diff audioio/AudioCallbackPlaySource.cpp @ 43:3c5756fb6a68

* Move some things around to facilitate plundering libraries for other
  applications without needing to duplicate so much code.
    sv/osc -> data/osc
    sv/audioio -> audioio
    sv/transform -> plugin/transform
    sv/document -> document (will rename to framework in next commit)

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/audioio/AudioCallbackPlaySource.cpp	Wed Oct 24 16:34:31 2007 +0000
@@ -0,0 +1,1493 @@
+/* -*- 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_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();
+    }
+
+    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;
+}
+
+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();
+}    
+
+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);
+}
+
+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, 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);
+
+        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 = std::max(reqd, ts->getRequiredInputSamples());
+        if (reqd == 0) reqd = 1;
+                
+        float *ib[channels];
+
+        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);
+
+    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 = 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->getRatio() < 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) 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();
+}
+