Chris@43: /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
Chris@43: 
Chris@43: /*
Chris@43:     Sonic Visualiser
Chris@43:     An audio file viewer and annotation editor.
Chris@43:     Centre for Digital Music, Queen Mary, University of London.
Chris@43:     This file copyright 2006 Chris Cannam and QMUL.
Chris@43:     
Chris@43:     This program is free software; you can redistribute it and/or
Chris@43:     modify it under the terms of the GNU General Public License as
Chris@43:     published by the Free Software Foundation; either version 2 of the
Chris@43:     License, or (at your option) any later version.  See the file
Chris@43:     COPYING included with this distribution for more information.
Chris@43: */
Chris@43: 
Chris@43: #include "AudioCallbackPlaySource.h"
Chris@43: 
Chris@43: #include "AudioGenerator.h"
Chris@43: 
Chris@43: #include "data/model/Model.h"
Chris@43: #include "view/ViewManager.h"
Chris@43: #include "base/PlayParameterRepository.h"
Chris@43: #include "base/Preferences.h"
Chris@43: #include "data/model/DenseTimeValueModel.h"
Chris@43: #include "data/model/WaveFileModel.h"
Chris@43: #include "data/model/SparseOneDimensionalModel.h"
Chris@43: #include "plugin/RealTimePluginInstance.h"
Chris@62: 
Chris@91: #include "AudioCallbackPlayTarget.h"
Chris@91: 
Chris@62: #include <rubberband/RubberBandStretcher.h>
Chris@62: using namespace RubberBand;
Chris@43: 
Chris@43: #include <iostream>
Chris@43: #include <cassert>
Chris@43: 
Chris@43: //#define DEBUG_AUDIO_PLAY_SOURCE 1
Chris@43: //#define DEBUG_AUDIO_PLAY_SOURCE_PLAYING 1
Chris@43: 
Chris@43: const size_t AudioCallbackPlaySource::m_ringBufferSize = 131071;
Chris@43: 
Chris@57: AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManager *manager,
Chris@57:                                                  QString clientName) :
Chris@43:     m_viewManager(manager),
Chris@43:     m_audioGenerator(new AudioGenerator()),
Chris@57:     m_clientName(clientName),
Chris@43:     m_readBuffers(0),
Chris@43:     m_writeBuffers(0),
Chris@43:     m_readBufferFill(0),
Chris@43:     m_writeBufferFill(0),
Chris@43:     m_bufferScavenger(1),
Chris@43:     m_sourceChannelCount(0),
Chris@43:     m_blockSize(1024),
Chris@43:     m_sourceSampleRate(0),
Chris@43:     m_targetSampleRate(0),
Chris@43:     m_playLatency(0),
Chris@91:     m_target(0),
Chris@91:     m_lastRetrievalTimestamp(0.0),
Chris@91:     m_lastRetrievedBlockSize(0),
Chris@43:     m_playing(false),
Chris@43:     m_exiting(false),
Chris@43:     m_lastModelEndFrame(0),
Chris@43:     m_outputLeft(0.0),
Chris@43:     m_outputRight(0.0),
Chris@43:     m_auditioningPlugin(0),
Chris@43:     m_auditioningPluginBypassed(false),
Chris@94:     m_playStartFrame(0),
Chris@94:     m_playStartFramePassed(false),
Chris@43:     m_timeStretcher(0),
Chris@91:     m_stretchRatio(1.0),
Chris@91:     m_stretcherInputCount(0),
Chris@91:     m_stretcherInputs(0),
Chris@91:     m_stretcherInputSizes(0),
Chris@43:     m_fillThread(0),
Chris@43:     m_converter(0),
Chris@43:     m_crapConverter(0),
Chris@43:     m_resampleQuality(Preferences::getInstance()->getResampleQuality())
Chris@43: {
Chris@43:     m_viewManager->setAudioPlaySource(this);
Chris@43: 
Chris@43:     connect(m_viewManager, SIGNAL(selectionChanged()),
Chris@43: 	    this, SLOT(selectionChanged()));
Chris@43:     connect(m_viewManager, SIGNAL(playLoopModeChanged()),
Chris@43: 	    this, SLOT(playLoopModeChanged()));
Chris@43:     connect(m_viewManager, SIGNAL(playSelectionModeChanged()),
Chris@43: 	    this, SLOT(playSelectionModeChanged()));
Chris@43: 
Chris@43:     connect(PlayParameterRepository::getInstance(),
Chris@43: 	    SIGNAL(playParametersChanged(PlayParameters *)),
Chris@43: 	    this, SLOT(playParametersChanged(PlayParameters *)));
Chris@43: 
Chris@43:     connect(Preferences::getInstance(),
Chris@43:             SIGNAL(propertyChanged(PropertyContainer::PropertyName)),
Chris@43:             this, SLOT(preferenceChanged(PropertyContainer::PropertyName)));
Chris@43: }
Chris@43: 
Chris@43: AudioCallbackPlaySource::~AudioCallbackPlaySource()
Chris@43: {
Chris@43:     m_exiting = true;
Chris@43: 
Chris@43:     if (m_fillThread) {
Chris@43: 	m_condition.wakeAll();
Chris@43: 	m_fillThread->wait();
Chris@43: 	delete m_fillThread;
Chris@43:     }
Chris@43: 
Chris@43:     clearModels();
Chris@43:     
Chris@43:     if (m_readBuffers != m_writeBuffers) {
Chris@43: 	delete m_readBuffers;
Chris@43:     }
Chris@43: 
Chris@43:     delete m_writeBuffers;
Chris@43: 
Chris@43:     delete m_audioGenerator;
Chris@43: 
Chris@91:     for (size_t i = 0; i < m_stretcherInputCount; ++i) {
Chris@91:         delete[] m_stretcherInputs[i];
Chris@91:     }
Chris@91:     delete[] m_stretcherInputSizes;
Chris@91:     delete[] m_stretcherInputs;
Chris@91: 
Chris@43:     m_bufferScavenger.scavenge(true);
Chris@43:     m_pluginScavenger.scavenge(true);
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::addModel(Model *model)
Chris@43: {
Chris@43:     if (m_models.find(model) != m_models.end()) return;
Chris@43: 
Chris@43:     bool canPlay = m_audioGenerator->addModel(model);
Chris@43: 
Chris@43:     m_mutex.lock();
Chris@43: 
Chris@43:     m_models.insert(model);
Chris@43:     if (model->getEndFrame() > m_lastModelEndFrame) {
Chris@43: 	m_lastModelEndFrame = model->getEndFrame();
Chris@43:     }
Chris@43: 
Chris@43:     bool buffersChanged = false, srChanged = false;
Chris@43: 
Chris@43:     size_t modelChannels = 1;
Chris@43:     DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
Chris@43:     if (dtvm) modelChannels = dtvm->getChannelCount();
Chris@43:     if (modelChannels > m_sourceChannelCount) {
Chris@43: 	m_sourceChannelCount = modelChannels;
Chris@43:     }
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "Adding model with " << modelChannels << " channels " << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     if (m_sourceSampleRate == 0) {
Chris@43: 
Chris@43: 	m_sourceSampleRate = model->getSampleRate();
Chris@43: 	srChanged = true;
Chris@43: 
Chris@43:     } else if (model->getSampleRate() != m_sourceSampleRate) {
Chris@43: 
Chris@43:         // If this is a dense time-value model and we have no other, we
Chris@43:         // can just switch to this model's sample rate
Chris@43: 
Chris@43:         if (dtvm) {
Chris@43: 
Chris@43:             bool conflicting = false;
Chris@43: 
Chris@43:             for (std::set<Model *>::const_iterator i = m_models.begin();
Chris@43:                  i != m_models.end(); ++i) {
Chris@43:                 // Only wave file models can be considered conflicting --
Chris@43:                 // writable wave file models are derived and we shouldn't
Chris@43:                 // take their rates into account.  Also, don't give any
Chris@43:                 // particular weight to a file that's already playing at
Chris@43:                 // the wrong rate anyway
Chris@43:                 WaveFileModel *wfm = dynamic_cast<WaveFileModel *>(*i);
Chris@43:                 if (wfm && wfm != dtvm &&
Chris@43:                     wfm->getSampleRate() != model->getSampleRate() &&
Chris@43:                     wfm->getSampleRate() == m_sourceSampleRate) {
Chris@43:                     std::cerr << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << std::endl;
Chris@43:                     conflicting = true;
Chris@43:                     break;
Chris@43:                 }
Chris@43:             }
Chris@43: 
Chris@43:             if (conflicting) {
Chris@43: 
Chris@43:                 std::cerr << "AudioCallbackPlaySource::addModel: ERROR: "
Chris@43:                           << "New model sample rate does not match" << std::endl
Chris@43:                           << "existing model(s) (new " << model->getSampleRate()
Chris@43:                           << " vs " << m_sourceSampleRate
Chris@43:                           << "), playback will be wrong"
Chris@43:                           << std::endl;
Chris@43:                 
Chris@43:                 emit sampleRateMismatch(model->getSampleRate(),
Chris@43:                                         m_sourceSampleRate,
Chris@43:                                         false);
Chris@43:             } else {
Chris@43:                 m_sourceSampleRate = model->getSampleRate();
Chris@43:                 srChanged = true;
Chris@43:             }
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@43:     if (!m_writeBuffers || (m_writeBuffers->size() < getTargetChannelCount())) {
Chris@43: 	clearRingBuffers(true, getTargetChannelCount());
Chris@43: 	buffersChanged = true;
Chris@43:     } else {
Chris@43: 	if (canPlay) clearRingBuffers(true);
Chris@43:     }
Chris@43: 
Chris@43:     if (buffersChanged || srChanged) {
Chris@43: 	if (m_converter) {
Chris@43: 	    src_delete(m_converter);
Chris@43:             src_delete(m_crapConverter);
Chris@43: 	    m_converter = 0;
Chris@43:             m_crapConverter = 0;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@43:     m_mutex.unlock();
Chris@43: 
Chris@43:     m_audioGenerator->setTargetChannelCount(getTargetChannelCount());
Chris@43: 
Chris@43:     if (!m_fillThread) {
Chris@43: 	m_fillThread = new FillThread(*this);
Chris@43: 	m_fillThread->start();
Chris@43:     }
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     if (buffersChanged || srChanged) {
Chris@43: 	emit modelReplaced();
Chris@43:     }
Chris@43: 
Chris@43:     connect(model, SIGNAL(modelChanged(size_t, size_t)),
Chris@43:             this, SLOT(modelChanged(size_t, size_t)));
Chris@43: 
Chris@43:     m_condition.wakeAll();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::modelChanged(size_t startFrame, size_t endFrame)
Chris@43: {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cerr << "AudioCallbackPlaySource::modelChanged(" << startFrame << "," << endFrame << ")" << std::endl;
Chris@43: #endif
Chris@93:     if (endFrame > m_lastModelEndFrame) {
Chris@93:         m_lastModelEndFrame = endFrame;
Chris@93:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::removeModel(Model *model)
Chris@43: {
Chris@43:     m_mutex.lock();
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     disconnect(model, SIGNAL(modelChanged(size_t, size_t)),
Chris@43:                this, SLOT(modelChanged(size_t, size_t)));
Chris@43: 
Chris@43:     m_models.erase(model);
Chris@43: 
Chris@43:     if (m_models.empty()) {
Chris@43: 	if (m_converter) {
Chris@43: 	    src_delete(m_converter);
Chris@43:             src_delete(m_crapConverter);
Chris@43: 	    m_converter = 0;
Chris@43:             m_crapConverter = 0;
Chris@43: 	}
Chris@43: 	m_sourceSampleRate = 0;
Chris@43:     }
Chris@43: 
Chris@43:     size_t lastEnd = 0;
Chris@43:     for (std::set<Model *>::const_iterator i = m_models.begin();
Chris@43: 	 i != m_models.end(); ++i) {
Chris@43: //	std::cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << std::endl;
Chris@43: 	if ((*i)->getEndFrame() > lastEnd) lastEnd = (*i)->getEndFrame();
Chris@43: //	std::cout << "(done, lastEnd now " << lastEnd << ")" << std::endl;
Chris@43:     }
Chris@43:     m_lastModelEndFrame = lastEnd;
Chris@43: 
Chris@43:     m_mutex.unlock();
Chris@43: 
Chris@43:     m_audioGenerator->removeModel(model);
Chris@43: 
Chris@43:     clearRingBuffers();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::clearModels()
Chris@43: {
Chris@43:     m_mutex.lock();
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "AudioCallbackPlaySource::clearModels()" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     m_models.clear();
Chris@43: 
Chris@43:     if (m_converter) {
Chris@43: 	src_delete(m_converter);
Chris@43:         src_delete(m_crapConverter);
Chris@43: 	m_converter = 0;
Chris@43:         m_crapConverter = 0;
Chris@43:     }
Chris@43: 
Chris@43:     m_lastModelEndFrame = 0;
Chris@43: 
Chris@43:     m_sourceSampleRate = 0;
Chris@43: 
Chris@43:     m_mutex.unlock();
Chris@43: 
Chris@43:     m_audioGenerator->clearModels();
Chris@93: 
Chris@93:     clearRingBuffers();
Chris@43: }    
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::clearRingBuffers(bool haveLock, size_t count)
Chris@43: {
Chris@43:     if (!haveLock) m_mutex.lock();
Chris@43: 
Chris@93:     rebuildRangeLists();
Chris@93: 
Chris@43:     if (count == 0) {
Chris@43: 	if (m_writeBuffers) count = m_writeBuffers->size();
Chris@43:     }
Chris@43: 
Chris@93:     m_writeBufferFill = getCurrentBufferedFrame();
Chris@43: 
Chris@43:     if (m_readBuffers != m_writeBuffers) {
Chris@43: 	delete m_writeBuffers;
Chris@43:     }
Chris@43: 
Chris@43:     m_writeBuffers = new RingBufferVector;
Chris@43: 
Chris@43:     for (size_t i = 0; i < count; ++i) {
Chris@43: 	m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize));
Chris@43:     }
Chris@43: 
Chris@43: //    std::cout << "AudioCallbackPlaySource::clearRingBuffers: Created "
Chris@43: //	      << count << " write buffers" << std::endl;
Chris@43: 
Chris@43:     if (!haveLock) {
Chris@43: 	m_mutex.unlock();
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::play(size_t startFrame)
Chris@43: {
Chris@43:     if (m_viewManager->getPlaySelectionMode() &&
Chris@43: 	!m_viewManager->getSelections().empty()) {
Chris@60: 
Chris@94:         std::cerr << "AudioCallbackPlaySource::play: constraining frame " << startFrame << " to selection = ";
Chris@94: 
Chris@60:         startFrame = m_viewManager->constrainFrameToSelection(startFrame);
Chris@60: 
Chris@94:         std::cerr << startFrame << std::endl;
Chris@94: 
Chris@43:     } else {
Chris@43: 	if (startFrame >= m_lastModelEndFrame) {
Chris@43: 	    startFrame = 0;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@60:     std::cerr << "play(" << startFrame << ") -> playback model ";
Chris@60: 
Chris@60:     startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame);
Chris@60: 
Chris@60:     std::cerr << startFrame << std::endl;
Chris@60: 
Chris@43:     // The fill thread will automatically empty its buffers before
Chris@43:     // starting again if we have not so far been playing, but not if
Chris@43:     // we're just re-seeking.
Chris@43: 
Chris@43:     m_mutex.lock();
Chris@91:     if (m_timeStretcher) {
Chris@91:         m_timeStretcher->reset();
Chris@91:     }
Chris@43:     if (m_playing) {
Chris@93:         std::cerr << "playing already, resetting" << std::endl;
Chris@43: 	m_readBufferFill = m_writeBufferFill = startFrame;
Chris@43: 	if (m_readBuffers) {
Chris@43: 	    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 		RingBuffer<float> *rb = getReadRingBuffer(c);
Chris@93:                 std::cerr << "reset ring buffer for channel " << c << std::endl;
Chris@43: 		if (rb) rb->reset();
Chris@43: 	    }
Chris@43: 	}
Chris@43: 	if (m_converter) src_reset(m_converter);
Chris@43:         if (m_crapConverter) src_reset(m_crapConverter);
Chris@43:     } else {
Chris@43: 	if (m_converter) src_reset(m_converter);
Chris@43:         if (m_crapConverter) src_reset(m_crapConverter);
Chris@43: 	m_readBufferFill = m_writeBufferFill = startFrame;
Chris@43:     }
Chris@43:     m_mutex.unlock();
Chris@43: 
Chris@43:     m_audioGenerator->reset();
Chris@43: 
Chris@94:     m_playStartFrame = startFrame;
Chris@94:     m_playStartFramePassed = false;
Chris@94:     m_playStartedAt = RealTime::zeroTime;
Chris@94:     if (m_target) {
Chris@94:         m_playStartedAt = RealTime::fromSeconds(m_target->getCurrentTime());
Chris@94:     }
Chris@94: 
Chris@43:     bool changed = !m_playing;
Chris@91:     m_lastRetrievalTimestamp = 0;
Chris@43:     m_playing = true;
Chris@43:     m_condition.wakeAll();
Chris@43:     if (changed) emit playStatusChanged(m_playing);
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::stop()
Chris@43: {
Chris@43:     bool changed = m_playing;
Chris@43:     m_playing = false;
Chris@43:     m_condition.wakeAll();
Chris@91:     m_lastRetrievalTimestamp = 0;
Chris@43:     if (changed) emit playStatusChanged(m_playing);
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::selectionChanged()
Chris@43: {
Chris@43:     if (m_viewManager->getPlaySelectionMode()) {
Chris@43: 	clearRingBuffers();
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::playLoopModeChanged()
Chris@43: {
Chris@43:     clearRingBuffers();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::playSelectionModeChanged()
Chris@43: {
Chris@43:     if (!m_viewManager->getSelections().empty()) {
Chris@43: 	clearRingBuffers();
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::playParametersChanged(PlayParameters *)
Chris@43: {
Chris@43:     clearRingBuffers();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::preferenceChanged(PropertyContainer::PropertyName n)
Chris@43: {
Chris@43:     if (n == "Resample Quality") {
Chris@43:         setResampleQuality(Preferences::getInstance()->getResampleQuality());
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::audioProcessingOverload()
Chris@43: {
Chris@43:     RealTimePluginInstance *ap = m_auditioningPlugin;
Chris@43:     if (ap && m_playing && !m_auditioningPluginBypassed) {
Chris@43:         m_auditioningPluginBypassed = true;
Chris@43:         emit audioOverloadPluginDisabled();
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@91: AudioCallbackPlaySource::setTarget(AudioCallbackPlayTarget *target, size_t size)
Chris@43: {
Chris@91:     m_target = target;
Chris@43: //    std::cout << "AudioCallbackPlaySource::setTargetBlockSize() -> " << size << std::endl;
Chris@43:     assert(size < m_ringBufferSize);
Chris@43:     m_blockSize = size;
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getTargetBlockSize() const
Chris@43: {
Chris@43: //    std::cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << std::endl;
Chris@43:     return m_blockSize;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setTargetPlayLatency(size_t latency)
Chris@43: {
Chris@43:     m_playLatency = latency;
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getTargetPlayLatency() const
Chris@43: {
Chris@43:     return m_playLatency;
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getCurrentPlayingFrame()
Chris@43: {
Chris@91:     // This method attempts to estimate which audio sample frame is
Chris@91:     // "currently coming through the speakers".
Chris@91: 
Chris@93:     size_t targetRate = getTargetSampleRate();
Chris@93:     size_t latency = m_playLatency; // at target rate
Chris@93:     RealTime latency_t = RealTime::frame2RealTime(latency, targetRate);
Chris@93: 
Chris@93:     return getCurrentFrame(latency_t);
Chris@93: }
Chris@93: 
Chris@93: size_t
Chris@93: AudioCallbackPlaySource::getCurrentBufferedFrame()
Chris@93: {
Chris@93:     return getCurrentFrame(RealTime::zeroTime);
Chris@93: }
Chris@93: 
Chris@93: size_t
Chris@93: AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t)
Chris@93: {
Chris@43:     bool resample = false;
Chris@91:     double resampleRatio = 1.0;
Chris@43: 
Chris@91:     // We resample when filling the ring buffer, and time-stretch when
Chris@91:     // draining it.  The buffer contains data at the "target rate" and
Chris@91:     // the latency provided by the target is also at the target rate.
Chris@91:     // Because of the multiple rates involved, we do the actual
Chris@91:     // calculation using RealTime instead.
Chris@43: 
Chris@91:     size_t sourceRate = getSourceSampleRate();
Chris@91:     size_t targetRate = getTargetSampleRate();
Chris@91: 
Chris@91:     if (sourceRate == 0 || targetRate == 0) return 0;
Chris@91: 
Chris@91:     size_t inbuffer = 0; // at target rate
Chris@91: 
Chris@43:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *rb = getReadRingBuffer(c);
Chris@43: 	if (rb) {
Chris@91: 	    size_t here = rb->getReadSpace();
Chris@91: 	    if (c == 0 || here < inbuffer) inbuffer = here;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@91:     size_t readBufferFill = m_readBufferFill;
Chris@91:     size_t lastRetrievedBlockSize = m_lastRetrievedBlockSize;
Chris@91:     double lastRetrievalTimestamp = m_lastRetrievalTimestamp;
Chris@91:     double currentTime = 0.0;
Chris@91:     if (m_target) currentTime = m_target->getCurrentTime();
Chris@91: 
Chris@91:     RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, targetRate);
Chris@91: 
Chris@91:     size_t stretchlat = 0;
Chris@91:     double timeRatio = 1.0;
Chris@91: 
Chris@91:     if (m_timeStretcher) {
Chris@91:         stretchlat = m_timeStretcher->getLatency();
Chris@91:         timeRatio = m_timeStretcher->getTimeRatio();
Chris@43:     }
Chris@43: 
Chris@91:     RealTime stretchlat_t = RealTime::frame2RealTime(stretchlat, targetRate);
Chris@43: 
Chris@91:     // When the target has just requested a block from us, the last
Chris@91:     // sample it obtained was our buffer fill frame count minus the
Chris@91:     // amount of read space (converted back to source sample rate)
Chris@91:     // remaining now.  That sample is not expected to be played until
Chris@91:     // the target's play latency has elapsed.  By the time the
Chris@91:     // following block is requested, that sample will be at the
Chris@91:     // target's play latency minus the last requested block size away
Chris@91:     // from being played.
Chris@91: 
Chris@91:     RealTime sincerequest_t = RealTime::zeroTime;
Chris@91:     RealTime lastretrieved_t = RealTime::zeroTime;
Chris@91: 
Chris@91:     if (m_target && lastRetrievalTimestamp != 0.0) {
Chris@91: 
Chris@91:         lastretrieved_t = RealTime::frame2RealTime
Chris@91:             (lastRetrievedBlockSize, targetRate);
Chris@91: 
Chris@91:         // calculate number of frames at target rate that have elapsed
Chris@91:         // since the end of the last call to getSourceSamples
Chris@91: 
Chris@91:         double elapsed = currentTime - lastRetrievalTimestamp;
Chris@91: 
Chris@91:         if (elapsed > 0.0) {
Chris@91:             sincerequest_t = RealTime::fromSeconds(elapsed);
Chris@91:         }
Chris@91: 
Chris@91:     } else {
Chris@91: 
Chris@91:         lastretrieved_t = RealTime::frame2RealTime
Chris@91:             (getTargetBlockSize(), targetRate);
Chris@62:     }
Chris@91: 
Chris@91:     RealTime bufferedto_t = RealTime::frame2RealTime(readBufferFill, sourceRate);
Chris@91: 
Chris@91:     if (timeRatio != 1.0) {
Chris@91:         lastretrieved_t = lastretrieved_t / timeRatio;
Chris@91:         sincerequest_t = sincerequest_t / timeRatio;
Chris@43:     }
Chris@43: 
Chris@43:     bool looping = m_viewManager->getPlayLoopMode();
Chris@43: 
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@91:     std::cerr << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n  stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n  since request: " << sincerequest_t << ", last retrieved: " << lastretrieved_t << std::endl;
Chris@91: #endif
Chris@43: 
Chris@91:     RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate);
Chris@60: 
Chris@93:     // Normally the range lists should contain at least one item each
Chris@93:     // -- if playback is unconstrained, that item should report the
Chris@93:     // entire source audio duration.
Chris@43: 
Chris@93:     if (m_rangeStarts.empty()) {
Chris@93:         rebuildRangeLists();
Chris@93:     }
Chris@92: 
Chris@93:     if (m_rangeStarts.empty()) {
Chris@93:         // this code is only used in case of error in rebuildRangeLists
Chris@93:         RealTime playing_t = bufferedto_t
Chris@93:             - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t
Chris@93:             + sincerequest_t;
Chris@93:         size_t frame = RealTime::realTime2Frame(playing_t, sourceRate);
Chris@93:         return m_viewManager->alignPlaybackFrameToReference(frame);
Chris@93:     }
Chris@43: 
Chris@91:     int inRange = 0;
Chris@91:     int index = 0;
Chris@91: 
Chris@93:     for (size_t i = 0; i < m_rangeStarts.size(); ++i) {
Chris@93:         if (bufferedto_t >= m_rangeStarts[i]) {
Chris@93:             inRange = index;
Chris@93:         } else {
Chris@93:             break;
Chris@93:         }
Chris@93:         ++index;
Chris@93:     }
Chris@93: 
Chris@93:     if (inRange >= m_rangeStarts.size()) inRange = m_rangeStarts.size()-1;
Chris@93: 
Chris@94:     RealTime playing_t = bufferedto_t;
Chris@93: 
Chris@93:     playing_t = playing_t
Chris@93:         - latency_t - stretchlat_t - lastretrieved_t - inbuffer_t
Chris@93:         + sincerequest_t;
Chris@94: 
Chris@94:     // This rather gross little hack is used to ensure that latency
Chris@94:     // compensation doesn't result in the playback pointer appearing
Chris@94:     // to start earlier than the actual playback does.  It doesn't
Chris@94:     // work properly (hence the bail-out in the middle) because if we
Chris@94:     // are playing a relatively short looped region, the playing time
Chris@94:     // estimated from the buffer fill frame may have wrapped around
Chris@94:     // the region boundary and end up being much smaller than the
Chris@94:     // theoretical play start frame, perhaps even for the entire
Chris@94:     // duration of playback!
Chris@94: 
Chris@94:     if (!m_playStartFramePassed) {
Chris@94:         RealTime playstart_t = RealTime::frame2RealTime(m_playStartFrame,
Chris@94:                                                         sourceRate);
Chris@94:         if (playing_t < playstart_t) {
Chris@94: //            std::cerr << "playing_t " << playing_t << " < playstart_t " 
Chris@94: //                      << playstart_t << std::endl;
Chris@94:             if (sincerequest_t > RealTime::zeroTime &&
Chris@94:                 m_playStartedAt + latency_t + stretchlat_t <
Chris@94:                 RealTime::fromSeconds(currentTime)) {
Chris@94: //                std::cerr << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << std::endl;
Chris@94:                 m_playStartFramePassed = true;
Chris@94:             } else {
Chris@94:                 playing_t = playstart_t;
Chris@94:             }
Chris@94:         } else {
Chris@94:             m_playStartFramePassed = true;
Chris@94:         }
Chris@94:     }
Chris@94: 
Chris@94:     playing_t = playing_t - m_rangeStarts[inRange];
Chris@93:  
Chris@93: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@93:     std::cerr << "playing_t as offset into range " << inRange << " (with start = " << m_rangeStarts[inRange] << ") = " << playing_t << std::endl;
Chris@93: #endif
Chris@93: 
Chris@93:     while (playing_t < RealTime::zeroTime) {
Chris@93: 
Chris@93:         if (inRange == 0) {
Chris@93:             if (looping) {
Chris@93:                 inRange = m_rangeStarts.size() - 1;
Chris@93:             } else {
Chris@93:                 break;
Chris@93:             }
Chris@93:         } else {
Chris@93:             --inRange;
Chris@93:         }
Chris@93: 
Chris@93:         playing_t = playing_t + m_rangeDurations[inRange];
Chris@93:     }
Chris@93: 
Chris@93:     playing_t = playing_t + m_rangeStarts[inRange];
Chris@93: 
Chris@93: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@93:     std::cerr << "  playing time: " << playing_t << std::endl;
Chris@93: #endif
Chris@93: 
Chris@93:     if (!looping) {
Chris@93:         if (inRange == m_rangeStarts.size()-1 &&
Chris@93:             playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) {
Chris@93:             stop();
Chris@93:         }
Chris@93:     }
Chris@93: 
Chris@93:     if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;
Chris@93: 
Chris@93:     size_t frame = RealTime::realTime2Frame(playing_t, sourceRate);
Chris@93:     return m_viewManager->alignPlaybackFrameToReference(frame);
Chris@93: }
Chris@93: 
Chris@93: void
Chris@93: AudioCallbackPlaySource::rebuildRangeLists()
Chris@93: {
Chris@93:     bool constrained = (m_viewManager->getPlaySelectionMode());
Chris@93: 
Chris@93:     m_rangeStarts.clear();
Chris@93:     m_rangeDurations.clear();
Chris@93: 
Chris@93:     size_t sourceRate = getSourceSampleRate();
Chris@93:     if (sourceRate == 0) return;
Chris@93: 
Chris@93:     RealTime end = RealTime::frame2RealTime(m_lastModelEndFrame, sourceRate);
Chris@93:     if (end == RealTime::zeroTime) return;
Chris@93: 
Chris@93:     if (!constrained) {
Chris@93:         m_rangeStarts.push_back(RealTime::zeroTime);
Chris@93:         m_rangeDurations.push_back(end);
Chris@93:         return;
Chris@93:     }
Chris@93: 
Chris@93:     MultiSelection::SelectionList selections = m_viewManager->getSelections();
Chris@93:     MultiSelection::SelectionList::const_iterator i;
Chris@93: 
Chris@93: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@93:     std::cerr << "AudioCallbackPlaySource::rebuildRangeLists" << std::endl;
Chris@93: #endif
Chris@93: 
Chris@93:     if (!selections.empty()) {
Chris@91: 
Chris@91:         for (i = selections.begin(); i != selections.end(); ++i) {
Chris@91:             
Chris@91:             RealTime start =
Chris@91:                 (RealTime::frame2RealTime
Chris@91:                  (m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()),
Chris@91:                   sourceRate));
Chris@91:             RealTime duration = 
Chris@91:                 (RealTime::frame2RealTime
Chris@91:                  (m_viewManager->alignReferenceToPlaybackFrame(i->getEndFrame()) -
Chris@91:                   m_viewManager->alignReferenceToPlaybackFrame(i->getStartFrame()),
Chris@91:                   sourceRate));
Chris@91:             
Chris@93:             m_rangeStarts.push_back(start);
Chris@93:             m_rangeDurations.push_back(duration);
Chris@91:         }
Chris@93:     } else {
Chris@93:         m_rangeStarts.push_back(RealTime::zeroTime);
Chris@93:         m_rangeDurations.push_back(end);
Chris@43:     }
Chris@43: 
Chris@93: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@93:     std::cerr << "Now have " << m_rangeStarts.size() << " play ranges" << std::endl;
Chris@91: #endif
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setOutputLevels(float left, float right)
Chris@43: {
Chris@43:     m_outputLeft = left;
Chris@43:     m_outputRight = right;
Chris@43: }
Chris@43: 
Chris@43: bool
Chris@43: AudioCallbackPlaySource::getOutputLevels(float &left, float &right)
Chris@43: {
Chris@43:     left = m_outputLeft;
Chris@43:     right = m_outputRight;
Chris@43:     return true;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setTargetSampleRate(size_t sr)
Chris@43: {
Chris@43:     m_targetSampleRate = sr;
Chris@43:     initialiseConverter();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::initialiseConverter()
Chris@43: {
Chris@43:     m_mutex.lock();
Chris@43: 
Chris@43:     if (m_converter) {
Chris@43:         src_delete(m_converter);
Chris@43:         src_delete(m_crapConverter);
Chris@43:         m_converter = 0;
Chris@43:         m_crapConverter = 0;
Chris@43:     }
Chris@43: 
Chris@43:     if (getSourceSampleRate() != getTargetSampleRate()) {
Chris@43: 
Chris@43: 	int err = 0;
Chris@43: 
Chris@43: 	m_converter = src_new(m_resampleQuality == 2 ? SRC_SINC_BEST_QUALITY :
Chris@43:                               m_resampleQuality == 1 ? SRC_SINC_MEDIUM_QUALITY :
Chris@43:                               m_resampleQuality == 0 ? SRC_SINC_FASTEST :
Chris@43:                                                        SRC_SINC_MEDIUM_QUALITY,
Chris@43: 			      getTargetChannelCount(), &err);
Chris@43: 
Chris@43:         if (m_converter) {
Chris@43:             m_crapConverter = src_new(SRC_LINEAR,
Chris@43:                                       getTargetChannelCount(),
Chris@43:                                       &err);
Chris@43:         }
Chris@43: 
Chris@43: 	if (!m_converter || !m_crapConverter) {
Chris@43: 	    std::cerr
Chris@43: 		<< "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: "
Chris@43: 		<< src_strerror(err) << std::endl;
Chris@43: 
Chris@43:             if (m_converter) {
Chris@43:                 src_delete(m_converter);
Chris@43:                 m_converter = 0;
Chris@43:             } 
Chris@43: 
Chris@43:             if (m_crapConverter) {
Chris@43:                 src_delete(m_crapConverter);
Chris@43:                 m_crapConverter = 0;
Chris@43:             }
Chris@43: 
Chris@43:             m_mutex.unlock();
Chris@43: 
Chris@43:             emit sampleRateMismatch(getSourceSampleRate(),
Chris@43:                                     getTargetSampleRate(),
Chris@43:                                     false);
Chris@43: 	} else {
Chris@43: 
Chris@43:             m_mutex.unlock();
Chris@43: 
Chris@43:             emit sampleRateMismatch(getSourceSampleRate(),
Chris@43:                                     getTargetSampleRate(),
Chris@43:                                     true);
Chris@43:         }
Chris@43:     } else {
Chris@43:         m_mutex.unlock();
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setResampleQuality(int q)
Chris@43: {
Chris@43:     if (q == m_resampleQuality) return;
Chris@43:     m_resampleQuality = q;
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cerr << "AudioCallbackPlaySource::setResampleQuality: setting to "
Chris@43:               << m_resampleQuality << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     initialiseConverter();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setAuditioningPlugin(RealTimePluginInstance *plugin)
Chris@43: {
Chris@43:     RealTimePluginInstance *formerPlugin = m_auditioningPlugin;
Chris@43:     m_auditioningPlugin = plugin;
Chris@43:     m_auditioningPluginBypassed = false;
Chris@43:     if (formerPlugin) m_pluginScavenger.claim(formerPlugin);
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setSoloModelSet(std::set<Model *> s)
Chris@43: {
Chris@43:     m_audioGenerator->setSoloModelSet(s);
Chris@43:     clearRingBuffers();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::clearSoloModelSet()
Chris@43: {
Chris@43:     m_audioGenerator->clearSoloModelSet();
Chris@43:     clearRingBuffers();
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getTargetSampleRate() const
Chris@43: {
Chris@43:     if (m_targetSampleRate) return m_targetSampleRate;
Chris@43:     else return getSourceSampleRate();
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getSourceChannelCount() const
Chris@43: {
Chris@43:     return m_sourceChannelCount;
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getTargetChannelCount() const
Chris@43: {
Chris@43:     if (m_sourceChannelCount < 2) return 2;
Chris@43:     return m_sourceChannelCount;
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getSourceSampleRate() const
Chris@43: {
Chris@43:     return m_sourceSampleRate;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@91: AudioCallbackPlaySource::setTimeStretch(float factor)
Chris@43: {
Chris@91:     m_stretchRatio = factor;
Chris@91: 
Chris@91:     if (m_timeStretcher || (factor == 1.f)) {
Chris@91:         // stretch ratio will be set in next process call if appropriate
Chris@62:         return;
Chris@62:     } else {
Chris@91:         m_stretcherInputCount = getTargetChannelCount();
Chris@62:         RubberBandStretcher *stretcher = new RubberBandStretcher
Chris@62:             (getTargetSampleRate(),
Chris@91:              m_stretcherInputCount,
Chris@62:              RubberBandStretcher::OptionProcessRealTime,
Chris@62:              factor);
Chris@91:         m_stretcherInputs = new float *[m_stretcherInputCount];
Chris@91:         m_stretcherInputSizes = new size_t[m_stretcherInputCount];
Chris@91:         for (size_t c = 0; c < m_stretcherInputCount; ++c) {
Chris@91:             m_stretcherInputSizes[c] = 16384;
Chris@91:             m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
Chris@91:         }
Chris@62:         m_timeStretcher = stretcher;
Chris@62:         return;
Chris@62:     }
Chris@43: }
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::getSourceSamples(size_t count, float **buffer)
Chris@43: {
Chris@43:     if (!m_playing) {
Chris@43: 	for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
Chris@43: 	    for (size_t i = 0; i < count; ++i) {
Chris@43: 		buffer[ch][i] = 0.0;
Chris@43: 	    }
Chris@43: 	}
Chris@43: 	return 0;
Chris@43:     }
Chris@43: 
Chris@43:     // Ensure that all buffers have at least the amount of data we
Chris@43:     // need -- else reduce the size of our requests correspondingly
Chris@43: 
Chris@43:     for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
Chris@43: 
Chris@43:         RingBuffer<float> *rb = getReadRingBuffer(ch);
Chris@43:         
Chris@43:         if (!rb) {
Chris@43:             std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
Chris@43:                       << "No ring buffer available for channel " << ch
Chris@43:                       << ", returning no data here" << std::endl;
Chris@43:             count = 0;
Chris@43:             break;
Chris@43:         }
Chris@43: 
Chris@43:         size_t rs = rb->getReadSpace();
Chris@43:         if (rs < count) {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:             std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
Chris@43:                       << "Ring buffer for channel " << ch << " has only "
Chris@43:                       << rs << " (of " << count << ") samples available, "
Chris@43:                       << "reducing request size" << std::endl;
Chris@43: #endif
Chris@43:             count = rs;
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@43:     if (count == 0) return 0;
Chris@43: 
Chris@62:     RubberBandStretcher *ts = m_timeStretcher;
Chris@62:     float ratio = ts ? ts->getTimeRatio() : 1.f;
Chris@91: 
Chris@91:     if (ratio != m_stretchRatio) {
Chris@91:         if (!ts) {
Chris@91:             std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << std::endl;
Chris@91:             m_stretchRatio = 1.f;
Chris@91:         } else {
Chris@91:             ts->setTimeRatio(m_stretchRatio);
Chris@91:         }
Chris@91:     }
Chris@91: 
Chris@91:     if (m_target) {
Chris@91:         m_lastRetrievedBlockSize = count;
Chris@91:         m_lastRetrievalTimestamp = m_target->getCurrentTime();
Chris@91:     }
Chris@43: 
Chris@62:     if (!ts || ratio == 1.f) {
Chris@43: 
Chris@43: 	size_t got = 0;
Chris@43: 
Chris@43: 	for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
Chris@43: 
Chris@43: 	    RingBuffer<float> *rb = getReadRingBuffer(ch);
Chris@43: 
Chris@43: 	    if (rb) {
Chris@43: 
Chris@43: 		// this is marginally more likely to leave our channels in
Chris@43: 		// sync after a processing failure than just passing "count":
Chris@43: 		size_t request = count;
Chris@43: 		if (ch > 0) request = got;
Chris@43: 
Chris@43: 		got = rb->read(buffer[ch], request);
Chris@43: 	    
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@43: 		std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << std::endl;
Chris@43: #endif
Chris@43: 	    }
Chris@43: 
Chris@43: 	    for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
Chris@43: 		for (size_t i = got; i < count; ++i) {
Chris@43: 		    buffer[ch][i] = 0.0;
Chris@43: 		}
Chris@43: 	    }
Chris@43: 	}
Chris@43: 
Chris@43:         applyAuditioningEffect(count, buffer);
Chris@43: 
Chris@43:         m_condition.wakeAll();
Chris@91: 
Chris@43: 	return got;
Chris@43:     }
Chris@43: 
Chris@62:     size_t channels = getTargetChannelCount();
Chris@91:     size_t available;
Chris@91:     int warned = 0;
Chris@91:     size_t fedToStretcher = 0;
Chris@43: 
Chris@91:     // The input block for a given output is approx output / ratio,
Chris@91:     // but we can't predict it exactly, for an adaptive timestretcher.
Chris@91: 
Chris@91:     while ((available = ts->available()) < count) {
Chris@91: 
Chris@91:         size_t reqd = lrintf((count - available) / ratio);
Chris@91:         reqd = std::max(reqd, ts->getSamplesRequired());
Chris@91:         if (reqd == 0) reqd = 1;
Chris@91:                 
Chris@91:         size_t got = reqd;
Chris@91: 
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@91:         std::cerr << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << std::endl;
Chris@62: #endif
Chris@43: 
Chris@91:         for (size_t c = 0; c < channels; ++c) {
Chris@91:             if (c >= m_stretcherInputCount) continue;
Chris@91:             if (reqd > m_stretcherInputSizes[c]) {
Chris@91:                 if (c == 0) {
Chris@91:                     std::cerr << "WARNING: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << std::endl;
Chris@91:                 }
Chris@91:                 delete[] m_stretcherInputs[c];
Chris@91:                 m_stretcherInputSizes[c] = reqd * 2;
Chris@91:                 m_stretcherInputs[c] = new float[m_stretcherInputSizes[c]];
Chris@91:             }
Chris@91:         }
Chris@43: 
Chris@91:         for (size_t c = 0; c < channels; ++c) {
Chris@91:             if (c >= m_stretcherInputCount) continue;
Chris@91:             RingBuffer<float> *rb = getReadRingBuffer(c);
Chris@91:             if (rb) {
Chris@91:                 size_t gotHere = rb->read(m_stretcherInputs[c], got);
Chris@91:                 if (gotHere < got) got = gotHere;
Chris@91:                 
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@91:                 if (c == 0) {
Chris@91:                     std::cerr << "feeding stretcher: got " << gotHere
Chris@91:                               << ", " << rb->getReadSpace() << " remain" << std::endl;
Chris@91:                 }
Chris@62: #endif
Chris@43:                 
Chris@91:             } else {
Chris@91:                 std::cerr << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << std::endl;
Chris@43:             }
Chris@43:         }
Chris@43: 
Chris@43:         if (got < reqd) {
Chris@43:             std::cerr << "WARNING: Read underrun in playback ("
Chris@43:                       << got << " < " << reqd << ")" << std::endl;
Chris@43:         }
Chris@43: 
Chris@91:         ts->process(m_stretcherInputs, got, false);
Chris@91: 
Chris@91:         fedToStretcher += got;
Chris@43: 
Chris@43:         if (got == 0) break;
Chris@43: 
Chris@62:         if (ts->available() == available) {
Chris@43:             std::cerr << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << std::endl;
Chris@43:             if (++warned == 5) break;
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@62:     ts->retrieve(buffer, count);
Chris@43: 
Chris@43:     applyAuditioningEffect(count, buffer);
Chris@43: 
Chris@43:     m_condition.wakeAll();
Chris@43: 
Chris@43:     return count;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::applyAuditioningEffect(size_t count, float **buffers)
Chris@43: {
Chris@43:     if (m_auditioningPluginBypassed) return;
Chris@43:     RealTimePluginInstance *plugin = m_auditioningPlugin;
Chris@43:     if (!plugin) return;
Chris@43: 
Chris@43:     if (plugin->getAudioInputCount() != getTargetChannelCount()) {
Chris@43: //        std::cerr << "plugin input count " << plugin->getAudioInputCount() 
Chris@43: //                  << " != our channel count " << getTargetChannelCount()
Chris@43: //                  << std::endl;
Chris@43:         return;
Chris@43:     }
Chris@43:     if (plugin->getAudioOutputCount() != getTargetChannelCount()) {
Chris@43: //        std::cerr << "plugin output count " << plugin->getAudioOutputCount() 
Chris@43: //                  << " != our channel count " << getTargetChannelCount()
Chris@43: //                  << std::endl;
Chris@43:         return;
Chris@43:     }
Chris@43:     if (plugin->getBufferSize() != count) {
Chris@43: //        std::cerr << "plugin buffer size " << plugin->getBufferSize() 
Chris@43: //                  << " != our block size " << count
Chris@43: //                  << std::endl;
Chris@43:         return;
Chris@43:     }
Chris@43: 
Chris@43:     float **ib = plugin->getAudioInputBuffers();
Chris@43:     float **ob = plugin->getAudioOutputBuffers();
Chris@43: 
Chris@43:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43:         for (size_t i = 0; i < count; ++i) {
Chris@43:             ib[c][i] = buffers[c][i];
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@43:     plugin->run(Vamp::RealTime::zeroTime);
Chris@43:     
Chris@43:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43:         for (size_t i = 0; i < count; ++i) {
Chris@43:             buffers[c][i] = ob[c][i];
Chris@43:         }
Chris@43:     }
Chris@43: }    
Chris@43: 
Chris@43: // Called from fill thread, m_playing true, mutex held
Chris@43: bool
Chris@43: AudioCallbackPlaySource::fillBuffers()
Chris@43: {
Chris@43:     static float *tmp = 0;
Chris@43:     static size_t tmpSize = 0;
Chris@43: 
Chris@43:     size_t space = 0;
Chris@43:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	if (wb) {
Chris@43: 	    size_t spaceHere = wb->getWriteSpace();
Chris@43: 	    if (c == 0 || spaceHere < space) space = spaceHere;
Chris@43: 	}
Chris@43:     }
Chris@43:     
Chris@43:     if (space == 0) return false;
Chris@43: 
Chris@43:     size_t f = m_writeBufferFill;
Chris@43: 	
Chris@43:     bool readWriteEqual = (m_readBuffers == m_writeBuffers);
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "buffered to " << f << " already" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     bool resample = (getSourceSampleRate() != getTargetSampleRate());
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     size_t channels = getTargetChannelCount();
Chris@43: 
Chris@43:     size_t orig = space;
Chris@43:     size_t got = 0;
Chris@43: 
Chris@43:     static float **bufferPtrs = 0;
Chris@43:     static size_t bufferPtrCount = 0;
Chris@43: 
Chris@43:     if (bufferPtrCount < channels) {
Chris@43: 	if (bufferPtrs) delete[] bufferPtrs;
Chris@43: 	bufferPtrs = new float *[channels];
Chris@43: 	bufferPtrCount = channels;
Chris@43:     }
Chris@43: 
Chris@43:     size_t generatorBlockSize = m_audioGenerator->getBlockSize();
Chris@43: 
Chris@43:     if (resample && !m_converter) {
Chris@43: 	static bool warned = false;
Chris@43: 	if (!warned) {
Chris@43: 	    std::cerr << "WARNING: sample rates differ, but no converter available!" << std::endl;
Chris@43: 	    warned = true;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@43:     if (resample && m_converter) {
Chris@43: 
Chris@43: 	double ratio =
Chris@43: 	    double(getTargetSampleRate()) / double(getSourceSampleRate());
Chris@43: 	orig = size_t(orig / ratio + 0.1);
Chris@43: 
Chris@43: 	// orig must be a multiple of generatorBlockSize
Chris@43: 	orig = (orig / generatorBlockSize) * generatorBlockSize;
Chris@43: 	if (orig == 0) return false;
Chris@43: 
Chris@43: 	size_t work = std::max(orig, space);
Chris@43: 
Chris@43: 	// We only allocate one buffer, but we use it in two halves.
Chris@43: 	// We place the non-interleaved values in the second half of
Chris@43: 	// the buffer (orig samples for channel 0, orig samples for
Chris@43: 	// channel 1 etc), and then interleave them into the first
Chris@43: 	// half of the buffer.  Then we resample back into the second
Chris@43: 	// half (interleaved) and de-interleave the results back to
Chris@43: 	// the start of the buffer for insertion into the ringbuffers.
Chris@43: 	// What a faff -- especially as we've already de-interleaved
Chris@43: 	// the audio data from the source file elsewhere before we
Chris@43: 	// even reach this point.
Chris@43: 	
Chris@43: 	if (tmpSize < channels * work * 2) {
Chris@43: 	    delete[] tmp;
Chris@43: 	    tmp = new float[channels * work * 2];
Chris@43: 	    tmpSize = channels * work * 2;
Chris@43: 	}
Chris@43: 
Chris@43: 	float *nonintlv = tmp + channels * work;
Chris@43: 	float *intlv = tmp;
Chris@43: 	float *srcout = tmp + channels * work;
Chris@43: 	
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 	    for (size_t i = 0; i < orig; ++i) {
Chris@43: 		nonintlv[channels * i + c] = 0.0f;
Chris@43: 	    }
Chris@43: 	}
Chris@43: 
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 	    bufferPtrs[c] = nonintlv + c * orig;
Chris@43: 	}
Chris@43: 
Chris@43: 	got = mixModels(f, orig, bufferPtrs);
Chris@43: 
Chris@43: 	// and interleave into first half
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 	    for (size_t i = 0; i < got; ++i) {
Chris@43: 		float sample = nonintlv[c * got + i];
Chris@43: 		intlv[channels * i + c] = sample;
Chris@43: 	    }
Chris@43: 	}
Chris@43: 		
Chris@43: 	SRC_DATA data;
Chris@43: 	data.data_in = intlv;
Chris@43: 	data.data_out = srcout;
Chris@43: 	data.input_frames = got;
Chris@43: 	data.output_frames = work;
Chris@43: 	data.src_ratio = ratio;
Chris@43: 	data.end_of_input = 0;
Chris@43: 	
Chris@43: 	int err = 0;
Chris@43: 
Chris@62:         if (m_timeStretcher && m_timeStretcher->getTimeRatio() < 0.4) {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:             std::cout << "Using crappy converter" << std::endl;
Chris@43: #endif
Chris@43:             err = src_process(m_crapConverter, &data);
Chris@43:         } else {
Chris@43:             err = src_process(m_converter, &data);
Chris@43:         }
Chris@43: 
Chris@43: 	size_t toCopy = size_t(got * ratio + 0.1);
Chris@43: 
Chris@43: 	if (err) {
Chris@43: 	    std::cerr
Chris@43: 		<< "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: "
Chris@43: 		<< src_strerror(err) << std::endl;
Chris@43: 	    //!!! Then what?
Chris@43: 	} else {
Chris@43: 	    got = data.input_frames_used;
Chris@43: 	    toCopy = data.output_frames_gen;
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 	    std::cout << "Resampled " << got << " frames to " << toCopy << " frames" << std::endl;
Chris@43: #endif
Chris@43: 	}
Chris@43: 	
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 	    for (size_t i = 0; i < toCopy; ++i) {
Chris@43: 		tmp[i] = srcout[channels * i + c];
Chris@43: 	    }
Chris@43: 	    RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	    if (wb) wb->write(tmp, toCopy);
Chris@43: 	}
Chris@43: 
Chris@43: 	m_writeBufferFill = f;
Chris@43: 	if (readWriteEqual) m_readBufferFill = f;
Chris@43: 
Chris@43:     } else {
Chris@43: 
Chris@43: 	// space must be a multiple of generatorBlockSize
Chris@43: 	space = (space / generatorBlockSize) * generatorBlockSize;
Chris@91: 	if (space == 0) {
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@91:             std::cout << "requested fill is less than generator block size of "
Chris@91:                       << generatorBlockSize << ", leaving it" << std::endl;
Chris@91: #endif
Chris@91:             return false;
Chris@91:         }
Chris@43: 
Chris@43: 	if (tmpSize < channels * space) {
Chris@43: 	    delete[] tmp;
Chris@43: 	    tmp = new float[channels * space];
Chris@43: 	    tmpSize = channels * space;
Chris@43: 	}
Chris@43: 
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 
Chris@43: 	    bufferPtrs[c] = tmp + c * space;
Chris@43: 	    
Chris@43: 	    for (size_t i = 0; i < space; ++i) {
Chris@43: 		tmp[c * space + i] = 0.0f;
Chris@43: 	    }
Chris@43: 	}
Chris@43: 
Chris@43: 	size_t got = mixModels(f, space, bufferPtrs);
Chris@43: 
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 
Chris@43: 	    RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	    if (wb) {
Chris@43:                 size_t actual = wb->write(bufferPtrs[c], got);
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 		std::cout << "Wrote " << actual << " samples for ch " << c << ", now "
Chris@43: 			  << wb->getReadSpace() << " to read" 
Chris@43: 			  << std::endl;
Chris@43: #endif
Chris@43:                 if (actual < got) {
Chris@43:                     std::cerr << "WARNING: Buffer overrun in channel " << c
Chris@43:                               << ": wrote " << actual << " of " << got
Chris@43:                               << " samples" << std::endl;
Chris@43:                 }
Chris@43:             }
Chris@43: 	}
Chris@43: 
Chris@43: 	m_writeBufferFill = f;
Chris@43: 	if (readWriteEqual) m_readBufferFill = f;
Chris@43: 
Chris@43: 	//!!! 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
Chris@43:     }
Chris@43: 
Chris@43:     return true;
Chris@43: }    
Chris@43: 
Chris@43: size_t
Chris@43: AudioCallbackPlaySource::mixModels(size_t &frame, size_t count, float **buffers)
Chris@43: {
Chris@43:     size_t processed = 0;
Chris@43:     size_t chunkStart = frame;
Chris@43:     size_t chunkSize = count;
Chris@43:     size_t selectionSize = 0;
Chris@43:     size_t nextChunkStart = chunkStart + chunkSize;
Chris@43:     
Chris@43:     bool looping = m_viewManager->getPlayLoopMode();
Chris@43:     bool constrained = (m_viewManager->getPlaySelectionMode() &&
Chris@43: 			!m_viewManager->getSelections().empty());
Chris@43: 
Chris@43:     static float **chunkBufferPtrs = 0;
Chris@43:     static size_t chunkBufferPtrCount = 0;
Chris@43:     size_t channels = getTargetChannelCount();
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     if (chunkBufferPtrCount < channels) {
Chris@43: 	if (chunkBufferPtrs) delete[] chunkBufferPtrs;
Chris@43: 	chunkBufferPtrs = new float *[channels];
Chris@43: 	chunkBufferPtrCount = channels;
Chris@43:     }
Chris@43: 
Chris@43:     for (size_t c = 0; c < channels; ++c) {
Chris@43: 	chunkBufferPtrs[c] = buffers[c];
Chris@43:     }
Chris@43: 
Chris@43:     while (processed < count) {
Chris@43: 	
Chris@43: 	chunkSize = count - processed;
Chris@43: 	nextChunkStart = chunkStart + chunkSize;
Chris@43: 	selectionSize = 0;
Chris@43: 
Chris@43: 	size_t fadeIn = 0, fadeOut = 0;
Chris@43: 
Chris@43: 	if (constrained) {
Chris@60: 
Chris@60:             size_t rChunkStart =
Chris@60:                 m_viewManager->alignPlaybackFrameToReference(chunkStart);
Chris@43: 	    
Chris@43: 	    Selection selection =
Chris@60: 		m_viewManager->getContainingSelection(rChunkStart, true);
Chris@43: 	    
Chris@43: 	    if (selection.isEmpty()) {
Chris@43: 		if (looping) {
Chris@43: 		    selection = *m_viewManager->getSelections().begin();
Chris@60: 		    chunkStart = m_viewManager->alignReferenceToPlaybackFrame
Chris@60:                         (selection.getStartFrame());
Chris@43: 		    fadeIn = 50;
Chris@43: 		}
Chris@43: 	    }
Chris@43: 
Chris@43: 	    if (selection.isEmpty()) {
Chris@43: 
Chris@43: 		chunkSize = 0;
Chris@43: 		nextChunkStart = chunkStart;
Chris@43: 
Chris@43: 	    } else {
Chris@43: 
Chris@60:                 size_t sf = m_viewManager->alignReferenceToPlaybackFrame
Chris@60:                     (selection.getStartFrame());
Chris@60:                 size_t ef = m_viewManager->alignReferenceToPlaybackFrame
Chris@60:                     (selection.getEndFrame());
Chris@43: 
Chris@60: 		selectionSize = ef - sf;
Chris@60: 
Chris@60: 		if (chunkStart < sf) {
Chris@60: 		    chunkStart = sf;
Chris@43: 		    fadeIn = 50;
Chris@43: 		}
Chris@43: 
Chris@43: 		nextChunkStart = chunkStart + chunkSize;
Chris@43: 
Chris@60: 		if (nextChunkStart >= ef) {
Chris@60: 		    nextChunkStart = ef;
Chris@43: 		    fadeOut = 50;
Chris@43: 		}
Chris@43: 
Chris@43: 		chunkSize = nextChunkStart - chunkStart;
Chris@43: 	    }
Chris@43: 	
Chris@43: 	} else if (looping && m_lastModelEndFrame > 0) {
Chris@43: 
Chris@43: 	    if (chunkStart >= m_lastModelEndFrame) {
Chris@43: 		chunkStart = 0;
Chris@43: 	    }
Chris@43: 	    if (chunkSize > m_lastModelEndFrame - chunkStart) {
Chris@43: 		chunkSize = m_lastModelEndFrame - chunkStart;
Chris@43: 	    }
Chris@43: 	    nextChunkStart = chunkStart + chunkSize;
Chris@43: 	}
Chris@43: 	
Chris@43: //	std::cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << std::endl;
Chris@43: 
Chris@43: 	if (!chunkSize) {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 	    std::cout << "Ending selection playback at " << nextChunkStart << std::endl;
Chris@43: #endif
Chris@43: 	    // We need to maintain full buffers so that the other
Chris@43: 	    // thread can tell where it's got to in the playback -- so
Chris@43: 	    // return the full amount here
Chris@43: 	    frame = frame + count;
Chris@43: 	    return count;
Chris@43: 	}
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 	std::cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43: 	size_t got = 0;
Chris@43: 
Chris@43: 	if (selectionSize < 100) {
Chris@43: 	    fadeIn = 0;
Chris@43: 	    fadeOut = 0;
Chris@43: 	} else if (selectionSize < 300) {
Chris@43: 	    if (fadeIn > 0) fadeIn = 10;
Chris@43: 	    if (fadeOut > 0) fadeOut = 10;
Chris@43: 	}
Chris@43: 
Chris@43: 	if (fadeIn > 0) {
Chris@43: 	    if (processed * 2 < fadeIn) {
Chris@43: 		fadeIn = processed * 2;
Chris@43: 	    }
Chris@43: 	}
Chris@43: 
Chris@43: 	if (fadeOut > 0) {
Chris@43: 	    if ((count - processed - chunkSize) * 2 < fadeOut) {
Chris@43: 		fadeOut = (count - processed - chunkSize) * 2;
Chris@43: 	    }
Chris@43: 	}
Chris@43: 
Chris@43: 	for (std::set<Model *>::iterator mi = m_models.begin();
Chris@43: 	     mi != m_models.end(); ++mi) {
Chris@43: 	    
Chris@43: 	    got = m_audioGenerator->mixModel(*mi, chunkStart, 
Chris@43: 					     chunkSize, chunkBufferPtrs,
Chris@43: 					     fadeIn, fadeOut);
Chris@43: 	}
Chris@43: 
Chris@43: 	for (size_t c = 0; c < channels; ++c) {
Chris@43: 	    chunkBufferPtrs[c] += chunkSize;
Chris@43: 	}
Chris@43: 
Chris@43: 	processed += chunkSize;
Chris@43: 	chunkStart = nextChunkStart;
Chris@43:     }
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     frame = nextChunkStart;
Chris@43:     return processed;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::unifyRingBuffers()
Chris@43: {
Chris@43:     if (m_readBuffers == m_writeBuffers) return;
Chris@43: 
Chris@43:     // only unify if there will be something to read
Chris@43:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	if (wb) {
Chris@43: 	    if (wb->getReadSpace() < m_blockSize * 2) {
Chris@43: 		if ((m_writeBufferFill + m_blockSize * 2) < 
Chris@43: 		    m_lastModelEndFrame) {
Chris@43: 		    // OK, we don't have enough and there's more to
Chris@43: 		    // read -- don't unify until we can do better
Chris@43: 		    return;
Chris@43: 		}
Chris@43: 	    }
Chris@43: 	    break;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@43:     size_t rf = m_readBufferFill;
Chris@43:     RingBuffer<float> *rb = getReadRingBuffer(0);
Chris@43:     if (rb) {
Chris@43: 	size_t rs = rb->getReadSpace();
Chris@43: 	//!!! incorrect when in non-contiguous selection, see comments elsewhere
Chris@43: //	std::cout << "rs = " << rs << std::endl;
Chris@43: 	if (rs < rf) rf -= rs;
Chris@43: 	else rf = 0;
Chris@43:     }
Chris@43:     
Chris@43:     //std::cout << "m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << std::endl;
Chris@43: 
Chris@43:     size_t wf = m_writeBufferFill;
Chris@43:     size_t skip = 0;
Chris@43:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	if (wb) {
Chris@43: 	    if (c == 0) {
Chris@43: 		
Chris@43: 		size_t wrs = wb->getReadSpace();
Chris@43: //		std::cout << "wrs = " << wrs << std::endl;
Chris@43: 
Chris@43: 		if (wrs < wf) wf -= wrs;
Chris@43: 		else wf = 0;
Chris@43: //		std::cout << "wf = " << wf << std::endl;
Chris@43: 		
Chris@43: 		if (wf < rf) skip = rf - wf;
Chris@43: 		if (skip == 0) break;
Chris@43: 	    }
Chris@43: 
Chris@43: //	    std::cout << "skipping " << skip << std::endl;
Chris@43: 	    wb->skip(skip);
Chris@43: 	}
Chris@43:     }
Chris@43: 		    
Chris@43:     m_bufferScavenger.claim(m_readBuffers);
Chris@43:     m_readBuffers = m_writeBuffers;
Chris@43:     m_readBufferFill = m_writeBufferFill;
Chris@43: //    std::cout << "unified" << std::endl;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::FillThread::run()
Chris@43: {
Chris@43:     AudioCallbackPlaySource &s(m_source);
Chris@43:     
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:     std::cout << "AudioCallbackPlaySourceFillThread starting" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43:     s.m_mutex.lock();
Chris@43: 
Chris@43:     bool previouslyPlaying = s.m_playing;
Chris@43:     bool work = false;
Chris@43: 
Chris@43:     while (!s.m_exiting) {
Chris@43: 
Chris@43: 	s.unifyRingBuffers();
Chris@43: 	s.m_bufferScavenger.scavenge();
Chris@43:         s.m_pluginScavenger.scavenge();
Chris@43: 
Chris@43: 	if (work && s.m_playing && s.getSourceSampleRate()) {
Chris@43: 	    
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 	    std::cout << "AudioCallbackPlaySourceFillThread: not waiting" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43: 	    s.m_mutex.unlock();
Chris@43: 	    s.m_mutex.lock();
Chris@43: 
Chris@43: 	} else {
Chris@43: 	    
Chris@43: 	    float ms = 100;
Chris@43: 	    if (s.getSourceSampleRate() > 0) {
Chris@43: 		ms = float(m_ringBufferSize) / float(s.getSourceSampleRate()) * 1000.0;
Chris@43: 	    }
Chris@43: 	    
Chris@43: 	    if (s.m_playing) ms /= 10;
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43:             if (!s.m_playing) std::cout << std::endl;
Chris@43: 	    std::cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << std::endl;
Chris@43: #endif
Chris@43: 	    
Chris@43: 	    s.m_condition.wait(&s.m_mutex, size_t(ms));
Chris@43: 	}
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 	std::cout << "AudioCallbackPlaySourceFillThread: awoken" << std::endl;
Chris@43: #endif
Chris@43: 
Chris@43: 	work = false;
Chris@43: 
Chris@43: 	if (!s.getSourceSampleRate()) continue;
Chris@43: 
Chris@43: 	bool playing = s.m_playing;
Chris@43: 
Chris@43: 	if (playing && !previouslyPlaying) {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@43: 	    std::cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << std::endl;
Chris@43: #endif
Chris@43: 	    for (size_t c = 0; c < s.getTargetChannelCount(); ++c) {
Chris@43: 		RingBuffer<float> *rb = s.getReadRingBuffer(c);
Chris@43: 		if (rb) rb->reset();
Chris@43: 	    }
Chris@43: 	}
Chris@43: 	previouslyPlaying = playing;
Chris@43: 
Chris@43: 	work = s.fillBuffers();
Chris@43:     }
Chris@43: 
Chris@43:     s.m_mutex.unlock();
Chris@43: }
Chris@43: