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@105: #include "base/ViewManagerBase.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@506: #include "data/model/ReadOnlyWaveFileModel.h"
Chris@43: #include "data/model/SparseOneDimensionalModel.h"
Chris@43: #include "plugin/RealTimePluginInstance.h"
Chris@62: 
Chris@468: #include "bqaudioio/SystemPlaybackTarget.h"
Chris@551: #include "bqaudioio/ResamplerWrapper.h"
Chris@91: 
Chris@559: #include "bqvec/VectorOps.h"
Chris@559: 
Chris@62: #include <rubberband/RubberBandStretcher.h>
Chris@62: using namespace RubberBand;
Chris@43: 
Chris@559: using breakfastquay::v_zero_channels;
Chris@559: 
Chris@43: #include <iostream>
Chris@43: #include <cassert>
Chris@43: 
Chris@510: //#define DEBUG_AUDIO_PLAY_SOURCE 1
Chris@43: //#define DEBUG_AUDIO_PLAY_SOURCE_PLAYING 1
Chris@43: 
Chris@366: static const int DEFAULT_RING_BUFFER_SIZE = 131071;
Chris@43: 
Chris@105: AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManagerBase *manager,
Chris@57:                                                  QString clientName) :
Chris@43:     m_viewManager(manager),
Chris@43:     m_audioGenerator(new AudioGenerator()),
Chris@468:     m_clientName(clientName.toUtf8().data()),
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@553:     m_deviceSampleRate(0),
Chris@559:     m_deviceChannelCount(0),
Chris@43:     m_playLatency(0),
Chris@91:     m_target(0),
Chris@91:     m_lastRetrievalTimestamp(0.0),
Chris@91:     m_lastRetrievedBlockSize(0),
Chris@102:     m_trustworthyTimestamps(true),
Chris@102:     m_lastCurrentFrame(0),
Chris@43:     m_playing(false),
Chris@43:     m_exiting(false),
Chris@43:     m_lastModelEndFrame(0),
Chris@193:     m_ringBufferSize(DEFAULT_RING_BUFFER_SIZE),
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@130:     m_monoStretcher(0),
Chris@91:     m_stretchRatio(1.0),
Chris@405:     m_stretchMono(false),
Chris@91:     m_stretcherInputCount(0),
Chris@91:     m_stretcherInputs(0),
Chris@91:     m_stretcherInputSizes(0),
Chris@551:     m_fillThread(0),
Chris@551:     m_resamplerWrapper(0)
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@300:     connect(this, SIGNAL(playStatusChanged(bool)),
Chris@300:             m_viewManager, SLOT(playStatusChanged(bool)));
Chris@300: 
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@177: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@233:     SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource entering" << endl;
Chris@177: #endif
Chris@43:     m_exiting = true;
Chris@43: 
Chris@43:     if (m_fillThread) {
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "AudioCallbackPlaySource dtor: awakening thread" << endl;
Chris@212: #endif
Chris@212:         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@366:     for (int 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@130:     delete m_timeStretcher;
Chris@130:     delete m_monoStretcher;
Chris@130: 
Chris@43:     m_bufferScavenger.scavenge(true);
Chris@43:     m_pluginScavenger.scavenge(true);
Chris@177: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@233:     SVDEBUG << "AudioCallbackPlaySource::~AudioCallbackPlaySource finishing" << endl;
Chris@177: #endif
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@418:     bool willPlay = 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@559:     bool buffersIncreased = false, srChanged = false;
Chris@43: 
Chris@366:     int modelChannels = 1;
Chris@506:     ReadOnlyWaveFileModel *rowfm = qobject_cast<ReadOnlyWaveFileModel *>(model);
Chris@506:     if (rowfm) modelChannels = rowfm->getChannelCount();
Chris@43:     if (modelChannels > m_sourceChannelCount) {
Chris@43: 	m_sourceChannelCount = modelChannels;
Chris@43:     }
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@295:     cout << "AudioCallbackPlaySource: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << endl;
Chris@43: #endif
Chris@43: 
Chris@43:     if (m_sourceSampleRate == 0) {
Chris@43: 
Chris@566:         SVDEBUG << "AudioCallbackPlaySource::addModel: Source rate changing from 0 to "
Chris@566:             << model->getSampleRate() << endl;
Chris@566: 
Chris@43: 	m_sourceSampleRate = model->getSampleRate();
Chris@43: 	srChanged = true;
Chris@43: 
Chris@43:     } else if (model->getSampleRate() != m_sourceSampleRate) {
Chris@43: 
Chris@506:         // If this is a read-only wave file model and we have no
Chris@506:         // other, we can just switch to this model's sample rate
Chris@43: 
Chris@506:         if (rowfm) {
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@506:                 // Only read-only wave file models should be
Chris@506:                 // considered conflicting -- writable wave file models
Chris@506:                 // are derived and we shouldn't take their rates into
Chris@506:                 // account.  Also, don't give any particular weight to
Chris@506:                 // a file that's already playing at the wrong rate
Chris@506:                 // anyway
Chris@506:                 ReadOnlyWaveFileModel *other =
Chris@506:                     qobject_cast<ReadOnlyWaveFileModel *>(*i);
Chris@506:                 if (other && other != rowfm &&
Chris@506:                     other->getSampleRate() != model->getSampleRate() &&
Chris@506:                     other->getSampleRate() == m_sourceSampleRate) {
Chris@233:                     SVDEBUG << "AudioCallbackPlaySource::addModel: Conflicting wave file model " << *i << " found" << endl;
Chris@43:                     conflicting = true;
Chris@43:                     break;
Chris@43:                 }
Chris@43:             }
Chris@43: 
Chris@43:             if (conflicting) {
Chris@43: 
Chris@233:                 SVDEBUG << "AudioCallbackPlaySource::addModel: ERROR: "
Chris@229:                           << "New model sample rate does not match" << endl
Chris@43:                           << "existing model(s) (new " << model->getSampleRate()
Chris@43:                           << " vs " << m_sourceSampleRate
Chris@43:                           << "), playback will be wrong"
Chris@229:                           << endl;
Chris@43:                 
Chris@43:                 emit sampleRateMismatch(model->getSampleRate(),
Chris@43:                                         m_sourceSampleRate,
Chris@43:                                         false);
Chris@43:             } else {
Chris@566:                 SVDEBUG << "AudioCallbackPlaySource::addModel: Source rate changing from "
Chris@566:                         << m_sourceSampleRate << " to " << model->getSampleRate() << endl;
Chris@566:                 
Chris@43:                 m_sourceSampleRate = model->getSampleRate();
Chris@43:                 srChanged = true;
Chris@43:             }
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@366:     if (!m_writeBuffers || (int)m_writeBuffers->size() < getTargetChannelCount()) {
Chris@570:         cerr << "m_writeBuffers size = " << (m_writeBuffers ? m_writeBuffers->size() : 0) << endl;
Chris@570:         cerr << "target channel count = " << (getTargetChannelCount()) << endl;
Chris@43: 	clearRingBuffers(true, getTargetChannelCount());
Chris@559: 	buffersIncreased = true;
Chris@43:     } else {
Chris@418: 	if (willPlay) clearRingBuffers(true);
Chris@43:     }
Chris@43: 
Chris@552:     if (srChanged) {
Chris@553: 
Chris@552:         SVCERR << "AudioCallbackPlaySource: Source rate changed" << endl;
Chris@553: 
Chris@552:         if (m_resamplerWrapper) {
Chris@552:             SVCERR << "AudioCallbackPlaySource: Source sample rate changed to "
Chris@552:                 << m_sourceSampleRate << ", updating resampler wrapper" << endl;
Chris@552:             m_resamplerWrapper->changeApplicationSampleRate
Chris@552:                 (int(round(m_sourceSampleRate)));
Chris@552:             m_resamplerWrapper->reset();
Chris@552:         }
Chris@553: 
Chris@553:         delete m_timeStretcher;
Chris@553:         delete m_monoStretcher;
Chris@553:         m_timeStretcher = 0;
Chris@553:         m_monoStretcher = 0;
Chris@553:         
Chris@553:         if (m_stretchRatio != 1.f) {
Chris@553:             setTimeStretch(m_stretchRatio);
Chris@553:         }
Chris@43:     }
Chris@43: 
Chris@164:     rebuildRangeLists();
Chris@164: 
Chris@43:     m_mutex.unlock();
Chris@43: 
Chris@43:     m_audioGenerator->setTargetChannelCount(getTargetChannelCount());
Chris@43: 
Chris@559:     if (buffersIncreased) {
Chris@570:         SVDEBUG << "AudioCallbackPlaySource::addModel: Number of buffers increased to " << getTargetChannelCount() << endl;
Chris@570:         if (getTargetChannelCount() > getDeviceChannelCount()) {
Chris@570:             SVDEBUG << "AudioCallbackPlaySource::addModel: This is more than the device channel count, signalling channelCountIncreased" << endl;
Chris@570:             emit channelCountIncreased(getTargetChannelCount());
Chris@570:         } else {
Chris@570:             SVDEBUG << "AudioCallbackPlaySource::addModel: This is no more than the device channel count (" << getDeviceChannelCount() << "), so taking no action" << endl;
Chris@570:         }
Chris@559:     }
Chris@559:     
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@559:     SVDEBUG << "AudioCallbackPlaySource::addModel: now have " << m_models.size() << " model(s)" << endl;
Chris@43: #endif
Chris@43: 
Chris@435:     connect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)),
Chris@435:             this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t)));
Chris@43: 
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "AudioCallbackPlaySource::addModel: awakening thread" << endl;
Chris@212: #endif
Chris@559:     
Chris@43:     m_condition.wakeAll();
Chris@43: }
Chris@43: 
Chris@43: void
Chris@435: AudioCallbackPlaySource::modelChangedWithin(sv_frame_t 
Chris@367: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@367:                                             startFrame
Chris@367: #endif
Chris@435:                                             , sv_frame_t endFrame)
Chris@43: {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@367:     SVDEBUG << "AudioCallbackPlaySource::modelChangedWithin(" << startFrame << "," << endFrame << ")" << endl;
Chris@43: #endif
Chris@93:     if (endFrame > m_lastModelEndFrame) {
Chris@93:         m_lastModelEndFrame = endFrame;
Chris@99:         rebuildRangeLists();
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@293:     cout << "AudioCallbackPlaySource::removeModel(" << model << ")" << endl;
Chris@43: #endif
Chris@43: 
Chris@435:     disconnect(model, SIGNAL(modelChangedWithin(sv_frame_t, sv_frame_t)),
Chris@435:                this, SLOT(modelChangedWithin(sv_frame_t, sv_frame_t)));
Chris@43: 
Chris@43:     m_models.erase(model);
Chris@43: 
Chris@566:     // I don't think we have to do this any more: if a new model is
Chris@566:     // loaded at a different rate, we'll hit the non-conflicting path
Chris@566:     // in addModel and the rate will be updated without problems; but
Chris@566:     // if a new model is loaded at the rate that we were using for the
Chris@566:     // last one, then we save work by not having reset this here
Chris@566:     //
Chris@566: //    if (m_models.empty()) {
Chris@566: //	m_sourceSampleRate = 0;
Chris@566: //    }
Chris@43: 
Chris@436:     sv_frame_t lastEnd = 0;
Chris@43:     for (std::set<Model *>::const_iterator i = m_models.begin();
Chris@43: 	 i != m_models.end(); ++i) {
Chris@164: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293: 	cout << "AudioCallbackPlaySource::removeModel(" << model << "): checking end frame on model " << *i << endl;
Chris@164: #endif
Chris@367: 	if ((*i)->getEndFrame() > lastEnd) {
Chris@367:             lastEnd = (*i)->getEndFrame();
Chris@367:         }
Chris@164: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293: 	cout << "(done, lastEnd now " << lastEnd << ")" << endl;
Chris@164: #endif
Chris@43:     }
Chris@43:     m_lastModelEndFrame = lastEnd;
Chris@43: 
Chris@212:     m_audioGenerator->removeModel(model);
Chris@212: 
Chris@43:     m_mutex.unlock();
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@293:     cout << "AudioCallbackPlaySource::clearModels()" << endl;
Chris@43: #endif
Chris@43: 
Chris@43:     m_models.clear();
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@366: AudioCallbackPlaySource::clearRingBuffers(bool haveLock, int count)
Chris@43: {
Chris@43:     if (!haveLock) m_mutex.lock();
Chris@43: 
Chris@445: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << "clearRingBuffers" << endl;
Chris@445: #endif
Chris@397: 
Chris@93:     rebuildRangeLists();
Chris@93: 
Chris@43:     if (count == 0) {
Chris@436: 	if (m_writeBuffers) count = int(m_writeBuffers->size());
Chris@43:     }
Chris@43: 
Chris@445: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << "current playing frame = " << getCurrentPlayingFrame() << endl;
Chris@397: 
Chris@563:     cout << "write buffer fill (before) = " << m_writeBufferFill << endl;
Chris@445: #endif
Chris@445:     
Chris@93:     m_writeBufferFill = getCurrentBufferedFrame();
Chris@43: 
Chris@445: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << "current buffered frame = " << m_writeBufferFill << endl;
Chris@445: #endif
Chris@397: 
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@366:     for (int i = 0; i < count; ++i) {
Chris@43: 	m_writeBuffers->push_back(new RingBuffer<float>(m_ringBufferSize));
Chris@43:     }
Chris@43: 
Chris@442:     m_audioGenerator->reset();
Chris@442:     
Chris@293: //    cout << "AudioCallbackPlaySource::clearRingBuffers: Created "
Chris@293: //	      << count << " write buffers" << endl;
Chris@43: 
Chris@43:     if (!haveLock) {
Chris@43: 	m_mutex.unlock();
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@434: AudioCallbackPlaySource::play(sv_frame_t startFrame)
Chris@43: {
Chris@540:     if (!m_target) return;
Chris@540:     
Chris@414:     if (!m_sourceSampleRate) {
Chris@563:         SVCERR << "AudioCallbackPlaySource::play: No source sample rate available, not playing" << endl;
Chris@414:         return;
Chris@414:     }
Chris@414:     
Chris@43:     if (m_viewManager->getPlaySelectionMode() &&
Chris@43: 	!m_viewManager->getSelections().empty()) {
Chris@60: 
Chris@563: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:         cout << "AudioCallbackPlaySource::play: constraining frame " << startFrame << " to selection = ";
Chris@563: #endif
Chris@94: 
Chris@60:         startFrame = m_viewManager->constrainFrameToSelection(startFrame);
Chris@60: 
Chris@563: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:         cout << startFrame << endl;
Chris@563: #endif
Chris@94: 
Chris@43:     } else {
Chris@454:         if (startFrame < 0) {
Chris@454:             startFrame = 0;
Chris@454:         }
Chris@43: 	if (startFrame >= m_lastModelEndFrame) {
Chris@43: 	    startFrame = 0;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@132: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << "play(" << startFrame << ") -> aligned playback model ";
Chris@132: #endif
Chris@60: 
Chris@60:     startFrame = m_viewManager->alignReferenceToPlaybackFrame(startFrame);
Chris@60: 
Chris@189: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << startFrame << endl;
Chris@189: #endif
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@102:     // NO -- we can end up playing some first -- always reset here
Chris@43: 
Chris@43:     m_mutex.lock();
Chris@102: 
Chris@91:     if (m_timeStretcher) {
Chris@91:         m_timeStretcher->reset();
Chris@91:     }
Chris@130:     if (m_monoStretcher) {
Chris@130:         m_monoStretcher->reset();
Chris@130:     }
Chris@102: 
Chris@102:     m_readBufferFill = m_writeBufferFill = startFrame;
Chris@102:     if (m_readBuffers) {
Chris@366:         for (int c = 0; c < getTargetChannelCount(); ++c) {
Chris@102:             RingBuffer<float> *rb = getReadRingBuffer(c);
Chris@132: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:             cout << "reset ring buffer for channel " << c << endl;
Chris@132: #endif
Chris@102:             if (rb) rb->reset();
Chris@102:         }
Chris@43:     }
Chris@102: 
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@102:     m_lastCurrentFrame = 0;
Chris@43:     m_playing = true;
Chris@212: 
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "AudioCallbackPlaySource::play: awakening thread" << endl;
Chris@212: #endif
Chris@212: 
Chris@43:     m_condition.wakeAll();
Chris@158:     if (changed) {
Chris@158:         emit playStatusChanged(m_playing);
Chris@158:         emit activity(tr("Play from %1").arg
Chris@158:                       (RealTime::frame2RealTime
Chris@158:                        (m_playStartFrame, m_sourceSampleRate).toText().c_str()));
Chris@158:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::stop()
Chris@43: {
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@233:     SVDEBUG << "AudioCallbackPlaySource::stop()" << endl;
Chris@212: #endif
Chris@43:     bool changed = m_playing;
Chris@43:     m_playing = false;
Chris@212: 
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "AudioCallbackPlaySource::stop: awakening thread" << endl;
Chris@212: #endif
Chris@212: 
Chris@43:     m_condition.wakeAll();
Chris@91:     m_lastRetrievalTimestamp = 0;
Chris@158:     if (changed) {
Chris@158:         emit playStatusChanged(m_playing);
Chris@158:         emit activity(tr("Stop at %1").arg
Chris@158:                       (RealTime::frame2RealTime
Chris@158:                        (m_lastCurrentFrame, m_sourceSampleRate).toText().c_str()));
Chris@158:     }
Chris@102:     m_lastCurrentFrame = 0;
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@552: AudioCallbackPlaySource::preferenceChanged(PropertyContainer::PropertyName )
Chris@43: {
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::audioProcessingOverload()
Chris@43: {
Chris@563:     SVCERR << "Audio processing overload!" << endl;
Chris@130: 
Chris@130:     if (!m_playing) return;
Chris@130: 
Chris@43:     RealTimePluginInstance *ap = m_auditioningPlugin;
Chris@130:     if (ap && !m_auditioningPluginBypassed) {
Chris@43:         m_auditioningPluginBypassed = true;
Chris@43:         emit audioOverloadPluginDisabled();
Chris@130:         return;
Chris@130:     }
Chris@130: 
Chris@130:     if (m_timeStretcher &&
Chris@130:         m_timeStretcher->getTimeRatio() < 1.0 &&
Chris@130:         m_stretcherInputCount > 1 &&
Chris@130:         m_monoStretcher && !m_stretchMono) {
Chris@130:         m_stretchMono = true;
Chris@130:         emit audioTimeStretchMultiChannelDisabled();
Chris@130:         return;
Chris@43:     }
Chris@43: }
Chris@43: 
Chris@43: void
Chris@468: AudioCallbackPlaySource::setSystemPlaybackTarget(breakfastquay::SystemPlaybackTarget *target)
Chris@43: {
Chris@559:     if (target == 0) {
Chris@559:         // reset target-related facts and figures
Chris@559:         m_deviceSampleRate = 0;
Chris@559:         m_deviceChannelCount = 0;
Chris@559:     }
Chris@91:     m_target = target;
Chris@468: }
Chris@468: 
Chris@468: void
Chris@551: AudioCallbackPlaySource::setResamplerWrapper(breakfastquay::ResamplerWrapper *w)
Chris@551: {
Chris@551:     m_resamplerWrapper = w;
Chris@552:     if (m_resamplerWrapper && m_sourceSampleRate != 0) {
Chris@552:         m_resamplerWrapper->changeApplicationSampleRate
Chris@552:             (int(round(m_sourceSampleRate)));
Chris@552:     }
Chris@551: }
Chris@551: 
Chris@551: void
Chris@468: AudioCallbackPlaySource::setSystemPlaybackBlockSize(int size)
Chris@468: {
Chris@293:     cout << "AudioCallbackPlaySource::setTarget: Block size -> " << size << endl;
Chris@193:     if (size != 0) {
Chris@193:         m_blockSize = size;
Chris@193:     }
Chris@193:     if (size * 4 > m_ringBufferSize) {
Chris@472: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:         cout << "AudioCallbackPlaySource::setTarget: Buffer size "
Chris@472:              << size << " > a quarter of ring buffer size "
Chris@472:              << m_ringBufferSize << ", calling for more ring buffer"
Chris@472:              << endl;
Chris@472: #endif
Chris@193:         m_ringBufferSize = size * 4;
Chris@193:         if (m_writeBuffers && !m_writeBuffers->empty()) {
Chris@193:             clearRingBuffers();
Chris@193:         }
Chris@193:     }
Chris@43: }
Chris@43: 
Chris@366: int
Chris@43: AudioCallbackPlaySource::getTargetBlockSize() const
Chris@43: {
Chris@293: //    cout << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << endl;
Chris@436:     return int(m_blockSize);
Chris@43: }
Chris@43: 
Chris@43: void
Chris@468: AudioCallbackPlaySource::setSystemPlaybackLatency(int latency)
Chris@43: {
Chris@43:     m_playLatency = latency;
Chris@43: }
Chris@43: 
Chris@434: sv_frame_t
Chris@43: AudioCallbackPlaySource::getTargetPlayLatency() const
Chris@43: {
Chris@43:     return m_playLatency;
Chris@43: }
Chris@43: 
Chris@434: sv_frame_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@553:     sv_samplerate_t deviceRate = getDeviceSampleRate();
Chris@436:     sv_frame_t latency = m_playLatency; // at target rate
Chris@402:     RealTime latency_t = RealTime::zeroTime;
Chris@402: 
Chris@553:     if (deviceRate != 0) {
Chris@553:         latency_t = RealTime::frame2RealTime(latency, deviceRate);
Chris@402:     }
Chris@93: 
Chris@93:     return getCurrentFrame(latency_t);
Chris@93: }
Chris@93: 
Chris@434: sv_frame_t
Chris@93: AudioCallbackPlaySource::getCurrentBufferedFrame()
Chris@93: {
Chris@93:     return getCurrentFrame(RealTime::zeroTime);
Chris@93: }
Chris@93: 
Chris@434: sv_frame_t
Chris@93: AudioCallbackPlaySource::getCurrentFrame(RealTime latency_t)
Chris@93: {
Chris@553:     // The ring buffers contain data at the source sample rate and all
Chris@553:     // processing (including time stretching) happens at this
Chris@553:     // rate. Resampling only happens after the audio data leaves this
Chris@553:     // class.
Chris@553:     
Chris@553:     // (But because historically more than one sample rate could have
Chris@553:     // been involved here, we do latency calculations using RealTime
Chris@553:     // values instead of samples.)
Chris@43: 
Chris@553:     sv_samplerate_t rate = getSourceSampleRate();
Chris@91: 
Chris@553:     if (rate == 0) return 0;
Chris@91: 
Chris@366:     int inbuffer = 0; // at target rate
Chris@91: 
Chris@366:     for (int c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *rb = getReadRingBuffer(c);
Chris@43: 	if (rb) {
Chris@366: 	    int here = rb->getReadSpace();
Chris@91: 	    if (c == 0 || here < inbuffer) inbuffer = here;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@436:     sv_frame_t readBufferFill = m_readBufferFill;
Chris@436:     sv_frame_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@102:     bool looping = m_viewManager->getPlayLoopMode();
Chris@102: 
Chris@553:     RealTime inbuffer_t = RealTime::frame2RealTime(inbuffer, rate);
Chris@91: 
Chris@436:     sv_frame_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@553:     RealTime stretchlat_t = RealTime::frame2RealTime(stretchlat, rate);
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@102:     if (m_target &&
Chris@102:         m_trustworthyTimestamps &&
Chris@102:         lastRetrievalTimestamp != 0.0) {
Chris@91: 
Chris@553:         lastretrieved_t = RealTime::frame2RealTime(lastRetrievedBlockSize, rate);
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@102:         if (m_trustworthyTimestamps && !looping) {
Chris@91: 
Chris@102:             // this adjustment seems to cause more problems when looping
Chris@102:             double elapsed = currentTime - lastRetrievalTimestamp;
Chris@102: 
Chris@102:             if (elapsed > 0.0) {
Chris@102:                 sincerequest_t = RealTime::fromSeconds(elapsed);
Chris@102:             }
Chris@91:         }
Chris@91: 
Chris@91:     } else {
Chris@91: 
Chris@553:         lastretrieved_t = RealTime::frame2RealTime(getTargetBlockSize(), rate);
Chris@62:     }
Chris@91: 
Chris@553:     RealTime bufferedto_t = RealTime::frame2RealTime(readBufferFill, rate);
Chris@91: 
Chris@91:     if (timeRatio != 1.0) {
Chris@91:         lastretrieved_t = lastretrieved_t / timeRatio;
Chris@91:         sincerequest_t = sincerequest_t / timeRatio;
Chris@163:         latency_t = latency_t / timeRatio;
Chris@43:     }
Chris@43: 
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     cout << "\nbuffered to: " << bufferedto_t << ", in buffer: " << inbuffer_t << ", time ratio " << timeRatio << "\n  stretcher latency: " << stretchlat_t << ", device latency: " << latency_t << "\n  since request: " << sincerequest_t << ", last retrieved quantity: " << lastretrieved_t << endl;
Chris@91: #endif
Chris@43: 
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@193:         if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;
Chris@553:         sv_frame_t frame = RealTime::realTime2Frame(playing_t, rate);
Chris@93:         return m_viewManager->alignPlaybackFrameToReference(frame);
Chris@93:     }
Chris@43: 
Chris@91:     int inRange = 0;
Chris@91:     int index = 0;
Chris@91: 
Chris@366:     for (int i = 0; i < (int)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@436:     if (inRange >= int(m_rangeStarts.size())) {
Chris@436:         inRange = int(m_rangeStarts.size())-1;
Chris@436:     }
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@553:         RealTime playstart_t = RealTime::frame2RealTime(m_playStartFrame, rate);
Chris@94:         if (playing_t < playstart_t) {
Chris@563: //            cout << "playing_t " << playing_t << " < playstart_t " 
Chris@293: //                      << playstart_t << endl;
Chris@122:             if (/*!!! sincerequest_t > RealTime::zeroTime && */
Chris@94:                 m_playStartedAt + latency_t + stretchlat_t <
Chris@94:                 RealTime::fromSeconds(currentTime)) {
Chris@563: //                cout << "but we've been playing for long enough that I think we should disregard it (it probably results from loop wrapping)" << 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@163:  
Chris@163: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     cout << "playing_t " << playing_t;
Chris@163: #endif
Chris@94: 
Chris@94:     playing_t = playing_t - m_rangeStarts[inRange];
Chris@93:  
Chris@93: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     cout << " as offset into range " << inRange << " (start =" << m_rangeStarts[inRange] << " duration =" << m_rangeDurations[inRange] << ") = " << playing_t << 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@436:                 inRange = int(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@563:     cout << "  playing time: " << playing_t << endl;
Chris@93: #endif
Chris@93: 
Chris@93:     if (!looping) {
Chris@366:         if (inRange == (int)m_rangeStarts.size()-1 &&
Chris@93:             playing_t >= m_rangeStarts[inRange] + m_rangeDurations[inRange]) {
Chris@563: cout << "Not looping, inRange " << inRange << " == rangeStarts.size()-1, playing_t " << playing_t << " >= m_rangeStarts[inRange] " << m_rangeStarts[inRange] << " + m_rangeDurations[inRange] " << m_rangeDurations[inRange] << " -- stopping" << endl;
Chris@93:             stop();
Chris@93:         }
Chris@93:     }
Chris@93: 
Chris@93:     if (playing_t < RealTime::zeroTime) playing_t = RealTime::zeroTime;
Chris@93: 
Chris@553:     sv_frame_t frame = RealTime::realTime2Frame(playing_t, rate);
Chris@102: 
Chris@102:     if (m_lastCurrentFrame > 0 && !looping) {
Chris@102:         if (frame < m_lastCurrentFrame) {
Chris@102:             frame = m_lastCurrentFrame;
Chris@102:         }
Chris@102:     }
Chris@102: 
Chris@102:     m_lastCurrentFrame = frame;
Chris@102: 
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@436:     sv_samplerate_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@233:     SVDEBUG << "AudioCallbackPlaySource::rebuildRangeLists" << 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@563:     cout << "Now have " << m_rangeStarts.size() << " play ranges" << endl;
Chris@91: #endif
Chris@43: }
Chris@43: 
Chris@43: void
Chris@43: AudioCallbackPlaySource::setOutputLevels(float left, float right)
Chris@43: {
Chris@574:     if (left > m_outputLeft) m_outputLeft = left;
Chris@574:     if (right > m_outputRight) 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@574:     m_outputLeft = 0.f;
Chris@574:     m_outputRight = 0.f;
Chris@43:     return true;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@468: AudioCallbackPlaySource::setSystemPlaybackSampleRate(int sr)
Chris@43: {
Chris@553:     m_deviceSampleRate = sr;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@559: AudioCallbackPlaySource::setSystemPlaybackChannelCount(int count)
Chris@43: {
Chris@559:     m_deviceChannelCount = count;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@107: AudioCallbackPlaySource::setAuditioningEffect(Auditionable *a)
Chris@43: {
Chris@107:     RealTimePluginInstance *plugin = dynamic_cast<RealTimePluginInstance *>(a);
Chris@107:     if (a && !plugin) {
Chris@563:         SVCERR << "WARNING: AudioCallbackPlaySource::setAuditioningEffect: auditionable object " << a << " is not a real-time plugin instance" << endl;
Chris@107:     }
Chris@204: 
Chris@204:     m_mutex.lock();
Chris@43:     m_auditioningPlugin = plugin;
Chris@43:     m_auditioningPluginBypassed = false;
Chris@204:     m_mutex.unlock();
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@434: sv_samplerate_t
Chris@553: AudioCallbackPlaySource::getDeviceSampleRate() const
Chris@43: {
Chris@553:     return m_deviceSampleRate;
Chris@43: }
Chris@43: 
Chris@366: int
Chris@43: AudioCallbackPlaySource::getSourceChannelCount() const
Chris@43: {
Chris@43:     return m_sourceChannelCount;
Chris@43: }
Chris@43: 
Chris@366: int
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@559: int
Chris@559: AudioCallbackPlaySource::getDeviceChannelCount() const
Chris@559: {
Chris@559:     return m_deviceChannelCount;
Chris@559: }
Chris@559: 
Chris@434: sv_samplerate_t
Chris@43: AudioCallbackPlaySource::getSourceSampleRate() const
Chris@43: {
Chris@43:     return m_sourceSampleRate;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@436: AudioCallbackPlaySource::setTimeStretch(double factor)
Chris@43: {
Chris@91:     m_stretchRatio = factor;
Chris@91: 
Chris@553:     int rate = int(getSourceSampleRate());
Chris@553:     if (!rate) return; // have to make our stretcher later
Chris@244: 
Chris@436:     if (m_timeStretcher || (factor == 1.0)) {
Chris@91:         // stretch ratio will be set in next process call if appropriate
Chris@62:     } else {
Chris@91:         m_stretcherInputCount = getTargetChannelCount();
Chris@62:         RubberBandStretcher *stretcher = new RubberBandStretcher
Chris@553:             (rate,
Chris@91:              m_stretcherInputCount,
Chris@62:              RubberBandStretcher::OptionProcessRealTime,
Chris@62:              factor);
Chris@130:         RubberBandStretcher *monoStretcher = new RubberBandStretcher
Chris@553:             (rate,
Chris@130:              1,
Chris@130:              RubberBandStretcher::OptionProcessRealTime,
Chris@130:              factor);
Chris@91:         m_stretcherInputs = new float *[m_stretcherInputCount];
Chris@436:         m_stretcherInputSizes = new sv_frame_t[m_stretcherInputCount];
Chris@366:         for (int 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@130:         m_monoStretcher = monoStretcher;
Chris@62:         m_timeStretcher = stretcher;
Chris@62:     }
Chris@158: 
Chris@158:     emit activity(tr("Change time-stretch factor to %1").arg(factor));
Chris@43: }
Chris@43: 
Chris@471: int
Chris@559: AudioCallbackPlaySource::getSourceSamples(float *const *buffer,
Chris@559:                                           int requestedChannels,
Chris@559:                                           int count)
Chris@43: {
Chris@559:     // In principle, the target will handle channel mapping in cases
Chris@559:     // where our channel count differs from the device's. But that
Chris@559:     // only holds if our channel count doesn't change -- i.e. if
Chris@559:     // getApplicationChannelCount() always returns the same value as
Chris@559:     // it did when the target was created, and if this function always
Chris@559:     // returns that number of channels.
Chris@559:     //
Chris@559:     // Unfortunately that can't hold for us -- we always have at least
Chris@559:     // 2 channels but if the user opens a new main model with more
Chris@559:     // channels than that (and more than the last main model) then our
Chris@559:     // target channel count necessarily gets increased.
Chris@559:     //
Chris@559:     // We have:
Chris@559:     // 
Chris@559:     // getSourceChannelCount() -> number of channels available to
Chris@559:     // provide from real model data
Chris@559:     //
Chris@559:     // getTargetChannelCount() -> number we will actually provide;
Chris@559:     // same as getSourceChannelCount() except that it is always at
Chris@559:     // least 2
Chris@559:     //
Chris@559:     // getDeviceChannelCount() -> number the device will emit, usually
Chris@559:     // equal to the value of getTargetChannelCount() at the time the
Chris@559:     // device was initialised, unless the device could not provide
Chris@559:     // that number
Chris@559:     //
Chris@559:     // requestedChannels -> number the device is expecting from us,
Chris@559:     // always equal to the value of getTargetChannelCount() at the
Chris@559:     // time the device was initialised
Chris@559:     //
Chris@559:     // If the requested channel count is at least the target channel
Chris@559:     // count, then we go ahead and provide the target channels as
Chris@559:     // expected. We just zero any spare channels.
Chris@559:     //
Chris@559:     // If the requested channel count is smaller than the target
Chris@559:     // channel count, then we don't know what to do and we provide
Chris@559:     // nothing. This shouldn't happen as long as management is on the
Chris@559:     // ball -- we emit channelCountIncreased() when the target channel
Chris@559:     // count increases, and whatever code "owns" the driver should
Chris@559:     // have reopened the audio device when it got that signal. But
Chris@559:     // there's a race condition there, which we accommodate with this
Chris@559:     // check.
Chris@559: 
Chris@559:     int channels = getTargetChannelCount();
Chris@559: 
Chris@43:     if (!m_playing) {
Chris@193: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:         cout << "AudioCallbackPlaySource::getSourceSamples: Not playing" << endl;
Chris@193: #endif
Chris@559:         v_zero_channels(buffer, requestedChannels, count);
Chris@471: 	return 0;
Chris@43:     }
Chris@559:     if (requestedChannels < channels) {
Chris@559:         SVDEBUG << "AudioCallbackPlaySource::getSourceSamples: Not enough device channels (" << requestedChannels << ", need " << channels << "); hoping device is about to be reopened" << endl;
Chris@559:         v_zero_channels(buffer, requestedChannels, count);
Chris@559:         return 0;
Chris@559:     }
Chris@559:     if (requestedChannels > channels) {
Chris@559:         v_zero_channels(buffer + channels, requestedChannels - channels, count);
Chris@559:     }
Chris@43: 
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     cout << "AudioCallbackPlaySource::getSourceSamples: Playing" << endl;
Chris@212: #endif
Chris@212: 
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@559:     for (int ch = 0; ch < channels; ++ch) {
Chris@43: 
Chris@43:         RingBuffer<float> *rb = getReadRingBuffer(ch);
Chris@43:         
Chris@43:         if (!rb) {
Chris@563:             SVCERR << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
Chris@43:                       << "No ring buffer available for channel " << ch
Chris@293:                       << ", returning no data here" << endl;
Chris@43:             count = 0;
Chris@43:             break;
Chris@43:         }
Chris@43: 
Chris@366:         int rs = rb->getReadSpace();
Chris@43:         if (rs < count) {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:             cerr << "WARNING: AudioCallbackPlaySource::getSourceSamples: "
Chris@43:                       << "Ring buffer for channel " << ch << " has only "
Chris@193:                       << rs << " (of " << count << ") samples available ("
Chris@193:                       << "ring buffer size is " << rb->getSize() << ", write "
Chris@193:                       << "space " << rb->getWriteSpace() << "), "
Chris@293:                       << "reducing request size" << endl;
Chris@43: #endif
Chris@43:             count = rs;
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@471:     if (count == 0) return 0;
Chris@43: 
Chris@62:     RubberBandStretcher *ts = m_timeStretcher;
Chris@130:     RubberBandStretcher *ms = m_monoStretcher;
Chris@130: 
Chris@436:     double ratio = ts ? ts->getTimeRatio() : 1.0;
Chris@91: 
Chris@91:     if (ratio != m_stretchRatio) {
Chris@91:         if (!ts) {
Chris@563:             SVCERR << "WARNING: AudioCallbackPlaySource::getSourceSamples: Time ratio change to " << m_stretchRatio << " is pending, but no stretcher is set" << endl;
Chris@436:             m_stretchRatio = 1.0;
Chris@91:         } else {
Chris@91:             ts->setTimeRatio(m_stretchRatio);
Chris@130:             if (ms) ms->setTimeRatio(m_stretchRatio);
Chris@130:             if (m_stretchRatio >= 1.0) m_stretchMono = false;
Chris@130:         }
Chris@130:     }
Chris@130: 
Chris@130:     int stretchChannels = m_stretcherInputCount;
Chris@130:     if (m_stretchMono) {
Chris@130:         if (ms) {
Chris@130:             ts = ms;
Chris@130:             stretchChannels = 1;
Chris@130:         } else {
Chris@130:             m_stretchMono = false;
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@130: 	int got = 0;
Chris@43: 
Chris@563: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:         cout << "channels == " << channels << endl;
Chris@563: #endif
Chris@555:         
Chris@559: 	for (int ch = 0; ch < channels; ++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@436: 		sv_frame_t request = count;
Chris@43: 		if (ch > 0) request = got;
Chris@43: 
Chris@436: 		got = rb->read(buffer[ch], int(request));
Chris@43: 	    
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@293: 		cout << "AudioCallbackPlaySource::getSamples: got " << got << " (of " << count << ") samples on channel " << ch << ", signalling for more (possibly)" << endl;
Chris@43: #endif
Chris@43: 	    }
Chris@43: 
Chris@559: 	    for (int ch = 0; ch < channels; ++ch) {
Chris@130: 		for (int 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@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "AudioCallbackPlaySource::getSamples: awakening thread" << endl;
Chris@212: #endif
Chris@212: 
Chris@43:         m_condition.wakeAll();
Chris@91: 
Chris@471: 	return got;
Chris@43:     }
Chris@43: 
Chris@436:     sv_frame_t available;
Chris@436:     sv_frame_t fedToStretcher = 0;
Chris@91:     int warned = 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@436:         sv_frame_t reqd = lrint(double(count - available) / ratio);
Chris@436:         reqd = std::max(reqd, sv_frame_t(ts->getSamplesRequired()));
Chris@91:         if (reqd == 0) reqd = 1;
Chris@91:                 
Chris@436:         sv_frame_t got = reqd;
Chris@91: 
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:         cout << "reqd = " <<reqd << ", channels = " << channels << ", ic = " << m_stretcherInputCount << endl;
Chris@62: #endif
Chris@43: 
Chris@366:         for (int c = 0; c < channels; ++c) {
Chris@131:             if (c >= m_stretcherInputCount) continue;
Chris@91:             if (reqd > m_stretcherInputSizes[c]) {
Chris@91:                 if (c == 0) {
Chris@563:                     SVDEBUG << "NOTE: resizing stretcher input buffer from " << m_stretcherInputSizes[c] << " to " << (reqd * 2) << 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@366:         for (int c = 0; c < channels; ++c) {
Chris@131:             if (c >= m_stretcherInputCount) continue;
Chris@91:             RingBuffer<float> *rb = getReadRingBuffer(c);
Chris@91:             if (rb) {
Chris@436:                 sv_frame_t gotHere;
Chris@130:                 if (stretchChannels == 1 && c > 0) {
Chris@436:                     gotHere = rb->readAdding(m_stretcherInputs[0], int(got));
Chris@130:                 } else {
Chris@436:                     gotHere = rb->read(m_stretcherInputs[c], int(got));
Chris@130:                 }
Chris@91:                 if (gotHere < got) got = gotHere;
Chris@91:                 
Chris@91: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@91:                 if (c == 0) {
Chris@563:                     cout << "feeding stretcher: got " << gotHere
Chris@229:                               << ", " << rb->getReadSpace() << " remain" << endl;
Chris@91:                 }
Chris@62: #endif
Chris@43:                 
Chris@91:             } else {
Chris@563:                 SVCERR << "WARNING: No ring buffer available for channel " << c << " in stretcher input block" << endl;
Chris@43:             }
Chris@43:         }
Chris@43: 
Chris@43:         if (got < reqd) {
Chris@563:             SVCERR << "WARNING: Read underrun in playback ("
Chris@293:                       << got << " < " << reqd << ")" << endl;
Chris@43:         }
Chris@43: 
Chris@463:         ts->process(m_stretcherInputs, size_t(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@563:             SVCERR << "WARNING: AudioCallbackPlaySource::getSamples: Added " << got << " samples to time stretcher, created no new available output samples (warned = " << warned << ")" << endl;
Chris@43:             if (++warned == 5) break;
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@463:     ts->retrieve(buffer, size_t(count));
Chris@43: 
Chris@559:     v_zero_channels(buffer + stretchChannels, channels - stretchChannels, count);
Chris@130: 
Chris@43:     applyAuditioningEffect(count, buffer);
Chris@43: 
Chris@212: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "AudioCallbackPlaySource::getSamples [stretched]: awakening thread" << endl;
Chris@212: #endif
Chris@212: 
Chris@43:     m_condition.wakeAll();
Chris@43: 
Chris@471:     return count;
Chris@43: }
Chris@43: 
Chris@43: void
Chris@559: AudioCallbackPlaySource::applyAuditioningEffect(sv_frame_t count, float *const *buffers)
Chris@43: {
Chris@43:     if (m_auditioningPluginBypassed) return;
Chris@43:     RealTimePluginInstance *plugin = m_auditioningPlugin;
Chris@43:     if (!plugin) return;
Chris@204:     
Chris@366:     if ((int)plugin->getAudioInputCount() != getTargetChannelCount()) {
Chris@563: //        cout << "plugin input count " << plugin->getAudioInputCount() 
Chris@43: //                  << " != our channel count " << getTargetChannelCount()
Chris@293: //                  << endl;
Chris@43:         return;
Chris@43:     }
Chris@366:     if ((int)plugin->getAudioOutputCount() != getTargetChannelCount()) {
Chris@563: //        cout << "plugin output count " << plugin->getAudioOutputCount() 
Chris@43: //                  << " != our channel count " << getTargetChannelCount()
Chris@293: //                  << endl;
Chris@43:         return;
Chris@43:     }
Chris@366:     if ((int)plugin->getBufferSize() < count) {
Chris@563: //        cout << "plugin buffer size " << plugin->getBufferSize() 
Chris@102: //                  << " < our block size " << count
Chris@293: //                  << endl;
Chris@43:         return;
Chris@43:     }
Chris@43: 
Chris@43:     float **ib = plugin->getAudioInputBuffers();
Chris@43:     float **ob = plugin->getAudioOutputBuffers();
Chris@43: 
Chris@366:     for (int c = 0; c < getTargetChannelCount(); ++c) {
Chris@366:         for (int i = 0; i < count; ++i) {
Chris@43:             ib[c][i] = buffers[c][i];
Chris@43:         }
Chris@43:     }
Chris@43: 
Chris@436:     plugin->run(Vamp::RealTime::zeroTime, int(count));
Chris@43:     
Chris@366:     for (int c = 0; c < getTargetChannelCount(); ++c) {
Chris@366:         for (int 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@436:     static sv_frame_t tmpSize = 0;
Chris@43: 
Chris@434:     sv_frame_t space = 0;
Chris@366:     for (int c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	if (wb) {
Chris@434: 	    sv_frame_t spaceHere = wb->getWriteSpace();
Chris@43: 	    if (c == 0 || spaceHere < space) space = spaceHere;
Chris@43: 	}
Chris@43:     }
Chris@43:     
Chris@103:     if (space == 0) {
Chris@103: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:         cout << "AudioCallbackPlaySourceFillThread: no space to fill" << endl;
Chris@103: #endif
Chris@103:         return false;
Chris@103:     }
Chris@43: 
Chris@544:     // space is now the number of samples that can be written on each
Chris@544:     // channel's write ringbuffer
Chris@544:     
Chris@434:     sv_frame_t f = m_writeBufferFill;
Chris@43: 	
Chris@43:     bool readWriteEqual = (m_readBuffers == m_writeBuffers);
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@193:     if (!readWriteEqual) {
Chris@293:         cout << "AudioCallbackPlaySourceFillThread: note read buffers != write buffers" << endl;
Chris@193:     }
Chris@293:     cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << endl;
Chris@43: #endif
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:     cout << "buffered to " << f << " already" << endl;
Chris@43: #endif
Chris@43: 
Chris@366:     int channels = getTargetChannelCount();
Chris@43: 
Chris@43:     static float **bufferPtrs = 0;
Chris@366:     static int 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@436:     sv_frame_t generatorBlockSize = m_audioGenerator->getBlockSize();
Chris@43: 
Chris@546:     // space must be a multiple of generatorBlockSize
Chris@546:     sv_frame_t reqSpace = space;
Chris@546:     space = (reqSpace / generatorBlockSize) * generatorBlockSize;
Chris@546:     if (space == 0) {
Chris@546: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@546:         cout << "requested fill of " << reqSpace
Chris@546:              << " is less than generator block size of "
Chris@546:              << generatorBlockSize << ", leaving it" << endl;
Chris@546: #endif
Chris@546:         return false;
Chris@43:     }
Chris@43: 
Chris@546:     if (tmpSize < channels * space) {
Chris@546:         delete[] tmp;
Chris@546:         tmp = new float[channels * space];
Chris@546:         tmpSize = channels * space;
Chris@546:     }
Chris@43: 
Chris@546:     for (int c = 0; c < channels; ++c) {
Chris@43: 
Chris@546:         bufferPtrs[c] = tmp + c * space;
Chris@546: 	    
Chris@546:         for (int i = 0; i < space; ++i) {
Chris@546:             tmp[c * space + i] = 0.0f;
Chris@546:         }
Chris@546:     }
Chris@43: 
Chris@546:     sv_frame_t got = mixModels(f, space, bufferPtrs); // also modifies f
Chris@43: 
Chris@546:     for (int c = 0; c < channels; ++c) {
Chris@43: 
Chris@546:         RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@546:         if (wb) {
Chris@546:             int actual = wb->write(bufferPtrs[c], int(got));
Chris@546: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@546:             cout << "Wrote " << actual << " samples for ch " << c << ", now "
Chris@546:                  << wb->getReadSpace() << " to read" 
Chris@546:                  << endl;
Chris@546: #endif
Chris@546:             if (actual < got) {
Chris@563:                 SVCERR << "WARNING: Buffer overrun in channel " << c
Chris@563:                        << ": wrote " << actual << " of " << got
Chris@563:                        << " samples" << endl;
Chris@546:             }
Chris@546:         }
Chris@546:     }
Chris@43: 
Chris@546:     m_writeBufferFill = f;
Chris@546:     if (readWriteEqual) m_readBufferFill = f;
Chris@43: 
Chris@163: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << "Read buffer fill is now " << m_readBufferFill << ", write buffer fill "
Chris@563:          << m_writeBufferFill << endl;
Chris@163: #endif
Chris@163: 
Chris@546:     //!!! 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:     return true;
Chris@43: }    
Chris@43: 
Chris@434: sv_frame_t
Chris@434: AudioCallbackPlaySource::mixModels(sv_frame_t &frame, sv_frame_t count, float **buffers)
Chris@43: {
Chris@434:     sv_frame_t processed = 0;
Chris@434:     sv_frame_t chunkStart = frame;
Chris@434:     sv_frame_t chunkSize = count;
Chris@434:     sv_frame_t selectionSize = 0;
Chris@434:     sv_frame_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@366:     int channels = getTargetChannelCount();
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563:     cout << "mixModels: start " << frame << ", size " << count << ", channels " << channels << endl;
Chris@43: #endif
Chris@563: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     if (constrained) {
Chris@563:         cout << "Manager has " << m_viewManager->getSelections().size() << " selection(s):" << endl;
Chris@563:         for (auto sel: m_viewManager->getSelections()) {
Chris@563:             cout << sel.getStartFrame() << " -> " << sel.getEndFrame()
Chris@563:                  << " (" << (sel.getEndFrame() - sel.getStartFrame()) << " frames)"
Chris@563:                  << endl;
Chris@563:         }
Chris@563:     }
Chris@563: #endif
Chris@563:     
Chris@563:     static float **chunkBufferPtrs = 0;
Chris@563:     static int chunkBufferPtrCount = 0;
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@366:     for (int 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@434: 	sv_frame_t fadeIn = 0, fadeOut = 0;
Chris@43: 
Chris@43: 	if (constrained) {
Chris@60: 
Chris@434:             sv_frame_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@434:                 sv_frame_t sf = m_viewManager->alignReferenceToPlaybackFrame
Chris@60:                     (selection.getStartFrame());
Chris@434:                 sv_frame_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@563: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563: 	cout << "chunkStart " << chunkStart << ", chunkSize " << chunkSize << ", nextChunkStart " << nextChunkStart << ", frame " << frame << ", count " << count << ", processed " << processed << endl;
Chris@563: #endif
Chris@563:         
Chris@43: 	if (!chunkSize) {
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@562:             if (frame < nextChunkStart) {
Chris@562:                 frame = nextChunkStart;
Chris@562:             }
Chris@562: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@562: 	    cout << "mixModels: ending at " << nextChunkStart << ", returning frame as "
Chris@562:                  << frame << endl;
Chris@562: #endif
Chris@43: 	    return count;
Chris@43: 	}
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@563: 	cout << "mixModels: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << endl;
Chris@43: #endif
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@366: 	    (void) m_audioGenerator->mixModel(*mi, chunkStart, 
Chris@366:                                               chunkSize, chunkBufferPtrs,
Chris@366:                                               fadeIn, fadeOut);
Chris@43: 	}
Chris@43: 
Chris@366: 	for (int 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@563:     cout << "mixModels returning " << processed << " frames to " << nextChunkStart << 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@366:     for (int 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@193: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:                     cout << "AudioCallbackPlaySource::unifyRingBuffers: Not unifying: write buffer has less (" << wb->getReadSpace() << ") than " << m_blockSize*2 << " to read and write buffer fill (" << m_writeBufferFill << ") is not close to end frame (" << m_lastModelEndFrame << ")" << endl;
Chris@193: #endif
Chris@43: 		    return;
Chris@43: 		}
Chris@43: 	    }
Chris@43: 	    break;
Chris@43: 	}
Chris@43:     }
Chris@43: 
Chris@436:     sv_frame_t rf = m_readBufferFill;
Chris@43:     RingBuffer<float> *rb = getReadRingBuffer(0);
Chris@43:     if (rb) {
Chris@366: 	int rs = rb->getReadSpace();
Chris@43: 	//!!! incorrect when in non-contiguous selection, see comments elsewhere
Chris@293: //	cout << "rs = " << rs << endl;
Chris@43: 	if (rs < rf) rf -= rs;
Chris@43: 	else rf = 0;
Chris@43:     }
Chris@43:     
Chris@193: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     cout << "AudioCallbackPlaySource::unifyRingBuffers: m_readBufferFill = " << m_readBufferFill << ", rf = " << rf << ", m_writeBufferFill = " << m_writeBufferFill << endl;
Chris@193: #endif
Chris@43: 
Chris@436:     sv_frame_t wf = m_writeBufferFill;
Chris@436:     sv_frame_t skip = 0;
Chris@366:     for (int c = 0; c < getTargetChannelCount(); ++c) {
Chris@43: 	RingBuffer<float> *wb = getWriteRingBuffer(c);
Chris@43: 	if (wb) {
Chris@43: 	    if (c == 0) {
Chris@43: 		
Chris@366: 		int wrs = wb->getReadSpace();
Chris@293: //		cout << "wrs = " << wrs << endl;
Chris@43: 
Chris@43: 		if (wrs < wf) wf -= wrs;
Chris@43: 		else wf = 0;
Chris@293: //		cout << "wf = " << wf << endl;
Chris@43: 		
Chris@43: 		if (wf < rf) skip = rf - wf;
Chris@43: 		if (skip == 0) break;
Chris@43: 	    }
Chris@43: 
Chris@293: //	    cout << "skipping " << skip << endl;
Chris@436: 	    wb->skip(int(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@193: #ifdef DEBUG_AUDIO_PLAY_SOURCE_PLAYING
Chris@563:     cout << "unified" << endl;
Chris@193: #endif
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@293:     cout << "AudioCallbackPlaySourceFillThread starting" << 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@293: 	    cout << "AudioCallbackPlaySourceFillThread: not waiting" << 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@436: 	    double ms = 100;
Chris@43: 	    if (s.getSourceSampleRate() > 0) {
Chris@436: 		ms = double(s.m_ringBufferSize) / 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@293:             if (!s.m_playing) cout << endl;
Chris@293: 	    cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms << "ms..." << endl;
Chris@43: #endif
Chris@43: 	    
Chris@366: 	    s.m_condition.wait(&s.m_mutex, int(ms));
Chris@43: 	}
Chris@43: 
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293: 	cout << "AudioCallbackPlaySourceFillThread: awoken" << endl;
Chris@43: #endif
Chris@43: 
Chris@43: 	work = false;
Chris@43: 
Chris@103: 	if (!s.getSourceSampleRate()) {
Chris@103: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293:             cout << "AudioCallbackPlaySourceFillThread: source sample rate is zero" << endl;
Chris@103: #endif
Chris@103:             continue;
Chris@103:         }
Chris@43: 
Chris@43: 	bool playing = s.m_playing;
Chris@43: 
Chris@43: 	if (playing && !previouslyPlaying) {
Chris@43: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@293: 	    cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << endl;
Chris@43: #endif
Chris@366: 	    for (int 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: