diff audio/AudioCallbackPlaySource.cpp @ 468:56acd9368532 bqaudioio

Initial work toward switching to bqaudioio library (so as to get I/O, not just O)

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/audio/AudioCallbackPlaySource.cpp	Tue Aug 04 13:27:42 2015 +0100
@@ -0,0 +1,1902 @@
+/* -*- 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 "bqaudioio/SystemPlaybackTarget.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
+
+static const int DEFAULT_RING_BUFFER_SIZE = 131071;
+
+AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManagerBase *manager,
+                                                 QString clientName) :
+    m_viewManager(manager),
+    m_audioGenerator(new AudioGenerator()),
+    m_clientName(clientName.toUtf8().data()),
+    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_ringBufferSize(DEFAULT_RING_BUFFER_SIZE),
+    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_stretchMono(false),
+    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(this, SIGNAL(playStatusChanged(bool)),
+            m_viewManager, SLOT(playStatusChanged(bool)));
+
+    connect(PlayParameterRepository::getInstance(),
+	    SIGNAL(playParametersChanged(PlayParameters *)),
+	    this, SLOT(playParametersChanged(PlayParameters *)));
+
+    connect(Preferences::getInstance(),
+            SIGNAL(propertyChanged(PropertyContainer::PropertyName)),
+            this, SLOT(preferenceChanged(PropertyContainer::PropertyName)));
+}
+
+AudioCallbackPlaySource::~AudioCallbackPlaySource()
+{
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource entering" << endl;
+#endif
+    m_exiting = true;
+
+    if (m_fillThread) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource dtor: awakening thread" << endl;
+#endif
+        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 (int 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);
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource finishing" << endl;
+#endif
+}
+
+void
+AudioCallbackPlaySource::addModel(Model *model)
+{
+    if (m_models.find(model) != m_models.end()) return;
+
+    bool willPlay = 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;
+
+    int 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
+    cout << "AudioCallbackPlaySource: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << 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) {
+                    SVDEBUG << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << endl;
+                    conflicting = true;
+                    break;
+                }
+            }
+
+            if (conflicting) {
+
+                SVDEBUG << "AudioCallbackPlaySource::addModel: ERROR: "
+                          << "New model sample rate does not match" << endl
+                          << "existing model(s) (new " << model->getSampleRate()
+                          << " vs " << m_sourceSampleRate
+                          << "), playback will be wrong"
+                          << endl;
+                
+                emit sampleRateMismatch(model->getSampleRate(),
+                                        m_sourceSampleRate,
+                                        false);
+            } else {
+                m_sourceSampleRate = model->getSampleRate();
+                srChanged = true;
+            }
+        }
+    }
+
+    if (!m_writeBuffers || (int)m_writeBuffers->size() < getTargetChannelCount()) {
+	clearRingBuffers(true, getTargetChannelCount());
+	buffersChanged = true;
+    } else {
+	if (willPlay) clearRingBuffers(true);
+    }
+
+    if (buffersChanged || srChanged) {
+	if (m_converter) {
+	    src_delete(m_converter);
+            src_delete(m_crapConverter);
+	    m_converter = 0;
+            m_crapConverter = 0;
+	}
+    }
+
+    rebuildRangeLists();
+
+    m_mutex.unlock();
+
+    m_audioGenerator->setTargetChannelCount(getTargetChannelCount());
+
+    if (!m_fillThread) {
+	m_fillThread = new FillThread(*this);
+	m_fillThread->start();
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << endl;
+#endif
+
+    if (buffersChanged || srChanged) {
+	emit modelReplaced();
+    }
+
+    connect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)),
+            this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t)));
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::addModel: awakening thread" << endl;
+#endif
+
+    m_condition.wakeAll();
+}
+
+void
+AudioCallbackPlaySource::modelChangedWithin(sv_frame_t 
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+                                            startFrame
+#endif
+                                            , sv_frame_t endFrame)
+{
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    SVDEBUG << "AudioCallbackPlaySource::modelChangedWithin(" << startFrame << "," << endFrame << ")" << endl;
+#endif
+    if (endFrame > m_lastModelEndFrame) {
+        m_lastModelEndFrame = endFrame;
+        rebuildRangeLists();
+    }
+}
+
+void
+AudioCallbackPlaySource::removeModel(Model *model)
+{
+    m_mutex.lock();
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << endl;
+#endif
+
+    disconnect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)),
+               this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_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;
+    }
+
+    sv_frame_t lastEnd = 0;
+    for (std::set<Model *>::const_iterator i = m_models.begin();
+	 i != m_models.end(); ++i) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << endl;
+#endif
+	if ((*i)->getEndFrame() > lastEnd) {
+            lastEnd = (*i)->getEndFrame();
+        }
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	cout << "(done, lastEnd now " << lastEnd << ")" << endl;
+#endif
+    }
+    m_lastModelEndFrame = lastEnd;
+
+    m_audioGenerator->removeModel(model);
+
+    m_mutex.unlock();
+
+    clearRingBuffers();
+}
+
+void
+AudioCallbackPlaySource::clearModels()
+{
+    m_mutex.lock();
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::clearModels()" << 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, int count)
+{
+    if (!haveLock) m_mutex.lock();
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cerr << "clearRingBuffers" << endl;
+#endif
+
+    rebuildRangeLists();
+
+    if (count == 0) {
+	if (m_writeBuffers) count = int(m_writeBuffers->size());
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cerr << "current playing frame = " << getCurrentPlayingFrame() << endl;
+
+    cerr << "write buffer fill (before) = " << m_writeBufferFill << endl;
+#endif
+    
+    m_writeBufferFill = getCurrentBufferedFrame();
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cerr << "current buffered frame = " << m_writeBufferFill << endl;
+#endif
+
+    if (m_readBuffers != m_writeBuffers) {
+	delete m_writeBuffers;
+    }
+
+    m_writeBuffers = new RingBufferVector;
+
+    for (int i = 0; i < count; ++i) {
+	m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize));
+    }
+
+    m_audioGenerator->reset();
+    
+//    cout << "AudioCallbackPlaySource::clearRingBuffers: Created "
+//	      << count << " write buffers" << endl;
+
+    if (!haveLock) {
+	m_mutex.unlock();
+    }
+}
+
+void
+AudioCallbackPlaySource::play(sv_frame_t startFrame)
+{
+    if (!m_sourceSampleRate) {
+        cerr << "AudioCallbackPlaySource::play: No source sample rate available, not playing" << endl;
+        return;
+    }
+    
+    if (m_viewManager->getPlaySelectionMode() &&
+	!m_viewManager->getSelections().empty()) {
+
+        SVDEBUG << "AudioCallbackPlaySource::play: constraining frame " << startFrame << " to selection = ";
+
+        startFrame = m_viewManager->constrainFrameToSelection(startFrame);
+
+        SVDEBUG << startFrame << endl;
+
+    } else {
+        if (startFrame < 0) {
+            startFrame = 0;
+        }
+	if (startFrame >= m_lastModelEndFrame) {
+	    startFrame = 0;
+	}
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cerr << "play(" << startFrame << ") -> playback model ";
+#endif
+
+    startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cerr << startFrame << endl;
+#endif
+
+    // 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 (int c = 0; c < getTargetChannelCount(); ++c) {
+            RingBuffer<float> *rb = getReadRingBuffer(c);
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+            cerr << "reset ring buffer for channel " << c << 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;
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::play: awakening thread" << endl;
+#endif
+
+    m_condition.wakeAll();
+    if (changed) {
+        emit playStatusChanged(m_playing);
+        emit activity(tr("Play from %1").arg
+                      (RealTime::frame2RealTime
+                       (m_playStartFrame, m_sourceSampleRate).toText().c_str()));
+    }
+}
+
+void
+AudioCallbackPlaySource::stop()
+{
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    SVDEBUG << "AudioCallbackPlaySource::stop()" << endl;
+#endif
+    bool changed = m_playing;
+    m_playing = false;
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::stop: awakening thread" << endl;
+#endif
+
+    m_condition.wakeAll();
+    m_lastRetrievalTimestamp = 0;
+    if (changed) {
+        emit playStatusChanged(m_playing);
+        emit activity(tr("Stop at %1").arg
+                      (RealTime::frame2RealTime
+                       (m_lastCurrentFrame, m_sourceSampleRate).toText().c_str()));
+    }
+    m_lastCurrentFrame = 0;
+}
+
+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()
+{
+    cerr << "Audio processing overload!" << 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::setSystemPlaybackTarget(breakfastquay::SystemPlaybackTarget *target)
+{
+    m_target = target;
+}
+
+void
+AudioCallbackPlaySource::setSystemPlaybackBlockSize(int size)
+{
+    cout << "AudioCallbackPlaySource::setTarget: Block size -> " << size << endl;
+    if (size != 0) {
+        m_blockSize = size;
+    }
+    if (size * 4 > m_ringBufferSize) {
+        SVDEBUG << "AudioCallbackPlaySource::setTarget: Buffer size "
+                  << size << " > a quarter of ring buffer size "
+                  << m_ringBufferSize << ", calling for more ring buffer"
+                  << endl;
+        m_ringBufferSize = size * 4;
+        if (m_writeBuffers && !m_writeBuffers->empty()) {
+            clearRingBuffers();
+        }
+    }
+}
+
+int
+AudioCallbackPlaySource::getTargetBlockSize() const
+{
+//    cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl;
+    return int(m_blockSize);
+}
+
+void
+AudioCallbackPlaySource::setSystemPlaybackLatency(int latency)
+{
+    m_playLatency = latency;
+}
+
+sv_frame_t
+AudioCallbackPlaySource::getTargetPlayLatency() const
+{
+    return m_playLatency;
+}
+
+sv_frame_t
+AudioCallbackPlaySource::getCurrentPlayingFrame()
+{
+    // This method attempts to estimate which audio sample frame is
+    // "currently coming through the speakers".
+
+    sv_samplerate_t targetRate = getTargetSampleRate();
+    sv_frame_t latency = m_playLatency; // at target rate
+    RealTime latency_t = RealTime::zeroTime;
+
+    if (targetRate != 0) {
+        latency_t = RealTime::frame2RealTime(latency, targetRate);
+    }
+
+    return getCurrentFrame(latency_t);
+}
+
+sv_frame_t
+AudioCallbackPlaySource::getCurrentBufferedFrame()
+{
+    return getCurrentFrame(RealTime::zeroTime);
+}
+
+sv_frame_t
+AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t)
+{
+    // 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.
+
+    sv_samplerate_t sourceRate = getSourceSampleRate();
+    sv_samplerate_t targetRate = getTargetSampleRate();
+
+    if (sourceRate == 0 || targetRate == 0) return 0;
+
+    int inbuffer = 0; // at target rate
+
+    for (int c = 0; c < getTargetChannelCount(); ++c) {
+	RingBuffer<float> *rb = getReadRingBuffer(c);
+	if (rb) {
+	    int here = rb->getReadSpace();
+	    if (c == 0 || here < inbuffer) inbuffer = here;
+	}
+    }
+
+    sv_frame_t readBufferFill = m_readBufferFill;
+    sv_frame_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);
+
+    sv_frame_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;
+        latency_t = latency_t / timeRatio;
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    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 quantity: " << lastretrieved_t << endl;
+#endif
+
+    // 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;
+        if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;
+        sv_frame_t frame = RealTime::realTime2Frame(playing_t, sourceRate);
+        return m_viewManager->alignPlaybackFrameToReference(frame);
+    }
+
+    int inRange = 0;
+    int index = 0;
+
+    for (int i = 0; i < (int)m_rangeStarts.size(); ++i) {
+        if (bufferedto_t >= m_rangeStarts[i]) {
+            inRange = index;
+        } else {
+            break;
+        }
+        ++index;
+    }
+
+    if (inRange >= int(m_rangeStarts.size())) {
+        inRange = int(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) {
+//            cerr << "playing_t " << playing_t << " < playstart_t " 
+//                      << playstart_t << endl;
+            if (/*!!! sincerequest_t > RealTime::zeroTime && */
+                m_playStartedAt + latency_t + stretchlat_t <
+                RealTime::fromSeconds(currentTime)) {
+//                cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << endl;
+                m_playStartFramePassed = true;
+            } else {
+                playing_t = playstart_t;
+            }
+        } else {
+            m_playStartFramePassed = true;
+        }
+    }
+ 
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    cerr << "playing_t " << playing_t;
+#endif
+
+    playing_t = playing_t - m_rangeStarts[inRange];
+ 
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    cerr << " as offset into range " << inRange << " (start =" << m_rangeStarts[inRange] << " duration =" << m_rangeDurations[inRange] << ") = " << playing_t << endl;
+#endif
+
+    while (playing_t < RealTime::zeroTime) {
+
+        if (inRange == 0) {
+            if (looping) {
+                inRange = int(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
+    cerr << "  playing time: " << playing_t << endl;
+#endif
+
+    if (!looping) {
+        if (inRange == (int)m_rangeStarts.size()-1 &&
+            playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) {
+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" << endl;
+            stop();
+        }
+    }
+
+    if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;
+
+    sv_frame_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();
+
+    sv_samplerate_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
+    SVDEBUG << "AudioCallbackPlaySource::rebuildRangeLists" << 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
+    cerr << "Now have " << m_rangeStarts.size() << " play ranges" << 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::setSystemPlaybackSampleRate(int sr)
+{
+    bool first = (m_targetSampleRate == 0);
+
+    m_targetSampleRate = sr;
+    initialiseConverter();
+
+    if (first && (m_stretchRatio != 1.f)) {
+        // couldn't create a stretcher before because we had no sample
+        // rate: make one now
+        setTimeStretch(m_stretchRatio);
+    }
+}
+
+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) {
+	    cerr
+		<< "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: "
+		<< src_strerror(err) << 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
+    SVDEBUG << "AudioCallbackPlaySource::setResampleQuality: setting to "
+              << m_resampleQuality << endl;
+#endif
+
+    initialiseConverter();
+}
+
+void
+AudioCallbackPlaySource::setAuditioningEffect(Auditionable *a)
+{
+    RealTimePluginInstance *plugin = dynamic_cast<RealTimePluginInstance *>(a);
+    if (a && !plugin) {
+        cerr << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << endl;
+    }
+
+    m_mutex.lock();
+    m_auditioningPlugin = plugin;
+    m_auditioningPluginBypassed = false;
+    m_mutex.unlock();
+}
+
+void
+AudioCallbackPlaySource::setSoloModelSet(std::set<Model *> s)
+{
+    m_audioGenerator->setSoloModelSet(s);
+    clearRingBuffers();
+}
+
+void
+AudioCallbackPlaySource::clearSoloModelSet()
+{
+    m_audioGenerator->clearSoloModelSet();
+    clearRingBuffers();
+}
+
+sv_samplerate_t
+AudioCallbackPlaySource::getTargetSampleRate() const
+{
+    if (m_targetSampleRate) return m_targetSampleRate;
+    else return getSourceSampleRate();
+}
+
+int
+AudioCallbackPlaySource::getSourceChannelCount() const
+{
+    return m_sourceChannelCount;
+}
+
+int
+AudioCallbackPlaySource::getTargetChannelCount() const
+{
+    if (m_sourceChannelCount < 2) return 2;
+    return m_sourceChannelCount;
+}
+
+sv_samplerate_t
+AudioCallbackPlaySource::getSourceSampleRate() const
+{
+    return m_sourceSampleRate;
+}
+
+void
+AudioCallbackPlaySource::setTimeStretch(double factor)
+{
+    m_stretchRatio = factor;
+
+    if (!getTargetSampleRate()) return; // have to make our stretcher later
+
+    if (m_timeStretcher || (factor == 1.0)) {
+        // stretch ratio will be set in next process call if appropriate
+    } else {
+        m_stretcherInputCount = getTargetChannelCount();
+        RubberBandStretcher *stretcher = new RubberBandStretcher
+            (int(getTargetSampleRate()),
+             m_stretcherInputCount,
+             RubberBandStretcher::OptionProcessRealTime,
+             factor);
+        RubberBandStretcher *monoStretcher = new RubberBandStretcher
+            (int(getTargetSampleRate()),
+             1,
+             RubberBandStretcher::OptionProcessRealTime,
+             factor);
+        m_stretcherInputs = new float *[m_stretcherInputCount];
+        m_stretcherInputSizes = new sv_frame_t[m_stretcherInputCount];
+        for (int c = 0; c < m_stretcherInputCount; ++c) {
+            m_stretcherInputSizes[c] = 16384;
+            m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
+        }
+        m_monoStretcher = monoStretcher;
+        m_timeStretcher = stretcher;
+    }
+
+    emit activity(tr("Change time-stretch factor to %1").arg(factor));
+}
+
+void
+AudioCallbackPlaySource::getSourceSamples(int count, float **buffer)
+{
+    if (!m_playing) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+        SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Not playing" << endl;
+#endif
+	for (int ch = 0; ch < getTargetChannelCount(); ++ch) {
+	    for (int i = 0; i < count; ++i) {
+		buffer[ch][i] = 0.0;
+	    }
+	}
+	return;
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Playing" << endl;
+#endif
+
+    // Ensure that all buffers have at least the amount of data we
+    // need -- else reduce the size of our requests correspondingly
+
+    for (int ch = 0; ch < getTargetChannelCount(); ++ch) {
+
+        RingBuffer<float> *rb = getReadRingBuffer(ch);
+        
+        if (!rb) {
+            cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
+                      << "No ring buffer available for channel " << ch
+                      << ", returning no data here" << endl;
+            count = 0;
+            break;
+        }
+
+        int rs = rb->getReadSpace();
+        if (rs < count) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+            cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
+                      << "Ring buffer for channel " << ch << " has only "
+                      << rs << " (of " << count << ") samples available ("
+                      << "ring buffer size is " << rb->getSize() << ", write "
+                      << "space " << rb->getWriteSpace() << "), "
+                      << "reducing request size" << endl;
+#endif
+            count = rs;
+        }
+    }
+
+    if (count == 0) return;
+
+    RubberBandStretcher *ts = m_timeStretcher;
+    RubberBandStretcher *ms = m_monoStretcher;
+
+    double ratio = ts ? ts->getTimeRatio() : 1.0;
+
+    if (ratio != m_stretchRatio) {
+        if (!ts) {
+            cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << endl;
+            m_stretchRatio = 1.0;
+        } 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 (int 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":
+		sv_frame_t request = count;
+		if (ch > 0) request = got;
+
+		got = rb->read(buffer[ch], int(request));
+	    
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+		cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << endl;
+#endif
+	    }
+
+	    for (int ch = 0; ch < getTargetChannelCount(); ++ch) {
+		for (int i = got; i < count; ++i) {
+		    buffer[ch][i] = 0.0;
+		}
+	    }
+	}
+
+        applyAuditioningEffect(count, buffer);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::getSamples: awakening thread" << endl;
+#endif
+
+        m_condition.wakeAll();
+
+	return;
+    }
+
+    int channels = getTargetChannelCount();
+    sv_frame_t available;
+    sv_frame_t fedToStretcher = 0;
+    int warned = 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) {
+
+        sv_frame_t reqd = lrint(double(count - available) / ratio);
+        reqd = std::max(reqd, sv_frame_t(ts->getSamplesRequired()));
+        if (reqd == 0) reqd = 1;
+                
+        sv_frame_t got = reqd;
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+        cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << endl;
+#endif
+
+        for (int c = 0; c < channels; ++c) {
+            if (c >= m_stretcherInputCount) continue;
+            if (reqd > m_stretcherInputSizes[c]) {
+                if (c == 0) {
+                    cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << endl;
+                }
+                delete[] m_stretcherInputs[c];
+                m_stretcherInputSizes[c] = reqd * 2;
+                m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
+            }
+        }
+
+        for (int c = 0; c < channels; ++c) {
+            if (c >= m_stretcherInputCount) continue;
+            RingBuffer<float> *rb = getReadRingBuffer(c);
+            if (rb) {
+                sv_frame_t gotHere;
+                if (stretchChannels == 1 && c > 0) {
+                    gotHere = rb->readAdding(m_stretcherInputs[0], int(got));
+                } else {
+                    gotHere = rb->read(m_stretcherInputs[c], int(got));
+                }
+                if (gotHere < got) got = gotHere;
+                
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+                if (c == 0) {
+                    SVDEBUG << "feeding stretcher: got " << gotHere
+                              << ", " << rb->getReadSpace() << " remain" << endl;
+                }
+#endif
+                
+            } else {
+                cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << endl;
+            }
+        }
+
+        if (got < reqd) {
+            cerr << "WARNING: Read underrun in playback ("
+                      << got << " < " << reqd << ")" << endl;
+        }
+
+        ts->process(m_stretcherInputs, size_t(got), false);
+
+        fedToStretcher += got;
+
+        if (got == 0) break;
+
+        if (ts->available() == available) {
+            cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << endl;
+            if (++warned == 5) break;
+        }
+    }
+
+    ts->retrieve(buffer, size_t(count));
+
+    for (int c = stretchChannels; c < getTargetChannelCount(); ++c) {
+        for (int i = 0; i < count; ++i) {
+            buffer[c][i] = buffer[0][i];
+        }
+    }
+
+    applyAuditioningEffect(count, buffer);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySource::getSamples [stretched]: awakening thread" << endl;
+#endif
+
+    m_condition.wakeAll();
+
+    return;
+}
+
+void
+AudioCallbackPlaySource::applyAuditioningEffect(sv_frame_t count, float **buffers)
+{
+    if (m_auditioningPluginBypassed) return;
+    RealTimePluginInstance *plugin = m_auditioningPlugin;
+    if (!plugin) return;
+    
+    if ((int)plugin->getAudioInputCount() != getTargetChannelCount()) {
+//        cerr << "plugin input count " << plugin->getAudioInputCount() 
+//                  << " != our channel count " << getTargetChannelCount()
+//                  << endl;
+        return;
+    }
+    if ((int)plugin->getAudioOutputCount() != getTargetChannelCount()) {
+//        cerr << "plugin output count " << plugin->getAudioOutputCount() 
+//                  << " != our channel count " << getTargetChannelCount()
+//                  << endl;
+        return;
+    }
+    if ((int)plugin->getBufferSize() < count) {
+//        cerr << "plugin buffer size " << plugin->getBufferSize() 
+//                  << " < our block size " << count
+//                  << endl;
+        return;
+    }
+
+    float **ib = plugin->getAudioInputBuffers();
+    float **ob = plugin->getAudioOutputBuffers();
+
+    for (int c = 0; c < getTargetChannelCount(); ++c) {
+        for (int i = 0; i < count; ++i) {
+            ib[c][i] = buffers[c][i];
+        }
+    }
+
+    plugin->run(Vamp::RealTime::zeroTime, int(count));
+    
+    for (int c = 0; c < getTargetChannelCount(); ++c) {
+        for (int 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 sv_frame_t tmpSize = 0;
+
+    sv_frame_t space = 0;
+    for (int c = 0; c < getTargetChannelCount(); ++c) {
+	RingBuffer<float> *wb = getWriteRingBuffer(c);
+	if (wb) {
+	    sv_frame_t spaceHere = wb->getWriteSpace();
+	    if (c == 0 || spaceHere < space) space = spaceHere;
+	}
+    }
+    
+    if (space == 0) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+        cout << "AudioCallbackPlaySourceFillThread: no space to fill" << endl;
+#endif
+        return false;
+    }
+
+    sv_frame_t f = m_writeBufferFill;
+	
+    bool readWriteEqual = (m_readBuffers == m_writeBuffers);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    if (!readWriteEqual) {
+        cout << "AudioCallbackPlaySourceFillThread: note read buffers != write buffers" << endl;
+    }
+    cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << endl;
+#endif
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "buffered to " << f << " already" << endl;
+#endif
+
+    bool resample = (getSourceSampleRate() != getTargetSampleRate());
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << endl;
+#endif
+
+    int channels = getTargetChannelCount();
+
+    sv_frame_t orig = space;
+    sv_frame_t got = 0;
+
+    static float **bufferPtrs = 0;
+    static int bufferPtrCount = 0;
+
+    if (bufferPtrCount < channels) {
+	if (bufferPtrs) delete[] bufferPtrs;
+	bufferPtrs = new float *[channels];
+	bufferPtrCount = channels;
+    }
+
+    sv_frame_t generatorBlockSize = m_audioGenerator->getBlockSize();
+
+    if (resample && !m_converter) {
+	static bool warned = false;
+	if (!warned) {
+	    cerr << "WARNING: sample rates differ, but no converter available!" << endl;
+	    warned = true;
+	}
+    }
+
+    if (resample && m_converter) {
+
+	double ratio =
+	    double(getTargetSampleRate()) / double(getSourceSampleRate());
+	orig = sv_frame_t(double(orig) / ratio + 0.1);
+
+	// orig must be a multiple of generatorBlockSize
+	orig = (orig / generatorBlockSize) * generatorBlockSize;
+	if (orig == 0) return false;
+
+	sv_frame_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 (int c = 0; c < channels; ++c) {
+	    for (int i = 0; i < orig; ++i) {
+		nonintlv[channels * i + c] = 0.0f;
+	    }
+	}
+
+	for (int c = 0; c < channels; ++c) {
+	    bufferPtrs[c] = nonintlv + c * orig;
+	}
+
+	got = mixModels(f, orig, bufferPtrs); // also modifies f
+
+	// and interleave into first half
+	for (int c = 0; c < channels; ++c) {
+	    for (int 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 = long(got);
+	data.output_frames = long(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
+            cout << "Using crappy converter" << endl;
+#endif
+            err = src_process(m_crapConverter, &data);
+        } else {
+            err = src_process(m_converter, &data);
+        }
+
+	sv_frame_t toCopy = sv_frame_t(double(got) * ratio + 0.1);
+
+	if (err) {
+	    cerr
+		<< "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: "
+		<< src_strerror(err) << endl;
+	    //!!! Then what?
+	} else {
+	    got = data.input_frames_used;
+	    toCopy = data.output_frames_gen;
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    cout << "Resampled " << got << " frames to " << toCopy << " frames" << endl;
+#endif
+	}
+	
+	for (int c = 0; c < channels; ++c) {
+	    for (int i = 0; i < toCopy; ++i) {
+		tmp[i] = srcout[channels * i + c];
+	    }
+	    RingBuffer<float> *wb = getWriteRingBuffer(c);
+	    if (wb) wb->write(tmp, int(toCopy));
+	}
+
+	m_writeBufferFill = f;
+	if (readWriteEqual) m_readBufferFill = f;
+
+    } else {
+
+	// space must be a multiple of generatorBlockSize
+        sv_frame_t reqSpace = space;
+	space = (reqSpace / generatorBlockSize) * generatorBlockSize;
+	if (space == 0) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+            cout << "requested fill of " << reqSpace
+                      << " is less than generator block size of "
+                      << generatorBlockSize << ", leaving it" << endl;
+#endif
+            return false;
+        }
+
+	if (tmpSize < channels * space) {
+	    delete[] tmp;
+	    tmp = new float[channels * space];
+	    tmpSize = channels * space;
+	}
+
+	for (int c = 0; c < channels; ++c) {
+
+	    bufferPtrs[c] = tmp + c * space;
+	    
+	    for (int i = 0; i < space; ++i) {
+		tmp[c * space + i] = 0.0f;
+	    }
+	}
+
+	sv_frame_t got = mixModels(f, space, bufferPtrs); // also modifies f
+
+	for (int c = 0; c < channels; ++c) {
+
+	    RingBuffer<float> *wb = getWriteRingBuffer(c);
+	    if (wb) {
+                int actual = wb->write(bufferPtrs[c], int(got));
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+		cout << "Wrote " << actual << " samples for ch " << c << ", now "
+			  << wb->getReadSpace() << " to read" 
+			  << endl;
+#endif
+                if (actual < got) {
+                    cerr << "WARNING: Buffer overrun in channel " << c
+                              << ": wrote " << actual << " of " << got
+                              << " samples" << endl;
+                }
+            }
+	}
+
+	m_writeBufferFill = f;
+	if (readWriteEqual) m_readBufferFill = f;
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+        cout << "Read buffer fill is now " << m_readBufferFill << endl;
+#endif
+
+	//!!! 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;
+}    
+
+sv_frame_t
+AudioCallbackPlaySource::mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers)
+{
+    sv_frame_t processed = 0;
+    sv_frame_t chunkStart = frame;
+    sv_frame_t chunkSize = count;
+    sv_frame_t selectionSize = 0;
+    sv_frame_t nextChunkStart = chunkStart + chunkSize;
+    
+    bool looping = m_viewManager->getPlayLoopMode();
+    bool constrained = (m_viewManager->getPlaySelectionMode() &&
+			!m_viewManager->getSelections().empty());
+
+    static float **chunkBufferPtrs = 0;
+    static int chunkBufferPtrCount = 0;
+    int channels = getTargetChannelCount();
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << endl;
+#endif
+
+    if (chunkBufferPtrCount < channels) {
+	if (chunkBufferPtrs) delete[] chunkBufferPtrs;
+	chunkBufferPtrs = new float *[channels];
+	chunkBufferPtrCount = channels;
+    }
+
+    for (int c = 0; c < channels; ++c) {
+	chunkBufferPtrs[c] = buffers[c];
+    }
+
+    while (processed < count) {
+	
+	chunkSize = count - processed;
+	nextChunkStart = chunkStart + chunkSize;
+	selectionSize = 0;
+
+	sv_frame_t fadeIn = 0, fadeOut = 0;
+
+	if (constrained) {
+
+            sv_frame_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 {
+
+                sv_frame_t sf = m_viewManager->alignReferenceToPlaybackFrame
+                    (selection.getStartFrame());
+                sv_frame_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;
+	}
+	
+//	cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << endl;
+
+	if (!chunkSize) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    cout << "Ending selection playback at " << nextChunkStart << 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
+	cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << endl;
+#endif
+
+	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) {
+	    
+	    (void) m_audioGenerator->mixModel(*mi, chunkStart, 
+                                              chunkSize, chunkBufferPtrs,
+                                              fadeIn, fadeOut);
+	}
+
+	for (int c = 0; c < channels; ++c) {
+	    chunkBufferPtrs[c] += chunkSize;
+	}
+
+	processed += chunkSize;
+	chunkStart = nextChunkStart;
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << 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 (int 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
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+                    SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: Not unifying: write buffer has less (" << wb->getReadSpace() << ") than " << m_blockSize*2 << " to read and write buffer fill (" << m_writeBufferFill << ") is not close to end frame (" << m_lastModelEndFrame << ")" << endl;
+#endif
+		    return;
+		}
+	    }
+	    break;
+	}
+    }
+
+    sv_frame_t rf = m_readBufferFill;
+    RingBuffer<float> *rb = getReadRingBuffer(0);
+    if (rb) {
+	int rs = rb->getReadSpace();
+	//!!! incorrect when in non-contiguous selection, see comments elsewhere
+//	cout << "rs = " << rs << endl;
+	if (rs < rf) rf -= rs;
+	else rf = 0;
+    }
+    
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    SVDEBUG << "AudioCallbackPlaySource::unifyRingBuffers: m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << endl;
+#endif
+
+    sv_frame_t wf = m_writeBufferFill;
+    sv_frame_t skip = 0;
+    for (int c = 0; c < getTargetChannelCount(); ++c) {
+	RingBuffer<float> *wb = getWriteRingBuffer(c);
+	if (wb) {
+	    if (c == 0) {
+		
+		int wrs = wb->getReadSpace();
+//		cout << "wrs = " << wrs << endl;
+
+		if (wrs < wf) wf -= wrs;
+		else wf = 0;
+//		cout << "wf = " << wf << endl;
+		
+		if (wf < rf) skip = rf - wf;
+		if (skip == 0) break;
+	    }
+
+//	    cout << "skipping " << skip << endl;
+	    wb->skip(int(skip));
+	}
+    }
+		    
+    m_bufferScavenger.claim(m_readBuffers);
+    m_readBuffers = m_writeBuffers;
+    m_readBufferFill = m_writeBufferFill;
+#ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
+    cerr << "unified" << endl;
+#endif
+}
+
+void
+AudioCallbackPlaySource::FillThread::run()
+{
+    AudioCallbackPlaySource &s(m_source);
+    
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    cout << "AudioCallbackPlaySourceFillThread starting" << 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
+	    cout << "AudioCallbackPlaySourceFillThread: not waiting" << endl;
+#endif
+
+	    s.m_mutex.unlock();
+	    s.m_mutex.lock();
+
+	} else {
+	    
+	    double ms = 100;
+	    if (s.getSourceSampleRate() > 0) {
+		ms = double(s.m_ringBufferSize) / s.getSourceSampleRate() * 1000.0;
+	    }
+	    
+	    if (s.m_playing) ms /= 10;
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+            if (!s.m_playing) cout << endl;
+	    cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << endl;
+#endif
+	    
+	    s.m_condition.wait(&s.m_mutex, int(ms));
+	}
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	cout << "AudioCallbackPlaySourceFillThread: awoken" << endl;
+#endif
+
+	work = false;
+
+	if (!s.getSourceSampleRate()) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+            cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << endl;
+#endif
+            continue;
+        }
+
+	bool playing = s.m_playing;
+
+	if (playing && !previouslyPlaying) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << endl;
+#endif
+	    for (int 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();
+}
+