lbajardsilogic@0: /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*-  vi:set ts=8 sts=4 sw=4: */
lbajardsilogic@0: 
lbajardsilogic@0: /*
lbajardsilogic@0:     Sonic Visualiser
lbajardsilogic@0:     An audio file viewer and annotation editor.
lbajardsilogic@0:     Centre for Digital Music, Queen Mary, University of London.
lbajardsilogic@0:     This file copyright 2006 Chris Cannam and QMUL.
ivand_qmul@125:     +
lbajardsilogic@0:     This program is free software; you can redistribute it and/or
lbajardsilogic@0:     modify it under the terms of the GNU General Public License as
lbajardsilogic@0:     published by the Free Software Foundation; either version 2 of the
lbajardsilogic@0:     License, or (at your option) any later version.  See the file
lbajardsilogic@0:     COPYING included with this distribution for more information.
lbajardsilogic@0: */
lbajardsilogic@0: 
lbajardsilogic@0: #include "AudioCallbackPlaySource.h"
lbajardsilogic@0: 
lbajardsilogic@0: #include "AudioGenerator.h"
lbajardsilogic@0: 
lbajardsilogic@0: #include "data/model/Model.h"
lbajardsilogic@0: #include "view/ViewManager.h"
lbajardsilogic@0: #include "base/PlayParameterRepository.h"
lbajardsilogic@0: #include "base/Preferences.h"
lbajardsilogic@0: #include "data/model/DenseTimeValueModel.h"
lbajardsilogic@0: #include "data/model/WaveFileModel.h"
lbajardsilogic@0: #include "data/model/SparseOneDimensionalModel.h"
lbajardsilogic@0: #include "plugin/RealTimePluginInstance.h"
lbajardsilogic@0: #include "PhaseVocoderTimeStretcher.h"
lbajardsilogic@0: 
lbajardsilogic@0: #include <iostream>
lbajardsilogic@0: #include <cassert>
lbajardsilogic@0: 
lbajardsilogic@0: //#define DEBUG_AUDIO_PLAY_SOURCE 1
lbajardsilogic@0: //#define DEBUG_AUDIO_PLAY_SOURCE_PLAYING 1
lbajardsilogic@0: 
lbajardsilogic@199: const size_t AudioCallbackPlaySource::m_ringBufferSize = 131071;
lbajardsilogic@199: //const size_t AudioCallbackPlaySource::m_ringBufferSize = 1764000;
lbajardsilogic@0: 
lbajardsilogic@0: AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManager *manager) :
lbajardsilogic@0:     m_viewManager(manager),
lbajardsilogic@0:     m_audioGenerator(new AudioGenerator()),
lbajardsilogic@0:     m_readBuffers(0),
lbajardsilogic@0:     m_writeBuffers(0),
lbajardsilogic@0:     m_readBufferFill(0),
lbajardsilogic@0:     m_writeBufferFill(0),
lbajardsilogic@0:     m_bufferScavenger(1),
lbajardsilogic@0:     m_sourceChannelCount(0),
lbajardsilogic@0:     m_blockSize(1024),
lbajardsilogic@82: 	m_sourceSampleRate(0),
lbajardsilogic@0:     m_targetSampleRate(0),
lbajardsilogic@0:     m_playLatency(0),
lbajardsilogic@0:     m_playing(false),
lbajardsilogic@0:     m_exiting(false),
lbajardsilogic@0:     m_lastModelEndFrame(0),
lbajardsilogic@0:     m_outputLeft(0.0),
lbajardsilogic@0:     m_outputRight(0.0),
lbajardsilogic@0:     m_auditioningPlugin(0),
lbajardsilogic@0:     m_auditioningPluginBypassed(false),
lbajardsilogic@0:     m_timeStretcher(0),
lbajardsilogic@0:     m_fillThread(0),
lbajardsilogic@0:     m_converter(0),
lbajardsilogic@0:     m_crapConverter(0),
lbajardsilogic@79:     m_resampleQuality(Preferences::getInstance()->getResampleQuality()),
lbajardsilogic@79: 	m_filterStack(0)
lbajardsilogic@0: {
lbajardsilogic@0:     m_viewManager->setAudioPlaySource(this);
lbajardsilogic@0: 
lbajardsilogic@0:     connect(m_viewManager, SIGNAL(selectionChanged()),
lbajardsilogic@0: 	    this, SLOT(selectionChanged()));
lbajardsilogic@0:     connect(m_viewManager, SIGNAL(playLoopModeChanged()),
lbajardsilogic@0: 	    this, SLOT(playLoopModeChanged()));
lbajardsilogic@0:     connect(m_viewManager, SIGNAL(playSelectionModeChanged()),
lbajardsilogic@0: 	    this, SLOT(playSelectionModeChanged()));
lbajardsilogic@0: 
lbajardsilogic@0:     connect(PlayParameterRepository::getInstance(),
lbajardsilogic@0: 	    SIGNAL(playParametersChanged(PlayParameters *)),
lbajardsilogic@0: 	    this, SLOT(playParametersChanged(PlayParameters *)));
lbajardsilogic@0: 
lbajardsilogic@0:     connect(Preferences::getInstance(),
lbajardsilogic@0:             SIGNAL(propertyChanged(PropertyContainer::PropertyName)),
lbajardsilogic@0:             this, SLOT(preferenceChanged(PropertyContainer::PropertyName)));
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: AudioCallbackPlaySource::~AudioCallbackPlaySource()
lbajardsilogic@0: {
lbajardsilogic@0:     m_exiting = true;
lbajardsilogic@0: 
lbajardsilogic@0:     if (m_fillThread) {
lbajardsilogic@0: 	m_condition.wakeAll();
lbajardsilogic@0: 	m_fillThread->wait();
lbajardsilogic@0: 	delete m_fillThread;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     clearModels();
lbajardsilogic@0:     
lbajardsilogic@0:     if (m_readBuffers != m_writeBuffers) {
lbajardsilogic@180: 		delete m_readBuffers;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     delete m_writeBuffers;
lbajardsilogic@0: 
lbajardsilogic@0:     delete m_audioGenerator;
lbajardsilogic@0: 
lbajardsilogic@0:     m_bufferScavenger.scavenge(true);
lbajardsilogic@0:     m_pluginScavenger.scavenge(true);
lbajardsilogic@0:     m_timeStretcherScavenger.scavenge(true);
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::addModel(Model *model)
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_models.find(model) != m_models.end()) return;
lbajardsilogic@0: 
lbajardsilogic@0:     bool canPlay = m_audioGenerator->addModel(model);
lbajardsilogic@0: 
lbajardsilogic@0:     m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@0:     m_models.insert(model);
lbajardsilogic@0:     if (model->getEndFrame() > m_lastModelEndFrame) {
lbajardsilogic@0: 	m_lastModelEndFrame = model->getEndFrame();
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     bool buffersChanged = false, srChanged = false;
lbajardsilogic@0: 
lbajardsilogic@0:     size_t modelChannels = 1;
lbajardsilogic@0:     DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
lbajardsilogic@0:     if (dtvm) modelChannels = dtvm->getChannelCount();
lbajardsilogic@0:     if (modelChannels > m_sourceChannelCount) {
lbajardsilogic@0: 	m_sourceChannelCount = modelChannels;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "Adding model with " << modelChannels << " channels " << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     if (m_sourceSampleRate == 0) {
lbajardsilogic@0: 
lbajardsilogic@0: 	m_sourceSampleRate = model->getSampleRate();
lbajardsilogic@0: 	srChanged = true;
lbajardsilogic@0: 
lbajardsilogic@0:     } else if (model->getSampleRate() != m_sourceSampleRate) {
lbajardsilogic@0: 
lbajardsilogic@0:         // If this is a dense time-value model and we have no other, we
lbajardsilogic@0:         // can just switch to this model's sample rate
lbajardsilogic@0: 
lbajardsilogic@0:         if (dtvm) {
lbajardsilogic@0: 
lbajardsilogic@0:             bool conflicting = false;
lbajardsilogic@0: 
lbajardsilogic@0:             for (std::set<Model *>::const_iterator i = m_models.begin();
lbajardsilogic@0:                  i != m_models.end(); ++i) {
lbajardsilogic@0:                 // Only wave file models can be considered conflicting --
lbajardsilogic@0:                 // writable wave file models are derived and we shouldn't
lbajardsilogic@0:                 // take their rates into account.  Also, don't give any
lbajardsilogic@0:                 // particular weight to a file that's already playing at
lbajardsilogic@0:                 // the wrong rate anyway
lbajardsilogic@0:                 WaveFileModel *wfm = dynamic_cast<WaveFileModel *>(*i);
lbajardsilogic@0:                 if (wfm && wfm != dtvm &&
lbajardsilogic@0:                     wfm->getSampleRate() != model->getSampleRate() &&
lbajardsilogic@0:                     wfm->getSampleRate() == m_sourceSampleRate) {
lbajardsilogic@0:                     std::cerr << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << std::endl;
lbajardsilogic@0:                     conflicting = true;
lbajardsilogic@0:                     break;
lbajardsilogic@0:                 }
lbajardsilogic@0:             }
lbajardsilogic@0: 
lbajardsilogic@0:             if (conflicting) {
lbajardsilogic@0: 
lbajardsilogic@0:                 std::cerr << "AudioCallbackPlaySource::addModel: ERROR: "
lbajardsilogic@0:                           << "New model sample rate does not match" << std::endl
lbajardsilogic@0:                           << "existing model(s) (new " << model->getSampleRate()
lbajardsilogic@0:                           << " vs " << m_sourceSampleRate
lbajardsilogic@0:                           << "), playback will be wrong"
lbajardsilogic@0:                           << std::endl;
lbajardsilogic@0:                 
lbajardsilogic@0:                 emit sampleRateMismatch(model->getSampleRate(),
lbajardsilogic@0:                                         m_sourceSampleRate,
lbajardsilogic@0:                                         false);
lbajardsilogic@0:             } else {
lbajardsilogic@0:                 m_sourceSampleRate = model->getSampleRate();
lbajardsilogic@0:                 srChanged = true;
lbajardsilogic@0:             }
lbajardsilogic@0:         }
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (!m_writeBuffers || (m_writeBuffers->size() < getTargetChannelCount())) {
lbajardsilogic@0: 	clearRingBuffers(true, getTargetChannelCount());
lbajardsilogic@0: 	buffersChanged = true;
lbajardsilogic@0:     } else {
lbajardsilogic@0: 	if (canPlay) clearRingBuffers(true);
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (buffersChanged || srChanged) {
lbajardsilogic@0: 	if (m_converter) {
lbajardsilogic@0: 	    src_delete(m_converter);
lbajardsilogic@0:             src_delete(m_crapConverter);
lbajardsilogic@0: 	    m_converter = 0;
lbajardsilogic@0:             m_crapConverter = 0;
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     m_mutex.unlock();
lbajardsilogic@0: 
lbajardsilogic@0:     m_audioGenerator->setTargetChannelCount(getTargetChannelCount());
lbajardsilogic@0: 
lbajardsilogic@0:     if (!m_fillThread) {
lbajardsilogic@0: 	m_fillThread = new FillThread(*this);
lbajardsilogic@0: 	m_fillThread->start();
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s) -- emitting modelReplaced" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     if (buffersChanged || srChanged) {
lbajardsilogic@0: 	emit modelReplaced();
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     m_condition.wakeAll();
lbajardsilogic@84: 
lbajardsilogic@84: 	m_filterStack->setSourceChannelCount(getTargetChannelCount());
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::removeModel(Model *model)
lbajardsilogic@0: {
lbajardsilogic@0:     m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     m_models.erase(model);
lbajardsilogic@0: 
lbajardsilogic@0:     if (m_models.empty()) {
lbajardsilogic@0: 	if (m_converter) {
lbajardsilogic@0: 	    src_delete(m_converter);
lbajardsilogic@0:             src_delete(m_crapConverter);
lbajardsilogic@0: 	    m_converter = 0;
lbajardsilogic@0:             m_crapConverter = 0;
lbajardsilogic@0: 	}
lbajardsilogic@0: 	m_sourceSampleRate = 0;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t lastEnd = 0;
lbajardsilogic@0:     for (std::set<Model *>::const_iterator i = m_models.begin();
lbajardsilogic@0: 	 i != m_models.end(); ++i) {
lbajardsilogic@0: //	std::cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << std::endl;
lbajardsilogic@0: 	if ((*i)->getEndFrame() > lastEnd) lastEnd = (*i)->getEndFrame();
lbajardsilogic@0: //	std::cout << "(done, lastEnd now " << lastEnd << ")" << std::endl;
lbajardsilogic@0:     }
lbajardsilogic@0:     m_lastModelEndFrame = lastEnd;
lbajardsilogic@0: 
lbajardsilogic@0:     m_mutex.unlock();
lbajardsilogic@0: 
lbajardsilogic@0:     m_audioGenerator->removeModel(model);
lbajardsilogic@0: 
lbajardsilogic@0:     clearRingBuffers();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::clearModels()
lbajardsilogic@0: {
lbajardsilogic@0:     m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "AudioCallbackPlaySource::clearModels()" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     m_models.clear();
lbajardsilogic@0: 
lbajardsilogic@0:     if (m_converter) {
lbajardsilogic@0: 	src_delete(m_converter);
lbajardsilogic@0:         src_delete(m_crapConverter);
lbajardsilogic@0: 	m_converter = 0;
lbajardsilogic@0:         m_crapConverter = 0;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     m_lastModelEndFrame = 0;
lbajardsilogic@0: 
lbajardsilogic@0:     m_sourceSampleRate = 0;
lbajardsilogic@0: 
lbajardsilogic@0:     m_mutex.unlock();
lbajardsilogic@0: 
lbajardsilogic@0:     m_audioGenerator->clearModels();
lbajardsilogic@0: }    
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::clearRingBuffers(bool haveLock, size_t count)
lbajardsilogic@0: {
lbajardsilogic@0:     if (!haveLock) m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@0:     if (count == 0) {
lbajardsilogic@0: 	if (m_writeBuffers) count = m_writeBuffers->size();
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t sf = m_readBufferFill;
lbajardsilogic@0:     RingBuffer<float> *rb = getReadRingBuffer(0);
lbajardsilogic@0:     if (rb) {
lbajardsilogic@0: 	//!!! This is incorrect if we're in a non-contiguous selection
lbajardsilogic@0: 	//Same goes for all related code (subtracting the read space
lbajardsilogic@0: 	//from the fill frame to try to establish where the effective
lbajardsilogic@0: 	//pre-resample/timestretch read pointer is)
lbajardsilogic@0: 	size_t rs = rb->getReadSpace();
lbajardsilogic@0: 	if (rs < sf) sf -= rs;
lbajardsilogic@0: 	else sf = 0;
lbajardsilogic@0:     }
lbajardsilogic@0:     m_writeBufferFill = sf;
lbajardsilogic@0: 
lbajardsilogic@0:     if (m_readBuffers != m_writeBuffers) {
lbajardsilogic@180: 		delete m_writeBuffers;
lbajardsilogic@180: 		m_writeBuffers = 0;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     m_writeBuffers = new RingBufferVector;
lbajardsilogic@0: 
lbajardsilogic@0:     for (size_t i = 0; i < count; ++i) {
lbajardsilogic@0: 	m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize));
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0: //    std::cout << "AudioCallbackPlaySource::clearRingBuffers: Created "
lbajardsilogic@0: //	      << count << " write buffers" << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0:     if (!haveLock) {
lbajardsilogic@0: 	m_mutex.unlock();
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::play(size_t startFrame)
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_viewManager->getPlaySelectionMode() &&
lbajardsilogic@0: 	!m_viewManager->getSelections().empty()) {
lbajardsilogic@0: 	MultiSelection::SelectionList selections = m_viewManager->getSelections();
lbajardsilogic@0: 	MultiSelection::SelectionList::iterator i = selections.begin();
lbajardsilogic@0: 	if (i != selections.end()) {
lbajardsilogic@0: 	    if (startFrame < i->getStartFrame()) {
lbajardsilogic@0: 		startFrame = i->getStartFrame();
lbajardsilogic@0: 	    } else {
lbajardsilogic@0: 		MultiSelection::SelectionList::iterator j = selections.end();
lbajardsilogic@0: 		--j;
lbajardsilogic@0: 		if (startFrame >= j->getEndFrame()) {
lbajardsilogic@0: 		    startFrame = i->getStartFrame();
lbajardsilogic@0: 		}
lbajardsilogic@0: 	    }
lbajardsilogic@0: 	}
lbajardsilogic@0:     } else {
lbajardsilogic@0: 	if (startFrame >= m_lastModelEndFrame) {
lbajardsilogic@0: 	    startFrame = 0;
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     // The fill thread will automatically empty its buffers before
lbajardsilogic@0:     // starting again if we have not so far been playing, but not if
lbajardsilogic@0:     // we're just re-seeking.
lbajardsilogic@0: 
lbajardsilogic@0:     m_mutex.lock();
lbajardsilogic@0:     if (m_playing) {
lbajardsilogic@0: 	m_readBufferFill = m_writeBufferFill = startFrame;
lbajardsilogic@0: 	if (m_readBuffers) {
lbajardsilogic@0: 	    for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@0: 		RingBuffer<float> *rb = getReadRingBuffer(c);
lbajardsilogic@0: 		if (rb) rb->reset();
lbajardsilogic@0: 	    }
lbajardsilogic@0: 	}
lbajardsilogic@0: 	if (m_converter) src_reset(m_converter);
lbajardsilogic@0:         if (m_crapConverter) src_reset(m_crapConverter);
lbajardsilogic@0:     } else {
lbajardsilogic@0: 	if (m_converter) src_reset(m_converter);
lbajardsilogic@0:         if (m_crapConverter) src_reset(m_crapConverter);
lbajardsilogic@0: 	m_readBufferFill = m_writeBufferFill = startFrame;
lbajardsilogic@0:     }
lbajardsilogic@0:     m_mutex.unlock();
lbajardsilogic@0: 
lbajardsilogic@0:     m_audioGenerator->reset();
lbajardsilogic@0: 
lbajardsilogic@0:     bool changed = !m_playing;
lbajardsilogic@0:     m_playing = true;
lbajardsilogic@0:     m_condition.wakeAll();
lbajardsilogic@0:     if (changed) emit playStatusChanged(m_playing);
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::stop()
lbajardsilogic@0: {
lbajardsilogic@0:     bool changed = m_playing;
lbajardsilogic@0:     m_playing = false;
lbajardsilogic@0:     m_condition.wakeAll();
lbajardsilogic@0:     if (changed) emit playStatusChanged(m_playing);
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::selectionChanged()
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_viewManager->getPlaySelectionMode()) {
lbajardsilogic@0: 	clearRingBuffers();
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::playLoopModeChanged()
lbajardsilogic@0: {
lbajardsilogic@0:     clearRingBuffers();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::playSelectionModeChanged()
lbajardsilogic@0: {
lbajardsilogic@0:     if (!m_viewManager->getSelections().empty()) {
lbajardsilogic@0: 	clearRingBuffers();
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::playParametersChanged(PlayParameters *)
lbajardsilogic@0: {
lbajardsilogic@0:     clearRingBuffers();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::preferenceChanged(PropertyContainer::PropertyName n)
lbajardsilogic@0: {
lbajardsilogic@0:     if (n == "Resample Quality") {
lbajardsilogic@0:         setResampleQuality(Preferences::getInstance()->getResampleQuality());
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::audioProcessingOverload()
lbajardsilogic@0: {
lbajardsilogic@0:     RealTimePluginInstance *ap = m_auditioningPlugin;
lbajardsilogic@0:     if (ap && m_playing && !m_auditioningPluginBypassed) {
lbajardsilogic@0:         m_auditioningPluginBypassed = true;
lbajardsilogic@0:         emit audioOverloadPluginDisabled();
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setTargetBlockSize(size_t size)
lbajardsilogic@0: {
lbajardsilogic@0: //    std::cout << "AudioCallbackPlaySource::setTargetBlockSize() -> " << size << std::endl;
lbajardsilogic@0:     assert(size < m_ringBufferSize);
lbajardsilogic@0:     m_blockSize = size;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getTargetBlockSize() const
lbajardsilogic@0: {
lbajardsilogic@0: //    std::cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << std::endl;
lbajardsilogic@0:     return m_blockSize;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setTargetPlayLatency(size_t latency)
lbajardsilogic@0: {
lbajardsilogic@0:     m_playLatency = latency;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getTargetPlayLatency() const
lbajardsilogic@0: {
lbajardsilogic@0:     return m_playLatency;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getCurrentPlayingFrame()
lbajardsilogic@0: {
lbajardsilogic@0:     bool resample = false;
lbajardsilogic@0:     double ratio = 1.0;
lbajardsilogic@0: 
lbajardsilogic@0:     if (getSourceSampleRate() != getTargetSampleRate()) {
lbajardsilogic@0: 	resample = true;
lbajardsilogic@0: 	ratio = double(getSourceSampleRate()) / double(getTargetSampleRate());
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t readSpace = 0;
lbajardsilogic@0:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@0: 	RingBuffer<float> *rb = getReadRingBuffer(c);
lbajardsilogic@0: 	if (rb) {
lbajardsilogic@0: 	    size_t spaceHere = rb->getReadSpace();
lbajardsilogic@0: 	    if (c == 0 || spaceHere < readSpace) readSpace = spaceHere;
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (resample) {
lbajardsilogic@0: 	readSpace = size_t(readSpace * ratio + 0.1);
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t latency = m_playLatency;
lbajardsilogic@0:     if (resample) latency = size_t(m_playLatency * ratio + 0.1);
lbajardsilogic@0: 
lbajardsilogic@0:     PhaseVocoderTimeStretcher *timeStretcher = m_timeStretcher;
lbajardsilogic@0:     if (timeStretcher) {
lbajardsilogic@0: 	latency += timeStretcher->getProcessingLatency();
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     latency += readSpace;
lbajardsilogic@0:     size_t bufferedFrame = m_readBufferFill;
lbajardsilogic@0: 
lbajardsilogic@0:     bool looping = m_viewManager->getPlayLoopMode();
lbajardsilogic@0:     bool constrained = (m_viewManager->getPlaySelectionMode() &&
lbajardsilogic@0: 			!m_viewManager->getSelections().empty());
lbajardsilogic@0: 
lbajardsilogic@0:     size_t framePlaying = bufferedFrame;
lbajardsilogic@0: 
lbajardsilogic@0:     if (looping && !constrained) {
lbajardsilogic@0: 	while (framePlaying < latency) framePlaying += m_lastModelEndFrame;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (framePlaying > latency) framePlaying -= latency;
lbajardsilogic@0:     else framePlaying = 0;
lbajardsilogic@0: 
lbajardsilogic@0:     if (!constrained) {
lbajardsilogic@0: 	if (!looping && framePlaying > m_lastModelEndFrame) {
lbajardsilogic@0: 	    framePlaying = m_lastModelEndFrame;
lbajardsilogic@0: 	    stop();
lbajardsilogic@0: 	}
lbajardsilogic@0: 	return framePlaying;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     MultiSelection::SelectionList selections = m_viewManager->getSelections();
lbajardsilogic@0:     MultiSelection::SelectionList::const_iterator i;
lbajardsilogic@0: 
lbajardsilogic@0: //    i = selections.begin();
lbajardsilogic@0: //    size_t rangeStart = i->getStartFrame();
lbajardsilogic@0: 
lbajardsilogic@0:     i = selections.end();
lbajardsilogic@0:     --i;
lbajardsilogic@0:     size_t rangeEnd = i->getEndFrame();
lbajardsilogic@0: 
lbajardsilogic@0:     for (i = selections.begin(); i != selections.end(); ++i) {
lbajardsilogic@0: 	if (i->contains(bufferedFrame)) break;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t f = bufferedFrame;
lbajardsilogic@0: 
lbajardsilogic@0: //    std::cout << "getCurrentPlayingFrame: f=" << f << ", latency=" << latency << ", rangeEnd=" << rangeEnd << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0:     if (i == selections.end()) {
lbajardsilogic@0: 	--i;
lbajardsilogic@0: 	if (i->getEndFrame() + latency < f) {
lbajardsilogic@0: //    std::cout << "framePlaying = " << framePlaying << ", rangeEnd = " << rangeEnd << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0: 	    if (!looping && (framePlaying > rangeEnd)) {
lbajardsilogic@0: //		std::cout << "STOPPING" << std::endl;
lbajardsilogic@0: 		stop();
lbajardsilogic@0: 		return rangeEnd;
lbajardsilogic@0: 	    } else {
lbajardsilogic@0: 		return framePlaying;
lbajardsilogic@0: 	    }
lbajardsilogic@0: 	} else {
lbajardsilogic@0: //	    std::cout << "latency <- " << latency << "-(" << f << "-" << i->getEndFrame() << ")" << std::endl;
lbajardsilogic@0: 	    latency -= (f - i->getEndFrame());
lbajardsilogic@0: 	    f = i->getEndFrame();
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0: //    std::cout << "i=(" << i->getStartFrame() << "," << i->getEndFrame() << ") f=" << f << ", latency=" << latency << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0:     while (latency > 0) {
lbajardsilogic@0: 	size_t offset = f - i->getStartFrame();
lbajardsilogic@0: 	if (offset >= latency) {
lbajardsilogic@0: 	    if (f > latency) {
lbajardsilogic@0: 		framePlaying = f - latency;
lbajardsilogic@0: 	    } else {
lbajardsilogic@0: 		framePlaying = 0;
lbajardsilogic@0: 	    }
lbajardsilogic@0: 	    break;
lbajardsilogic@0: 	} else {
lbajardsilogic@0: 	    if (i == selections.begin()) {
lbajardsilogic@0: 		if (looping) {
lbajardsilogic@0: 		    i = selections.end();
lbajardsilogic@0: 		}
lbajardsilogic@0: 	    }
lbajardsilogic@0: 	    latency -= offset;
lbajardsilogic@0: 	    --i;
lbajardsilogic@0: 	    f = i->getEndFrame();
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     return framePlaying;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setOutputLevels(float left, float right)
lbajardsilogic@0: {
lbajardsilogic@0:     m_outputLeft = left;
lbajardsilogic@0:     m_outputRight = right;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: bool
lbajardsilogic@0: AudioCallbackPlaySource::getOutputLevels(float &left, float &right)
lbajardsilogic@0: {
lbajardsilogic@0:     left = m_outputLeft;
lbajardsilogic@0:     right = m_outputRight;
lbajardsilogic@0:     return true;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setTargetSampleRate(size_t sr)
lbajardsilogic@0: {
lbajardsilogic@0:     m_targetSampleRate = sr;
lbajardsilogic@0:     initialiseConverter();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::initialiseConverter()
lbajardsilogic@0: {
lbajardsilogic@0:     m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@0:     if (m_converter) {
lbajardsilogic@0:         src_delete(m_converter);
lbajardsilogic@0:         src_delete(m_crapConverter);
lbajardsilogic@0:         m_converter = 0;
lbajardsilogic@0:         m_crapConverter = 0;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (getSourceSampleRate() != getTargetSampleRate()) {
lbajardsilogic@0: 
lbajardsilogic@0: 	int err = 0;
lbajardsilogic@0: 
lbajardsilogic@0: 	m_converter = src_new(m_resampleQuality == 2 ? SRC_SINC_BEST_QUALITY :
lbajardsilogic@0:                               m_resampleQuality == 1 ? SRC_SINC_MEDIUM_QUALITY :
lbajardsilogic@0:                               m_resampleQuality == 0 ? SRC_SINC_FASTEST :
lbajardsilogic@0:                                                        SRC_SINC_MEDIUM_QUALITY,
lbajardsilogic@0: 			      getTargetChannelCount(), &err);
lbajardsilogic@0: 
lbajardsilogic@0:         if (m_converter) {
lbajardsilogic@0:             m_crapConverter = src_new(SRC_LINEAR,
lbajardsilogic@0:                                       getTargetChannelCount(),
lbajardsilogic@0:                                       &err);
lbajardsilogic@0:         }
lbajardsilogic@0: 
lbajardsilogic@0: 	if (!m_converter || !m_crapConverter) {
lbajardsilogic@0: 	    std::cerr
lbajardsilogic@0: 		<< "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: "
lbajardsilogic@0: 		<< src_strerror(err) << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0:             if (m_converter) {
lbajardsilogic@0:                 src_delete(m_converter);
lbajardsilogic@0:                 m_converter = 0;
lbajardsilogic@0:             } 
lbajardsilogic@0: 
lbajardsilogic@0:             if (m_crapConverter) {
lbajardsilogic@0:                 src_delete(m_crapConverter);
lbajardsilogic@0:                 m_crapConverter = 0;
lbajardsilogic@0:             }
lbajardsilogic@0: 
lbajardsilogic@0:             m_mutex.unlock();
lbajardsilogic@0: 
lbajardsilogic@0:             emit sampleRateMismatch(getSourceSampleRate(),
lbajardsilogic@0:                                     getTargetSampleRate(),
lbajardsilogic@0:                                     false);
lbajardsilogic@0: 	} else {
lbajardsilogic@0: 
lbajardsilogic@0:             m_mutex.unlock();
lbajardsilogic@0: 
lbajardsilogic@0:             emit sampleRateMismatch(getSourceSampleRate(),
lbajardsilogic@0:                                     getTargetSampleRate(),
lbajardsilogic@0:                                     true);
lbajardsilogic@0:         }
lbajardsilogic@0:     } else {
lbajardsilogic@0:         m_mutex.unlock();
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setResampleQuality(int q)
lbajardsilogic@0: {
lbajardsilogic@0:     if (q == m_resampleQuality) return;
lbajardsilogic@0:     m_resampleQuality = q;
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cerr << "AudioCallbackPlaySource::setResampleQuality: setting to "
lbajardsilogic@0:               << m_resampleQuality << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     initialiseConverter();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setAuditioningPlugin(RealTimePluginInstance *plugin)
lbajardsilogic@0: {
lbajardsilogic@0:     RealTimePluginInstance *formerPlugin = m_auditioningPlugin;
lbajardsilogic@0:     m_auditioningPlugin = plugin;
lbajardsilogic@0:     m_auditioningPluginBypassed = false;
lbajardsilogic@0:     if (formerPlugin) m_pluginScavenger.claim(formerPlugin);
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getTargetSampleRate() const
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_targetSampleRate) return m_targetSampleRate;
lbajardsilogic@0:     else return getSourceSampleRate();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getSourceChannelCount() const
lbajardsilogic@0: {
lbajardsilogic@0:     return m_sourceChannelCount;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getTargetChannelCount() const
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_sourceChannelCount < 2) return 2;
lbajardsilogic@0:     return m_sourceChannelCount;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getSourceSampleRate() const
lbajardsilogic@0: {
lbajardsilogic@0:     return m_sourceSampleRate;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::setTimeStretch(float factor, bool sharpen, bool mono)
lbajardsilogic@0: {
lbajardsilogic@0:     // Avoid locks -- create, assign, mark old one for scavenging
lbajardsilogic@0:     // later (as a call to getSourceSamples may still be using it)
lbajardsilogic@0: 
lbajardsilogic@0:     PhaseVocoderTimeStretcher *existingStretcher = m_timeStretcher;
lbajardsilogic@0: 
lbajardsilogic@0:     size_t channels = getTargetChannelCount();
lbajardsilogic@0:     if (mono) channels = 1;
lbajardsilogic@0: 
lbajardsilogic@0:     if (existingStretcher &&
lbajardsilogic@0:         existingStretcher->getRatio() == factor &&
lbajardsilogic@0:         existingStretcher->getSharpening() == sharpen &&
lbajardsilogic@0:         existingStretcher->getChannelCount() == channels) {
lbajardsilogic@106: 		return;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (factor != 1) {
lbajardsilogic@0: 
lbajardsilogic@0:         if (existingStretcher &&
lbajardsilogic@0:             existingStretcher->getSharpening() == sharpen &&
lbajardsilogic@0:             existingStretcher->getChannelCount() == channels) {
lbajardsilogic@106: 			existingStretcher->setRatio(factor);
lbajardsilogic@106: 			return;
lbajardsilogic@0:         }
lbajardsilogic@0: 
lbajardsilogic@106: 		PhaseVocoderTimeStretcher *newStretcher = new PhaseVocoderTimeStretcher
lbajardsilogic@0: 	    (getTargetSampleRate(),
lbajardsilogic@0:              channels,
lbajardsilogic@0:              factor,
lbajardsilogic@0:              sharpen,
lbajardsilogic@0:              getTargetBlockSize());
lbajardsilogic@0: 
lbajardsilogic@106: 		m_timeStretcher = newStretcher;
lbajardsilogic@0: 
lbajardsilogic@0:     } else {
lbajardsilogic@106: 		m_timeStretcher = 0;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (existingStretcher) {
lbajardsilogic@106: 		m_timeStretcherScavenger.claim(existingStretcher);
lbajardsilogic@0:     }
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::getSourceSamples(size_t count, float **buffer)
lbajardsilogic@0: {
lbajardsilogic@0:     if (!m_playing) {
lbajardsilogic@105: 		for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
lbajardsilogic@105: 			for (size_t i = 0; i < count; ++i) {
lbajardsilogic@105: 				buffer[ch][i] = 0.0;
lbajardsilogic@105: 			}
lbajardsilogic@105: 		}
lbajardsilogic@105: 		return 0;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     // Ensure that all buffers have at least the amount of data we
lbajardsilogic@0:     // need -- else reduce the size of our requests correspondingly
lbajardsilogic@0: 
lbajardsilogic@0:     for (size_t ch = 0; ch < getTargetChannelCount(); ++ch) {
lbajardsilogic@0: 
lbajardsilogic@0:         RingBuffer<float> *rb = getReadRingBuffer(ch);
lbajardsilogic@0:         
lbajardsilogic@0:         if (!rb) {
lbajardsilogic@0:             std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
lbajardsilogic@0:                       << "No ring buffer available for channel " << ch
lbajardsilogic@0:                       << ", returning no data here" << std::endl;
lbajardsilogic@0:             count = 0;
lbajardsilogic@0:             break;
lbajardsilogic@0:         }
lbajardsilogic@0: 
lbajardsilogic@0:         size_t rs = rb->getReadSpace();
lbajardsilogic@0:         if (rs < count) {
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:             std::cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
lbajardsilogic@0:                       << "Ring buffer for channel " << ch << " has only "
lbajardsilogic@0:                       << rs << " (of " << count << ") samples available, "
lbajardsilogic@0:                       << "reducing request size" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0:             count = rs;
lbajardsilogic@0:         }
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (count == 0) return 0;
lbajardsilogic@0: 
lbajardsilogic@79: 	applyRealTimeFilters(count, buffer);
lbajardsilogic@79: 
lbajardsilogic@106: 	applyAuditioningEffect(count, buffer);
lbajardsilogic@106: 
lbajardsilogic@0:     m_condition.wakeAll();
lbajardsilogic@0: 
lbajardsilogic@0:     return count;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::applyAuditioningEffect(size_t count, float **buffers)
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_auditioningPluginBypassed) return;
lbajardsilogic@0:     RealTimePluginInstance *plugin = m_auditioningPlugin;
lbajardsilogic@0:     if (!plugin) return;
lbajardsilogic@0: 
lbajardsilogic@0:     if (plugin->getAudioInputCount() != getTargetChannelCount()) {
lbajardsilogic@0: //        std::cerr << "plugin input count " << plugin->getAudioInputCount() 
lbajardsilogic@0: //                  << " != our channel count " << getTargetChannelCount()
lbajardsilogic@0: //                  << std::endl;
lbajardsilogic@0:         return;
lbajardsilogic@0:     }
lbajardsilogic@0:     if (plugin->getAudioOutputCount() != getTargetChannelCount()) {
lbajardsilogic@0: //        std::cerr << "plugin output count " << plugin->getAudioOutputCount() 
lbajardsilogic@0: //                  << " != our channel count " << getTargetChannelCount()
lbajardsilogic@0: //                  << std::endl;
lbajardsilogic@0:         return;
lbajardsilogic@0:     }
lbajardsilogic@0:     if (plugin->getBufferSize() != count) {
lbajardsilogic@0: //        std::cerr << "plugin buffer size " << plugin->getBufferSize() 
lbajardsilogic@0: //                  << " != our block size " << count
lbajardsilogic@0: //                  << std::endl;
lbajardsilogic@0:         return;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     float **ib = plugin->getAudioInputBuffers();
lbajardsilogic@0:     float **ob = plugin->getAudioOutputBuffers();
lbajardsilogic@0: 
lbajardsilogic@0:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@0:         for (size_t i = 0; i < count; ++i) {
lbajardsilogic@0:             ib[c][i] = buffers[c][i];
lbajardsilogic@0:         }
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     plugin->run(Vamp::RealTime::zeroTime);
lbajardsilogic@0:     
lbajardsilogic@0:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@0:         for (size_t i = 0; i < count; ++i) {
lbajardsilogic@0:             buffers[c][i] = ob[c][i];
lbajardsilogic@0:         }
lbajardsilogic@0:     }
lbajardsilogic@0: }    
lbajardsilogic@0: 
lbajardsilogic@0: // Called from fill thread, m_playing true, mutex held
lbajardsilogic@0: bool
lbajardsilogic@0: AudioCallbackPlaySource::fillBuffers()
lbajardsilogic@0: {
lbajardsilogic@0:     static float *tmp = 0;
lbajardsilogic@0:     static size_t tmpSize = 0;
lbajardsilogic@0: 
lbajardsilogic@0:     size_t space = 0;
lbajardsilogic@0:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@106: 		RingBuffer<float> *wb = getWriteRingBuffer(c);
lbajardsilogic@106: 		if (wb) {
lbajardsilogic@106: 			size_t spaceHere = wb->getWriteSpace();
lbajardsilogic@106: 			if (c == 0 || spaceHere < space) space = spaceHere;
lbajardsilogic@106: 		}
lbajardsilogic@0:     }
lbajardsilogic@0:     
lbajardsilogic@0:     if (space == 0) return false;
lbajardsilogic@0: 
lbajardsilogic@0:     size_t f = m_writeBufferFill;
lbajardsilogic@0: 	
lbajardsilogic@0:     bool readWriteEqual = (m_readBuffers == m_writeBuffers);
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "buffered to " << f << " already" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     bool resample = (getSourceSampleRate() != getTargetSampleRate());
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     size_t channels = getTargetChannelCount();
lbajardsilogic@0: 
lbajardsilogic@0:     size_t orig = space;
lbajardsilogic@0:     size_t got = 0;
lbajardsilogic@0: 
lbajardsilogic@0:     static float **bufferPtrs = 0;
lbajardsilogic@0:     static size_t bufferPtrCount = 0;
lbajardsilogic@0: 
lbajardsilogic@0:     if (bufferPtrCount < channels) {
lbajardsilogic@106: 		if (bufferPtrs) delete[] bufferPtrs;
lbajardsilogic@106: 		bufferPtrs = new float *[channels];
lbajardsilogic@106: 		bufferPtrCount = channels;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t generatorBlockSize = m_audioGenerator->getBlockSize();
lbajardsilogic@0: 
lbajardsilogic@0:     if (resample && !m_converter) {
lbajardsilogic@106: 		static bool warned = false;
lbajardsilogic@106: 		if (!warned) {
lbajardsilogic@106: 			std::cerr << "WARNING: sample rates differ, but no converter available!" << std::endl;
lbajardsilogic@106: 			warned = true;
lbajardsilogic@106: 		}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     if (resample && m_converter) {
lbajardsilogic@0: 
lbajardsilogic@106: 		double ratio =
lbajardsilogic@106: 			double(getTargetSampleRate()) / double(getSourceSampleRate());
lbajardsilogic@106: 		orig = size_t(orig / ratio + 0.1);
lbajardsilogic@0: 
lbajardsilogic@106: 		// orig must be a multiple of generatorBlockSize
lbajardsilogic@106: 		orig = (orig / generatorBlockSize) * generatorBlockSize;
lbajardsilogic@106: 		if (orig == 0) return false;
lbajardsilogic@0: 
lbajardsilogic@191: 		size_t work = MAX(orig, space);
lbajardsilogic@0: 
lbajardsilogic@106: 		// We only allocate one buffer, but we use it in two halves.
lbajardsilogic@106: 		// We place the non-interleaved values in the second half of
lbajardsilogic@106: 		// the buffer (orig samples for channel 0, orig samples for
lbajardsilogic@106: 		// channel 1 etc), and then interleave them into the first
lbajardsilogic@106: 		// half of the buffer.  Then we resample back into the second
lbajardsilogic@106: 		// half (interleaved) and de-interleave the results back to
lbajardsilogic@106: 		// the start of the buffer for insertion into the ringbuffers.
lbajardsilogic@106: 		// What a faff -- especially as we've already de-interleaved
lbajardsilogic@106: 		// the audio data from the source file elsewhere before we
lbajardsilogic@106: 		// even reach this point.
lbajardsilogic@106: 		
lbajardsilogic@106: 		if (tmpSize < channels * work * 2) {
lbajardsilogic@106: 			delete[] tmp;
lbajardsilogic@106: 			tmp = new float[channels * work * 2];
lbajardsilogic@106: 			tmpSize = channels * work * 2;
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		float *nonintlv = tmp + channels * work;
lbajardsilogic@106: 		float *intlv = tmp;
lbajardsilogic@106: 		float *srcout = tmp + channels * work;
lbajardsilogic@106: 		
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 			for (size_t i = 0; i < orig; ++i) {
lbajardsilogic@106: 				nonintlv[channels * i + c] = 0.0f;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 			bufferPtrs[c] = nonintlv + c * orig;
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		got = mixModels(f, orig, bufferPtrs);
lbajardsilogic@0: 
lbajardsilogic@106: 		// and interleave into first half
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 			for (size_t i = 0; i < got; ++i) {
lbajardsilogic@106: 				float sample = nonintlv[c * got + i];
lbajardsilogic@106: 				intlv[channels * i + c] = sample;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@106: 			
lbajardsilogic@106: 		SRC_DATA data;
lbajardsilogic@106: 		data.data_in = intlv;
lbajardsilogic@106: 		data.data_out = srcout;
lbajardsilogic@106: 		data.input_frames = got;
lbajardsilogic@106: 		data.output_frames = work;
lbajardsilogic@106: 		data.src_ratio = ratio;
lbajardsilogic@106: 		data.end_of_input = 0;
lbajardsilogic@0: 		
lbajardsilogic@106: 		int err = 0;
lbajardsilogic@0: 
lbajardsilogic@106: 		if (m_timeStretcher && m_timeStretcher->getRatio() < 0.4) {
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 			std::cout << "Using crappy converter" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@106: 			src_process(m_crapConverter, &data);
lbajardsilogic@106: 		} else {
lbajardsilogic@106: 			src_process(m_converter, &data);
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		size_t toCopy = size_t(got * ratio + 0.1);
lbajardsilogic@0: 
lbajardsilogic@106: 		if (err) {
lbajardsilogic@106: 			std::cerr
lbajardsilogic@106: 			<< "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: "
lbajardsilogic@106: 			<< src_strerror(err) << std::endl;
lbajardsilogic@106: 			//!!! Then what?
lbajardsilogic@106: 		} else {
lbajardsilogic@106: 			got = data.input_frames_used;
lbajardsilogic@106: 			toCopy = data.output_frames_gen;
lbajardsilogic@106: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 			std::cout << "Resampled " << got << " frames to " << toCopy << " frames" << std::endl;
lbajardsilogic@106: #endif
lbajardsilogic@106: 		}
lbajardsilogic@106: 		
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 			for (size_t i = 0; i < toCopy; ++i) {
lbajardsilogic@106: 				tmp[i] = srcout[channels * i + c];
lbajardsilogic@106: 			}
lbajardsilogic@106: 			RingBuffer<float> *wb = getWriteRingBuffer(c);
lbajardsilogic@106: 			if (wb) wb->write(tmp, toCopy);
lbajardsilogic@106: 		}
lbajardsilogic@106: 
lbajardsilogic@106: 		m_writeBufferFill = f;
lbajardsilogic@106: 		if (readWriteEqual) m_readBufferFill = f;
lbajardsilogic@106: 
lbajardsilogic@0: 	} else {
lbajardsilogic@106: 
lbajardsilogic@106: 		// space must be a multiple of generatorBlockSize
lbajardsilogic@106: 		space = (space / generatorBlockSize) * generatorBlockSize;
lbajardsilogic@106: 		if (space == 0) return false;
lbajardsilogic@106: 
lbajardsilogic@106: 		if (tmpSize < channels * space) {
lbajardsilogic@106: 			delete[] tmp;
lbajardsilogic@106: 			tmp = new float[channels * space];
lbajardsilogic@106: 			tmpSize = channels * space;
lbajardsilogic@106: 		}
lbajardsilogic@106: 
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 
lbajardsilogic@106: 			bufferPtrs[c] = tmp + c * space;
lbajardsilogic@106: 		    
lbajardsilogic@106: 			for (size_t i = 0; i < space; ++i) {
lbajardsilogic@106: 				tmp[c * space + i] = 0.0f;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@106: 
lbajardsilogic@106: 		size_t got = mixModels(f, space, bufferPtrs);
lbajardsilogic@106: 
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 
lbajardsilogic@106: 			RingBuffer<float> *wb = getWriteRingBuffer(c);
lbajardsilogic@106: 			if (wb) {
lbajardsilogic@106: 				size_t actual = wb->write(bufferPtrs[c], got);
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 			std::cout << "Wrote " << actual << " samples for ch " << c << ", now "
lbajardsilogic@106: 				  << wb->getReadSpace() << " to read" 
lbajardsilogic@106: 				  << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@106: 				if (actual < got) {
lbajardsilogic@106: 					std::cerr << "WARNING: Buffer overrun in channel " << c
lbajardsilogic@106: 							  << ": wrote " << actual << " of " << got
lbajardsilogic@106: 							  << " samples" << std::endl;
lbajardsilogic@106: 				}
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		m_writeBufferFill = f;
lbajardsilogic@106: 		if (readWriteEqual) m_readBufferFill = f;
lbajardsilogic@0: 
lbajardsilogic@106: 		//!!! 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
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     return true;
lbajardsilogic@0: }    
lbajardsilogic@0: 
lbajardsilogic@0: size_t
lbajardsilogic@0: AudioCallbackPlaySource::mixModels(size_t &frame, size_t count, float **buffers)
lbajardsilogic@0: {
lbajardsilogic@0:     size_t processed = 0;
lbajardsilogic@0:     size_t chunkStart = frame;
lbajardsilogic@0:     size_t chunkSize = count;
lbajardsilogic@0:     size_t selectionSize = 0;
lbajardsilogic@0:     size_t nextChunkStart = chunkStart + chunkSize;
lbajardsilogic@0:     
lbajardsilogic@0:     bool looping = m_viewManager->getPlayLoopMode();
lbajardsilogic@0:     bool constrained = (m_viewManager->getPlaySelectionMode() &&
lbajardsilogic@0: 			!m_viewManager->getSelections().empty());
lbajardsilogic@0: 
lbajardsilogic@0:     static float **chunkBufferPtrs = 0;
lbajardsilogic@0:     static size_t chunkBufferPtrCount = 0;
lbajardsilogic@0:     size_t channels = getTargetChannelCount();
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     if (chunkBufferPtrCount < channels) {
lbajardsilogic@106: 		if (chunkBufferPtrs) delete[] chunkBufferPtrs;
lbajardsilogic@106: 		chunkBufferPtrs = new float *[channels];
lbajardsilogic@106: 		chunkBufferPtrCount = channels;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 		chunkBufferPtrs[c] = buffers[c];
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     while (processed < count) {
lbajardsilogic@0: 	
lbajardsilogic@106: 		chunkSize = count - processed;
lbajardsilogic@106: 		nextChunkStart = chunkStart + chunkSize;
lbajardsilogic@106: 		selectionSize = 0;
lbajardsilogic@0: 
lbajardsilogic@106: 		size_t fadeIn = 0, fadeOut = 0;
lbajardsilogic@0: 
lbajardsilogic@106: 		if (constrained) {
lbajardsilogic@106: 		    
lbajardsilogic@106: 			Selection selection =
lbajardsilogic@106: 			m_viewManager->getContainingSelection(chunkStart, true);
lbajardsilogic@106: 		    
lbajardsilogic@106: 			if (selection.isEmpty()) {
lbajardsilogic@106: 				if (looping) {
lbajardsilogic@106: 					selection = *m_viewManager->getSelections().begin();
lbajardsilogic@106: 					chunkStart = selection.getStartFrame();
lbajardsilogic@106: 					fadeIn = 50;
lbajardsilogic@106: 				}
lbajardsilogic@106: 			}
lbajardsilogic@106: 
lbajardsilogic@106: 			if (selection.isEmpty()) {
lbajardsilogic@106: 
lbajardsilogic@106: 				chunkSize = 0;
lbajardsilogic@106: 				nextChunkStart = chunkStart;
lbajardsilogic@106: 
lbajardsilogic@106: 			} else {
lbajardsilogic@106: 
lbajardsilogic@106: 				selectionSize =
lbajardsilogic@106: 					selection.getEndFrame() -
lbajardsilogic@106: 					selection.getStartFrame();
lbajardsilogic@106: 
lbajardsilogic@106: 				if (chunkStart < selection.getStartFrame()) {
lbajardsilogic@106: 					chunkStart = selection.getStartFrame();
lbajardsilogic@106: 					fadeIn = 50;
lbajardsilogic@106: 				}
lbajardsilogic@106: 
lbajardsilogic@106: 				nextChunkStart = chunkStart + chunkSize;
lbajardsilogic@106: 
lbajardsilogic@106: 				if (nextChunkStart >= selection.getEndFrame()) {
lbajardsilogic@106: 					nextChunkStart = selection.getEndFrame();
lbajardsilogic@106: 					fadeOut = 50;
lbajardsilogic@106: 				}
lbajardsilogic@106: 
lbajardsilogic@106: 				chunkSize = nextChunkStart - chunkStart;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		
lbajardsilogic@106: 		} else if (looping && m_lastModelEndFrame > 0) {
lbajardsilogic@106: 
lbajardsilogic@106: 			if (chunkStart >= m_lastModelEndFrame) {
lbajardsilogic@106: 				chunkStart = 0;
lbajardsilogic@106: 			}
lbajardsilogic@106: 			if (chunkSize > m_lastModelEndFrame - chunkStart) {
lbajardsilogic@106: 				chunkSize = m_lastModelEndFrame - chunkStart;
lbajardsilogic@106: 			}
lbajardsilogic@106: 			nextChunkStart = chunkStart + chunkSize;
lbajardsilogic@0: 		}
lbajardsilogic@106: 		
lbajardsilogic@106: 	//	std::cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << std::endl;
lbajardsilogic@0: 
lbajardsilogic@106: 		if (!chunkSize) {
lbajardsilogic@106: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 			std::cout << "Ending selection playback at " << nextChunkStart << std::endl;
lbajardsilogic@106: #endif
lbajardsilogic@106: 			// We need to maintain full buffers so that the other
lbajardsilogic@106: 			// thread can tell where it's got to in the playback -- so
lbajardsilogic@106: 			// return the full amount here
lbajardsilogic@106: 			frame = frame + count;
lbajardsilogic@106: 			return count;
lbajardsilogic@0: 		}
lbajardsilogic@0: 
lbajardsilogic@106: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 		std::cout << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << std::endl;
lbajardsilogic@106: #endif
lbajardsilogic@0: 
lbajardsilogic@106: 		size_t got = 0;
lbajardsilogic@106: 
lbajardsilogic@106: 		if (selectionSize < 100) {
lbajardsilogic@106: 			fadeIn = 0;
lbajardsilogic@106: 			fadeOut = 0;
lbajardsilogic@106: 		} else if (selectionSize < 300) {
lbajardsilogic@106: 			if (fadeIn > 0) fadeIn = 10;
lbajardsilogic@106: 			if (fadeOut > 0) fadeOut = 10;
lbajardsilogic@0: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		if (fadeIn > 0) {
lbajardsilogic@106: 			if (processed * 2 < fadeIn) {
lbajardsilogic@106: 				fadeIn = processed * 2;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		if (fadeOut > 0) {
lbajardsilogic@106: 			if ((count - processed - chunkSize) * 2 < fadeOut) {
lbajardsilogic@106: 				fadeOut = (count - processed - chunkSize) * 2;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		for (std::set<Model *>::iterator mi = m_models.begin();
lbajardsilogic@106: 			 mi != m_models.end(); ++mi) {
lbajardsilogic@106: 		    
lbajardsilogic@106: 			got = m_audioGenerator->mixModel(*mi, chunkStart, 
lbajardsilogic@106: 							 chunkSize, chunkBufferPtrs,
lbajardsilogic@106: 							 fadeIn, fadeOut);
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 			chunkBufferPtrs[c] += chunkSize;
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@106: 		processed += chunkSize;
lbajardsilogic@106: 		chunkStart = nextChunkStart;
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "Returning selection playback " << processed << " frames to " << nextChunkStart << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     frame = nextChunkStart;
lbajardsilogic@0:     return processed;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::unifyRingBuffers()
lbajardsilogic@0: {
lbajardsilogic@0:     if (m_readBuffers == m_writeBuffers) return;
lbajardsilogic@0: 
lbajardsilogic@0:     // only unify if there will be something to read
lbajardsilogic@0:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@0: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
lbajardsilogic@0: 	if (wb) {
lbajardsilogic@0: 	    if (wb->getReadSpace() < m_blockSize * 2) {
lbajardsilogic@0: 		if ((m_writeBufferFill + m_blockSize * 2) < 
lbajardsilogic@0: 		    m_lastModelEndFrame) {
lbajardsilogic@0: 		    // OK, we don't have enough and there's more to
lbajardsilogic@0: 		    // read -- don't unify until we can do better
lbajardsilogic@0: 		    return;
lbajardsilogic@0: 		}
lbajardsilogic@0: 	    }
lbajardsilogic@0: 	    break;
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     size_t rf = m_readBufferFill;
lbajardsilogic@0:     RingBuffer<float> *rb = getReadRingBuffer(0);
lbajardsilogic@0:     if (rb) {
lbajardsilogic@0: 	size_t rs = rb->getReadSpace();
lbajardsilogic@0: 	//!!! incorrect when in non-contiguous selection, see comments elsewhere
lbajardsilogic@0: //	std::cout << "rs = " << rs << std::endl;
lbajardsilogic@0: 	if (rs < rf) rf -= rs;
lbajardsilogic@0: 	else rf = 0;
lbajardsilogic@0:     }
lbajardsilogic@0:     
lbajardsilogic@0:     //std::cout << "m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0:     size_t wf = m_writeBufferFill;
lbajardsilogic@0:     size_t skip = 0;
lbajardsilogic@0:     for (size_t c = 0; c < getTargetChannelCount(); ++c) {
lbajardsilogic@0: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
lbajardsilogic@0: 	if (wb) {
lbajardsilogic@0: 	    if (c == 0) {
lbajardsilogic@0: 		
lbajardsilogic@0: 		size_t wrs = wb->getReadSpace();
lbajardsilogic@0: //		std::cout << "wrs = " << wrs << std::endl;
lbajardsilogic@0: 
lbajardsilogic@0: 		if (wrs < wf) wf -= wrs;
lbajardsilogic@0: 		else wf = 0;
lbajardsilogic@0: //		std::cout << "wf = " << wf << std::endl;
lbajardsilogic@0: 		
lbajardsilogic@0: 		if (wf < rf) skip = rf - wf;
lbajardsilogic@0: 		if (skip == 0) break;
lbajardsilogic@0: 	    }
lbajardsilogic@0: 
lbajardsilogic@0: //	    std::cout << "skipping " << skip << std::endl;
lbajardsilogic@0: 	    wb->skip(skip);
lbajardsilogic@0: 	}
lbajardsilogic@0:     }
lbajardsilogic@0: 		    
lbajardsilogic@0:     m_bufferScavenger.claim(m_readBuffers);
lbajardsilogic@0:     m_readBuffers = m_writeBuffers;
lbajardsilogic@0:     m_readBufferFill = m_writeBufferFill;
lbajardsilogic@0: //    std::cout << "unified" << std::endl;
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@0: void
lbajardsilogic@0: AudioCallbackPlaySource::FillThread::run()
lbajardsilogic@0: {
lbajardsilogic@0:     AudioCallbackPlaySource &s(m_source);
lbajardsilogic@0:     
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@0:     std::cout << "AudioCallbackPlaySourceFillThread starting" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@0:     s.m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@0:     bool previouslyPlaying = s.m_playing;
lbajardsilogic@0:     bool work = false;
lbajardsilogic@0: 
lbajardsilogic@0:     while (!s.m_exiting) {
lbajardsilogic@0: 
lbajardsilogic@106: 		s.unifyRingBuffers();
lbajardsilogic@106: 		s.m_bufferScavenger.scavenge();
lbajardsilogic@106: 		s.m_pluginScavenger.scavenge();
lbajardsilogic@106: 		s.m_timeStretcherScavenger.scavenge();
lbajardsilogic@0: 
lbajardsilogic@106: 		if (work && s.m_playing && s.getSourceSampleRate()) {
lbajardsilogic@106: 		    
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 			std::cout << "AudioCallbackPlaySourceFillThread: not waiting" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@106: 			s.m_mutex.unlock();
lbajardsilogic@106: 			s.m_mutex.lock();
lbajardsilogic@0: 
lbajardsilogic@106: 		} else {
lbajardsilogic@106: 		    
lbajardsilogic@106: 			float ms = 100;
lbajardsilogic@106: 			if (s.getSourceSampleRate() > 0) {
lbajardsilogic@106: 				ms = float(m_ringBufferSize) / float(s.getSourceSampleRate()) * 1000.0;
lbajardsilogic@106: 			}
lbajardsilogic@106: 		    
lbajardsilogic@106: 			if (s.m_playing) ms /= 10;
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 				if (!s.m_playing) std::cout << std::endl;
lbajardsilogic@106: 			std::cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@106: 		    
lbajardsilogic@106: 			s.m_condition.wait(&s.m_mutex, size_t(ms));
lbajardsilogic@106: 		}
lbajardsilogic@0: 
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 		std::cout << "AudioCallbackPlaySourceFillThread: awoken" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@0: 
lbajardsilogic@106: 		work = false;
lbajardsilogic@0: 
lbajardsilogic@106: 		if (!s.getSourceSampleRate()) continue;
lbajardsilogic@0: 
lbajardsilogic@106: 		bool playing = s.m_playing;
lbajardsilogic@0: 
lbajardsilogic@106: 		if (playing && !previouslyPlaying) {
lbajardsilogic@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
lbajardsilogic@106: 			std::cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << std::endl;
lbajardsilogic@0: #endif
lbajardsilogic@106: 			for (size_t c = 0; c < s.getTargetChannelCount(); ++c) {
lbajardsilogic@106: 				RingBuffer<float> *rb = s.getReadRingBuffer(c);
lbajardsilogic@106: 				if (rb) rb->reset();
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@106: 		previouslyPlaying = playing;
lbajardsilogic@0: 
lbajardsilogic@106: 		work = s.fillBuffers();
lbajardsilogic@0:     }
lbajardsilogic@0: 
lbajardsilogic@0:     s.m_mutex.unlock();
lbajardsilogic@0: }
lbajardsilogic@0: 
lbajardsilogic@79: void AudioCallbackPlaySource::applyRealTimeFilters(size_t count, float **buffers)
lbajardsilogic@79: {
lbajardsilogic@79: 	if (!m_filterStack) return;
lbajardsilogic@79: 
lbajardsilogic@106: 	size_t required = m_filterStack->getRequiredInputSamples(count);
lbajardsilogic@106: 
lbajardsilogic@106: 	size_t channels = getTargetChannelCount();
lbajardsilogic@106: 
lbajardsilogic@106: 	size_t got = required;
lbajardsilogic@106: 
lbajardsilogic@106: 	//if no filters are available
lbajardsilogic@106: 	if (required == 0)
lbajardsilogic@106: 	{
lbajardsilogic@106: 		got = count;
lbajardsilogic@106: 		for (size_t ch = 0; ch < channels; ++ch) 
lbajardsilogic@106: 		{
lbajardsilogic@106: 			RingBuffer<float> *rb = getReadRingBuffer(ch);
lbajardsilogic@106: 			if (rb) {
lbajardsilogic@106: 				size_t gotHere = rb->read(buffers[ch], got);
lbajardsilogic@106: 				if (gotHere < got)
lbajardsilogic@106: 					got = gotHere;
lbajardsilogic@106: 		    }
lbajardsilogic@106: 
lbajardsilogic@106: 			for (size_t ch = 0; ch < channels; ++ch) {
lbajardsilogic@106: 				for (size_t i = got; i < count; ++i) {
lbajardsilogic@106: 					buffers[ch][i] = 0.0;
lbajardsilogic@106: 				}
lbajardsilogic@106: 			}
lbajardsilogic@106: 		}
lbajardsilogic@106: 		return;
lbajardsilogic@106: 	}
lbajardsilogic@106: 
lbajardsilogic@106: 	float **ib = (float**) malloc(channels*sizeof(float*));
lbajardsilogic@106: 
lbajardsilogic@106: 	for (size_t c = 0; c < channels; ++c) 
lbajardsilogic@106: 	{
lbajardsilogic@106: 		ib[c] = (float*) malloc(required*sizeof(float));
lbajardsilogic@106: 		RingBuffer<float> *rb = getReadRingBuffer(c);
lbajardsilogic@110: 		if (!rb) {
lbajardsilogic@110:             std::cerr << "WARNING: AudioCallbackPlaySource::applyRealTimeFilters: "
lbajardsilogic@110:                       << "No ring buffer available for channel " << c
lbajardsilogic@110:                       << ", returning no data here" << std::endl;
lbajardsilogic@110:             return;
lbajardsilogic@110:         }
lbajardsilogic@110: 		size_t rs = rb->getReadSpace();
lbajardsilogic@110: 		if (rs < required) {
lbajardsilogic@110: 			std::cerr << "WARNING: AudioCallbackPlaySource::applyRealTimeFilters: "
lbajardsilogic@110:                       << "Ring buffer for channel " << c << " has only "
lbajardsilogic@110:                       << rs << " (of " << got << ") samples available, "
lbajardsilogic@110:                       << "exit" << std::endl;
lbajardsilogic@110: 			return;
lbajardsilogic@110: 		}
lbajardsilogic@106:         if (rb) {
lbajardsilogic@106: 			size_t gotHere = rb->peek(ib[c], got);
lbajardsilogic@106: 			if (gotHere < got)
lbajardsilogic@106: 				got = gotHere;
lbajardsilogic@106: 		}
lbajardsilogic@106: 	}
lbajardsilogic@106: 	if (got < required)
lbajardsilogic@106: 	{
lbajardsilogic@106: 		std::cerr << "ERROR applyRealTimeFilters(): Read underrun in playback ("
lbajardsilogic@106:               << got << " < " << required << ")" << std::endl;
lbajardsilogic@106: 		return; 
lbajardsilogic@106: 	}
lbajardsilogic@106: 
lbajardsilogic@106:     m_filterStack->putInput(ib, required);
lbajardsilogic@106: 
lbajardsilogic@106: 	m_filterStack->getOutput(buffers, count);
lbajardsilogic@106: 
lbajardsilogic@106: 	//move the read pointer
lbajardsilogic@106: 	got = m_filterStack->getRequiredSkipSamples();
lbajardsilogic@106: 	for (size_t c = 0; c < channels; ++c) 
lbajardsilogic@106: 	{
lbajardsilogic@106: 		RingBuffer<float> *rb = getReadRingBuffer(c);
lbajardsilogic@106:         if (rb) {
lbajardsilogic@106: 			size_t gotHere = rb->skip(got);
lbajardsilogic@106: 			if (gotHere < got)
lbajardsilogic@106: 				got = gotHere;
lbajardsilogic@106: 		}
lbajardsilogic@106: 	}
lbajardsilogic@106: 	
lbajardsilogic@106: 	//delete
lbajardsilogic@106: 	for (size_t c = 0; c < channels; ++c) {
lbajardsilogic@106: 		delete ib[c];
lbajardsilogic@106: 	}
lbajardsilogic@106: 	delete ib;
lbajardsilogic@79: }