size_t sf = m_readBufferFill;
    RingBuffer<float> *rb = getReadRingBuffer(0);
    if (rb) {
	//!!! This is incorrect if we're in a non-contiguous selection
	//Same goes for all related code (subtracting the read space
	//from the fill frame to try to establish where the effective
	//pre-resample/timestretch read pointer is)
	size_t rs = rb->getReadSpace();
	if (rs < sf) sf -= rs;
	else sf = 0;
    }
    m_writeBufferFill = sf;

    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()) {
	MultiSelection::SelectionList selections = m_viewManager->getSelections();
	MultiSelection::SelectionList::iterator i = selections.begin();
	if (i != selections.end()) {
	    if (startFrame < i->getStartFrame()) {
		startFrame = i->getStartFrame();
	    } else {
		MultiSelection::SelectionList::iterator j = selections.end();
		--j;
		if (startFrame >= j->getEndFrame()) {
		    startFrame = i->getStartFrame();
		}
	    }
	}
    } else {
	if (startFrame >= m_lastModelEndFrame) {
	    startFrame = 0;
	}
    }

    // 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.

    m_mutex.lock();
    if (m_playing) {
	m_readBufferFill = m_writeBufferFill = startFrame;
	if (m_readBuffers) {
	    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
		RingBuffer<float> *rb = getReadRingBuffer(c);
		if (rb) rb->reset();
	    }
	}
	if (m_converter) src_reset(m_converter);
        if (m_crapConverter) src_reset(m_crapConverter);
    } else {
	if (m_converter) src_reset(m_converter);
        if (m_crapConverter) src_reset(m_crapConverter);
	m_readBufferFill = m_writeBufferFill = startFrame;
    }
    m_mutex.unlock();

    m_audioGenerator->reset();

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

void
AudioCallbackPlaySource::setTargetBlockSize(size_t size)
{
//    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()
{
    bool resample = false;
    double ratio = 1.0;

    if (getSourceSampleRate() != getTargetSampleRate()) {
	resample = true;
	ratio = double(getSourceSampleRate()) / double(getTargetSampleRate());
    }

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

    if (resample) {
	readSpace = size_t(readSpace * ratio + 0.1);
    }

    size_t latency = m_playLatency;
    if (resample) latency = size_t(m_playLatency * ratio + 0.1);

    PhaseVocoderTimeStretcher *timeStretcher = m_timeStretcher;
    if (timeStretcher) {
	latency += timeStretcher->getProcessingLatency();
    }

    latency += readSpace;
    size_t bufferedFrame = m_readBufferFill;

    bool looping = m_viewManager->getPlayLoopMode();
    bool constrained = (m_viewManager->getPlaySelectionMode() &&
			!m_viewManager->getSelections().empty());

    size_t framePlaying = bufferedFrame;

    if (looping && !constrained) {
	while (framePlaying < latency) framePlaying += m_lastModelEndFrame;
    }

    if (framePlaying > latency) framePlaying -= latency;
    else framePlaying = 0;

    if (!constrained) {
	if (!looping && framePlaying > m_lastModelEndFrame) {
	    framePlaying = m_lastModelEndFrame;
	    stop();
	}
	return framePlaying;
    }

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

//    i = selections.begin();
//    size_t rangeStart = i->getStartFrame();

    i = selections.end();
    --i;
    size_t rangeEnd = i->getEndFrame();

    for (i = selections.begin(); i != selections.end(); ++i) {
	if (i->contains(bufferedFrame)) break;
    }

    size_t f = bufferedFrame;

//    std::cout << "getCurrentPlayingFrame: f=" << f << ", latency=" << latency << ", rangeEnd=" << rangeEnd << std::endl;

    if (i == selections.end()) {
	--i;
	if (i->getEndFrame() + latency < f) {
//    std::cout << "framePlaying = " << framePlaying << ", rangeEnd = " << rangeEnd << std::endl;

	    if (!looping && (framePlaying > rangeEnd)) {
//		std::cout << "STOPPING" << std::endl;
		stop();
		return rangeEnd;
	    } else {
		return framePlaying;
	    }
	} else {
//	    std::cout << "latency <- " << latency << "-(" << f << "-" << i->getEndFrame() << ")" << std::endl;
	    latency -= (f - i->getEndFrame());
	    f = i->getEndFrame();
	}
    }

//    std::cout << "i=(" << i->getStartFrame() << "," << i->getEndFrame() << ") f=" << f << ", latency=" << latency << std::endl;

    while (latency > 0) {
	size_t offset = f - i->getStartFrame();
	if (offset >= latency) {
	    if (f > latency) {
		framePlaying = f - latency;
	    } else {
		framePlaying = 0;
	    }
	    break;
	} else {
	    if (i == selections.begin()) {
		if (looping) {
		    i = selections.end();
		}
	    }
	    latency -= offset;
	    --i;
	    f = i->getEndFrame();
	}
    }

    return framePlaying;
}

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::setAuditioningPlugin(RealTimePluginInstance *plugin)
{
    RealTimePluginInstance *formerPlugin = m_auditioningPlugin;
    m_auditioningPlugin = plugin;
    m_auditioningPluginBypassed = false;
    if (formerPlugin) m_pluginScavenger.claim(formerPlugin);
}

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, bool sharpen, bool mono)
{
    // Avoid locks -- create, assign, mark old one for scavenging
    // later (as a call to getSourceSamples may still be using it)

    PhaseVocoderTimeStretcher *existingStretcher = m_timeStretcher;

    size_t channels = getTargetChannelCount();
    if (mono) channels = 1;

    if (existingStretcher &&
        existingStretcher->getRatio() == factor &&
        existingStretcher->getSharpening() == sharpen &&
        existingStretcher->getChannelCount() == channels) {
	return;
    }

    if (factor != 1) {

        if (existingStretcher &&
            existingStretcher->getSharpening() == sharpen &&
            existingStretcher->getChannelCount() == channels) {
            existingStretcher->setRatio(factor);
            return;
        }

	PhaseVocoderTimeStretcher *newStretcher = new PhaseVocoderTimeStretcher
	    (getTargetSampleRate(),
             channels,
             factor,
             sharpen,
             getTargetBlockSize());

	m_timeStretcher = newStretcher;

    } else {
	m_timeStretcher = 0;
    }

    if (existingStretcher) {
	m_timeStretcherScavenger.claim(existingStretcher);
    }
}

size_t
AudioCallbackPlaySource::getSourceSamples(size_t count, float **buffer)
{
    if (!m_playing) {
	for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
	    for (size_t 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;

    PhaseVocoderTimeStretcher *ts = m_timeStretcher;

    if (!ts || ts->getRatio() == 1) {

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

        applyAuditioningEffect(count, buffer);

		applyRealTimeFilters(count, buffer);

        m_condition.wakeAll();
	return got;
    }

    float ratio = ts->getRatio();

//            std::cout << "ratio = " << ratio << std::endl;

    size_t channels = getTargetChannelCount();
    bool mix = (channels > 1 && ts->getChannelCount() == 1);

    size_t available;

    int warned = 0;

    // We want output blocks of e.g. 1024 (probably fixed, certainly
    // bounded).  We can provide input blocks of any size (unbounded)
    // at the timestretcher's request.  The input block for a given
    // output is approx output / ratio, but we can't predict it
    // exactly, for an adaptive timestretcher.  The stretcher will
    // need some additional buffer space.  See the time stretcher code
    // and comments.

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

        size_t reqd = lrintf((count - available) / ratio);
        reqd = max(reqd, ts->getRequiredInputSamples());
        if (reqd == 0) reqd = 1;
                
        //float *ib[channels];
		float **ib = (float**) malloc(channels*sizeof(float*));

        size_t got = reqd;

        if (mix) {
            for (size_t c = 0; c < channels; ++c) {
                if (c == 0) ib[c] = new float[reqd]; //!!! fix -- this is a rt function
                else ib[c] = 0;
                RingBuffer<float> *rb = getReadRingBuffer(c);
                if (rb) {
                    size_t gotHere;
                    if (c > 0) gotHere = rb->readAdding(ib[0], got);
                    else gotHere = rb->read(ib[0], got);
                    if (gotHere < got) got = gotHere;
                }
            }
        } else {
            for (size_t c = 0; c < channels; ++c) {
                ib[c] = new float[reqd]; //!!! fix -- this is a rt function
                RingBuffer<float> *rb = getReadRingBuffer(c);
                if (rb) {
                    size_t gotHere = rb->read(ib[c], got);
                    if (gotHere < got) got = gotHere;
                }
            }
        }

        if (got < reqd) {
            std::cerr << "WARNING: Read underrun in playback ("
                      << got << " < " << reqd << ")" << std::endl;
        }
                
        ts->putInput(ib, got);

        for (size_t c = 0; c < channels; ++c) {
            delete[] ib[c];
        }

        if (got == 0) break;

        if (ts->getAvailableOutputSamples() == 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->getOutput(buffer, count);

    if (mix) {
        for (size_t c = 1; c < channels; ++c) {
            for (size_t i = 0; i < count; ++i) {
                buffer[c][i] = buffer[0][i] / channels;
            }
        }
        for (size_t i = 0; i < count; ++i) {
            buffer[0][i] /= channels;
        }
    }

    applyAuditioningEffect(count, buffer);

	applyRealTimeFilters(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);
    
    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) 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 = 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->getRatio() < 0.4) {
#ifdef DEBUG_AUDIO_PLAY_SOURCE
            std::cout << "Using crappy converter" << std::endl;
#endif
            src_process(m_crapConverter, &data);
        } else {
            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) 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) {
	    
	    Selection selection =
		m_viewManager->getContainingSelection(chunkStart, true);
	    
	    if (selection.isEmpty()) {
		if (looping) {
		    selection = *m_viewManager->getSelections().begin();
		    chunkStart = selection.getStartFrame();
		    fadeIn = 50;
		}
	    }

	    if (selection.isEmpty()) {

		chunkSize = 0;
		nextChunkStart = chunkStart;

	    } else {

		selectionSize =
		    selection.getEndFrame() -
		    selection.getStartFrame();

		if (chunkStart < selection.getStartFrame()) {
		    chunkStart = selection.getStartFrame();
		    fadeIn = 50;
		}

		nextChunkStart = chunkStart + chunkSize;

		if (nextChunkStart >= selection.getEndFrame()) {
		    nextChunkStart = selection.getEndFrame();
		    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();
	s.m_timeStretcherScavenger.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()) 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();
}

void AudioCallbackPlaySource::applyRealTimeFilters(size_t count, float **buffers)
{
	if (!m_filterStack) return;

	size_t required = m_filterStack->getRequiredInputSamples(count);

	if (required <= count)
	{
		m_filterStack->putInput(buffers, count);
		
	} else 
	{
		size_t missing = required - count;

		size_t channels = getTargetChannelCount();

		size_t got = required;

		float **ib = (float**) malloc(channels*sizeof(float*));
        
        for (size_t c = 0; c < channels; ++c) {
			ib[c] = (float*) malloc(required*sizeof(float));
			for (int i=0; i<count; i++)
			{
				ib[c][i] = buffers[c][i];
			}
            RingBuffer<float> *rb = getReadRingBuffer(c);
            if (rb) {
				size_t gotHere = rb->peek(ib[c]+count, missing);
				if (gotHere < got)
					got = gotHere;
			}
        }
		if (got < missing)
		{
			std::cerr << "ERROR applyRealTimeFilters(): Read underrun in playback ("
                      << got << " < " << required << ")" << std::endl;
			return; 
		}

        m_filterStack->putInput(ib, required);

		for (size_t c = 0; c < channels; ++c) {
			delete ib[c];
		}
		delete ib;
	}
	m_filterStack->getOutput(buffers, count);

}