Chris@0: /* -*- c-basic-offset: 4 -*-  vi:set ts=8 sts=4 sw=4: */
Chris@0: 
Chris@0: /*
Chris@0:     A waveform viewer and audio annotation editor.
Chris@1:     Chris Cannam, Queen Mary University of London, 2005-2006
Chris@0:     
Chris@0:     This is experimental software.  Not for distribution.
Chris@0: */
Chris@0: 
Chris@0: #include "AudioCallbackPlaySource.h"
Chris@0: 
Chris@0: #include "AudioGenerator.h"
Chris@0: 
Chris@0: #include "base/Model.h"
Chris@0: #include "base/ViewManager.h"
Chris@0: #include "model/DenseTimeValueModel.h"
Chris@0: #include "model/SparseOneDimensionalModel.h"
Chris@0: #include "dsp/timestretching/IntegerTimeStretcher.h"
Chris@0: 
Chris@0: #include <iostream>
Chris@0: 
Chris@0: //#define DEBUG_AUDIO_PLAY_SOURCE 1
Chris@0: 
Chris@0: //const size_t AudioCallbackPlaySource::m_ringBufferSize = 102400;
Chris@0: const size_t AudioCallbackPlaySource::m_ringBufferSize = 131071;
Chris@0: 
Chris@0: AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManager *manager) :
Chris@0:     m_viewManager(manager),
Chris@0:     m_audioGenerator(new AudioGenerator(manager)),
Chris@0:     m_bufferCount(0),
Chris@0:     m_blockSize(1024),
Chris@0:     m_sourceSampleRate(0),
Chris@0:     m_targetSampleRate(0),
Chris@0:     m_playLatency(0),
Chris@0:     m_playing(false),
Chris@0:     m_exiting(false),
Chris@0:     m_bufferedToFrame(0),
Chris@0:     m_outputLeft(0.0),
Chris@0:     m_outputRight(0.0),
Chris@0:     m_slowdownCounter(0),
Chris@0:     m_timeStretcher(0),
Chris@0:     m_fillThread(0),
Chris@0:     m_converter(0)
Chris@0: {
Chris@0:     // preallocate some slots, to avoid reallocation in an
Chris@0:     // un-thread-safe manner later
Chris@0:     while (m_buffers.size() < 20) m_buffers.push_back(0);
Chris@0: 
Chris@0:     m_viewManager->setAudioPlaySource(this);
Chris@3: 
Chris@3:     connect(m_viewManager, SIGNAL(selectionChanged()),
Chris@3: 	    this, SLOT(selectionChanged()));
Chris@3:     connect(m_viewManager, SIGNAL(playLoopModeChanged()),
Chris@3: 	    this, SLOT(playLoopModeChanged()));
Chris@3:     connect(m_viewManager, SIGNAL(playSelectionModeChanged()),
Chris@3: 	    this, SLOT(playSelectionModeChanged()));
Chris@0: }
Chris@0: 
Chris@0: AudioCallbackPlaySource::~AudioCallbackPlaySource()
Chris@0: {
Chris@0:     m_exiting = true;
Chris@0: 
Chris@0:     if (m_fillThread) {
Chris@0: 	m_condition.wakeAll();
Chris@0: 	m_fillThread->wait();
Chris@0: 	delete m_fillThread;
Chris@0:     }
Chris@0: 
Chris@0:     clearModels();
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::addModel(Model *model)
Chris@0: {
Chris@0:     m_mutex.lock();
Chris@0: 
Chris@0:     m_models.insert(model);
Chris@0: 
Chris@0:     bool buffersChanged = false, srChanged = false;
Chris@0: 
Chris@0:     if (m_sourceSampleRate == 0) {
Chris@0: 
Chris@0: 	m_sourceSampleRate = model->getSampleRate();
Chris@0: 	srChanged = true;
Chris@0: 
Chris@0:     } else if (model->getSampleRate() != m_sourceSampleRate) {
Chris@0: 	std::cerr << "AudioCallbackPlaySource::addModel: ERROR: "
Chris@0: 		  << "New model sample rate does not match" << std::endl
Chris@0: 		  << "existing model(s) (new " << model->getSampleRate()
Chris@0: 		  << " vs " << m_sourceSampleRate
Chris@0: 		  << "), playback will be wrong"
Chris@0: 		  << std::endl;
Chris@0:     }
Chris@0: 
Chris@0:     size_t sz = m_ringBufferSize;
Chris@0:     if (m_bufferCount > 0) {
Chris@0: 	sz = m_buffers[0]->getSize();
Chris@0:     }
Chris@0: 
Chris@0:     size_t modelChannels = 1;
Chris@0:     DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
Chris@0:     if (dtvm) modelChannels = dtvm->getChannelCount();
Chris@0: 
Chris@0:     while (m_bufferCount < modelChannels) {
Chris@0: 
Chris@0: 	if (m_buffers.size() < modelChannels) {
Chris@0: 	    // This is a hideously chancy operation -- the RT thread
Chris@0: 	    // could be using this vector.  We allocated several slots
Chris@0: 	    // in the ctor to avoid exactly this, but if we ever end
Chris@0: 	    // up with more channels than that (!) then we're just
Chris@0: 	    // going to have to risk it
Chris@0: 	    m_buffers.push_back(new RingBuffer<float>(sz));
Chris@0: 
Chris@0: 	} else {
Chris@0: 	    // The usual case
Chris@0: 	    m_buffers[m_bufferCount] = new RingBuffer<float>(sz);
Chris@0: 	}
Chris@0: 
Chris@0: 	++m_bufferCount;
Chris@0: 	buffersChanged = true;
Chris@0:     }
Chris@0: 
Chris@0:     if (buffersChanged) {
Chris@0: 	m_audioGenerator->setTargetChannelCount(m_bufferCount);
Chris@0:     }
Chris@0: 
Chris@0:     if (buffersChanged || srChanged) {
Chris@0: 	if (m_converter) {
Chris@0: 	    src_delete(m_converter);
Chris@0: 	    m_converter = 0;
Chris@0: 	}
Chris@0:     }
Chris@0: 
Chris@0:     m_audioGenerator->addModel(model);
Chris@0: 
Chris@0:     m_mutex.unlock();
Chris@0: 
Chris@0:     if (!m_fillThread) {
Chris@0: 	m_fillThread = new AudioCallbackPlaySourceFillThread(*this);
Chris@0: 	m_fillThread->start();
Chris@0:     }
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0:     std::cerr << "AudioCallbackPlaySource::addModel: emitting modelReplaced" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@1:     if (buffersChanged || srChanged) {
Chris@1: 	emit modelReplaced();
Chris@0:     }
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::removeModel(Model *model)
Chris@0: {
Chris@0:     m_mutex.lock();
Chris@0: 
Chris@0:     m_models.erase(model);
Chris@0: 
Chris@0:     if (m_models.empty()) {
Chris@0: 	if (m_converter) {
Chris@0: 	    src_delete(m_converter);
Chris@0: 	    m_converter = 0;
Chris@0: 	}
Chris@0: 	m_sourceSampleRate = 0;
Chris@0:     }
Chris@0: 
Chris@0:     m_audioGenerator->removeModel(model);
Chris@0: 
Chris@0:     m_mutex.unlock();
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::clearModels()
Chris@0: {
Chris@0:     m_mutex.lock();
Chris@0: 
Chris@0:     m_models.clear();
Chris@0: 
Chris@0:     if (m_converter) {
Chris@0: 	src_delete(m_converter);
Chris@0: 	m_converter = 0;
Chris@0:     }
Chris@0: 
Chris@0:     m_audioGenerator->clearModels();
Chris@0: 
Chris@0:     m_sourceSampleRate = 0;
Chris@0: 
Chris@0:     m_mutex.unlock();
Chris@0: }    
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::play(size_t startFrame)
Chris@0: {
Chris@0:     // The fill thread will automatically empty its buffers before
Chris@0:     // starting again if we have not so far been playing, but not if
Chris@0:     // we're just re-seeking.
Chris@0: 
Chris@0:     if (m_playing) {
Chris@0: 	m_mutex.lock();
Chris@0: 	m_bufferedToFrame = startFrame;
Chris@0: 	for (size_t c = 0; c < m_bufferCount; ++c) {
Chris@0: 	    getRingBuffer(c).reset();
Chris@0: 	    if (m_converter) src_reset(m_converter);
Chris@0: 	}
Chris@0: 	m_mutex.unlock();
Chris@0:     } else {
Chris@0: 	m_bufferedToFrame = startFrame;
Chris@0:     }
Chris@0: 
Chris@0:     m_audioGenerator->reset();
Chris@0: 
Chris@0:     m_playing = true;
Chris@0:     m_condition.wakeAll();
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::stop()
Chris@0: {
Chris@0:     m_playing = false;
Chris@0:     m_condition.wakeAll();
Chris@0: }
Chris@0: 
Chris@0: void
Chris@3: AudioCallbackPlaySource::selectionChanged()
Chris@3: {
Chris@3:     if (m_viewManager->getPlaySelectionMode()) {
Chris@3: 	m_mutex.lock();
Chris@3: 	for (size_t c = 0; c < m_bufferCount; ++c) {
Chris@3: 	    getRingBuffer(c).reset();
Chris@3: 	}
Chris@3: 	m_mutex.unlock();
Chris@3:     }
Chris@3: }
Chris@3: 
Chris@3: void
Chris@3: AudioCallbackPlaySource::playLoopModeChanged()
Chris@3: {
Chris@3:     m_mutex.lock();
Chris@3:     for (size_t c = 0; c < m_bufferCount; ++c) {
Chris@3: 	getRingBuffer(c).reset();
Chris@3:     }
Chris@3:     m_mutex.unlock();
Chris@3: }
Chris@3: 
Chris@3: void
Chris@3: AudioCallbackPlaySource::playSelectionModeChanged()
Chris@3: {
Chris@3:     if (!m_viewManager->getSelections().empty()) {
Chris@3: 	m_mutex.lock();
Chris@3: 	for (size_t c = 0; c < m_bufferCount; ++c) {
Chris@3: 	    getRingBuffer(c).reset();
Chris@3: 	}
Chris@3: 	m_mutex.unlock();
Chris@3:     }
Chris@3: }
Chris@3: 
Chris@3: void
Chris@0: AudioCallbackPlaySource::setTargetBlockSize(size_t size)
Chris@0: {
Chris@0:     std::cerr << "AudioCallbackPlaySource::setTargetBlockSize() -> " << size << std::endl;
Chris@0:     m_blockSize = size;
Chris@0:     for (size_t i = 0; i < m_bufferCount; ++i) {
Chris@0: 	getRingBuffer(i).resize(m_ringBufferSize);
Chris@0:     }
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getTargetBlockSize() const
Chris@0: {
Chris@0:     std::cerr << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << std::endl;
Chris@0:     return m_blockSize;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::setTargetPlayLatency(size_t latency)
Chris@0: {
Chris@0:     m_playLatency = latency;
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getTargetPlayLatency() const
Chris@0: {
Chris@0:     return m_playLatency;
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getCurrentPlayingFrame()
Chris@0: {
Chris@0:     bool resample = false;
Chris@0:     double ratio = 1.0;
Chris@0: 
Chris@0:     if (getSourceSampleRate() != getTargetSampleRate()) {
Chris@0: 	resample = true;
Chris@0: 	ratio = double(getSourceSampleRate()) / double(getTargetSampleRate());
Chris@0:     }
Chris@0: 
Chris@0:     size_t readSpace = 0;
Chris@0:     for (size_t c = 0; c < getSourceChannelCount(); ++c) {
Chris@0: 	size_t spaceHere = getRingBuffer(c).getReadSpace();
Chris@0: 	if (c == 0 || spaceHere < readSpace) readSpace = spaceHere;
Chris@0:     }
Chris@0: 
Chris@0:     if (resample) {
Chris@0: 	readSpace = size_t(readSpace * ratio + 0.1);
Chris@0:     }
Chris@0: 
Chris@0:     size_t latency = m_playLatency;
Chris@0:     if (resample) latency = size_t(m_playLatency * ratio + 0.1);
Chris@0:     
Chris@0:     TimeStretcherData *timeStretcher = m_timeStretcher;
Chris@0:     if (timeStretcher) {
Chris@0: 	latency += timeStretcher->getStretcher(0)->getProcessingLatency();
Chris@0:     }
Chris@0: 
Chris@3:     latency += readSpace;
Chris@3:     size_t bufferedFrame = m_bufferedToFrame;
Chris@3: 
Chris@3:     size_t framePlaying = bufferedFrame;
Chris@3:     if (framePlaying > latency) framePlaying -= latency;
Chris@3:     else framePlaying = 0;
Chris@3: 
Chris@3:     if (!m_viewManager->getPlaySelectionMode()) {
Chris@3: 	return framePlaying;
Chris@0:     }
Chris@0: 
Chris@3:     ViewManager::SelectionList selections = m_viewManager->getSelections();
Chris@3:     if (selections.empty()) {
Chris@3: 	return framePlaying;
Chris@3:     }
Chris@3: 
Chris@3:     ViewManager::SelectionList::const_iterator i;
Chris@3: 
Chris@3:     for (i = selections.begin(); i != selections.end(); ++i) {
Chris@3: 	if (i->contains(bufferedFrame)) break;
Chris@3:     }
Chris@3: 
Chris@3:     size_t f = bufferedFrame;
Chris@3: 
Chris@3:     std::cerr << "getCurrentPlayingFrame: f=" << f << ", latency=" << latency << std::endl;
Chris@3: 
Chris@3:     if (i == selections.end()) {
Chris@3: 	--i;
Chris@3: 	if (i->getEndFrame() + latency < f) {
Chris@3: 	    return framePlaying;
Chris@3: 	} else {
Chris@3: 	    std::cerr << "latency <- " << latency << "-(" << f << "-" << i->getEndFrame() << ")" << std::endl;
Chris@3: 	    latency -= (f - i->getEndFrame());
Chris@3: 	    f = i->getEndFrame();
Chris@3: 	}
Chris@3:     }
Chris@3: 
Chris@3:     std::cerr << "i=(" << i->getStartFrame() << "," << i->getEndFrame() << ") f=" << f << ", latency=" << latency << std::endl;
Chris@3: 
Chris@3:     while (latency > 0) {
Chris@3: 	size_t offset = f - i->getStartFrame();
Chris@3: 	if (offset >= latency) {
Chris@3: 	    if (f > latency) {
Chris@3: 		framePlaying = f - latency;
Chris@3: 	    } else {
Chris@3: 		framePlaying = 0;
Chris@3: 	    }
Chris@3: 	    break;
Chris@3: 	} else {
Chris@3: 	    if (i == selections.begin()) {
Chris@3: 		if (m_viewManager->getPlayLoopMode()) {
Chris@3: 		    i = selections.end();
Chris@3: 		}
Chris@3: 	    }
Chris@3: 	    latency -= offset;
Chris@3: 	    --i;
Chris@3: 	    f = i->getEndFrame();
Chris@3: 	}
Chris@3:     }
Chris@0: 
Chris@0:     return framePlaying;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::setOutputLevels(float left, float right)
Chris@0: {
Chris@0:     m_outputLeft = left;
Chris@0:     m_outputRight = right;
Chris@0: }
Chris@0: 
Chris@0: bool
Chris@0: AudioCallbackPlaySource::getOutputLevels(float &left, float &right)
Chris@0: {
Chris@0:     left = m_outputLeft;
Chris@0:     right = m_outputRight;
Chris@0:     return true;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::setTargetSampleRate(size_t sr)
Chris@0: {
Chris@0:     m_targetSampleRate = sr;
Chris@1: 
Chris@1:     if (getSourceSampleRate() != getTargetSampleRate()) {
Chris@1: 
Chris@1: 	int err = 0;
Chris@1: 	m_converter = src_new(SRC_SINC_BEST_QUALITY, m_bufferCount, &err);
Chris@1: 	if (!m_converter) {
Chris@1: 	    std::cerr
Chris@1: 		<< "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: "
Chris@1: 		<< src_strerror(err) << std::endl;
Chris@1: 	}
Chris@1: 
Chris@1: 	emit sampleRateMismatch(getSourceSampleRate(), getTargetSampleRate());
Chris@1:     }
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getTargetSampleRate() const
Chris@0: {
Chris@0:     if (m_targetSampleRate) return m_targetSampleRate;
Chris@0:     else return getSourceSampleRate();
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getSourceChannelCount() const
Chris@0: {
Chris@0:     return m_bufferCount;
Chris@0: }
Chris@0: 
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getSourceSampleRate() const
Chris@0: {
Chris@0:     return m_sourceSampleRate;
Chris@0: }
Chris@0: 
Chris@0: AudioCallbackPlaySource::TimeStretcherData::TimeStretcherData(size_t channels,
Chris@0: 							      size_t factor,
Chris@0: 							      size_t blockSize) :
Chris@0:     m_factor(factor),
Chris@0:     m_blockSize(blockSize)
Chris@0: {
Chris@0:     std::cerr << "TimeStretcherData::TimeStretcherData(" << channels << ", " << factor << ", " << blockSize << ")" << std::endl;
Chris@0: 
Chris@0:     for (size_t ch = 0; ch < channels; ++ch) {
Chris@0: 	m_stretcher[ch] = StretcherBuffer
Chris@0: 	    //!!! We really need to measure performance and work out
Chris@0: 	    //what sort of quality level to use -- or at least to
Chris@0: 	    //allow the user to configure it
Chris@0: 	    (new IntegerTimeStretcher(factor, blockSize, 128),
Chris@0: 	     new double[blockSize * factor]);
Chris@0:     }
Chris@0:     m_stretchInputBuffer = new double[blockSize];
Chris@0: }
Chris@0: 
Chris@0: AudioCallbackPlaySource::TimeStretcherData::~TimeStretcherData()
Chris@0: {
Chris@0:     std::cerr << "IntegerTimeStretcher::~IntegerTimeStretcher" << std::endl;
Chris@0: 
Chris@0:     while (!m_stretcher.empty()) {
Chris@0: 	delete m_stretcher.begin()->second.first;
Chris@0: 	delete[] m_stretcher.begin()->second.second;
Chris@0: 	m_stretcher.erase(m_stretcher.begin());
Chris@0:     }
Chris@0:     delete m_stretchInputBuffer;
Chris@0: }
Chris@0: 
Chris@0: IntegerTimeStretcher *
Chris@0: AudioCallbackPlaySource::TimeStretcherData::getStretcher(size_t channel)
Chris@0: {
Chris@0:     return m_stretcher[channel].first;
Chris@0: }
Chris@0: 
Chris@0: double *
Chris@0: AudioCallbackPlaySource::TimeStretcherData::getOutputBuffer(size_t channel)
Chris@0: {
Chris@0:     return m_stretcher[channel].second;
Chris@0: }
Chris@0: 
Chris@0: double *
Chris@0: AudioCallbackPlaySource::TimeStretcherData::getInputBuffer()
Chris@0: {
Chris@0:     return m_stretchInputBuffer;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::TimeStretcherData::run(size_t channel)
Chris@0: {
Chris@0:     getStretcher(channel)->process(getInputBuffer(),
Chris@0: 				   getOutputBuffer(channel),
Chris@0: 				   m_blockSize);
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::setSlowdownFactor(size_t factor)
Chris@0: {
Chris@0:     // Avoid locks -- create, assign, mark old one for scavenging
Chris@0:     // later (as a call to getSourceSamples may still be using it)
Chris@0: 
Chris@0:     TimeStretcherData *existingStretcher = m_timeStretcher;
Chris@0: 
Chris@0:     if (existingStretcher && existingStretcher->getFactor() == factor) {
Chris@0: 	return;
Chris@0:     }
Chris@0: 
Chris@0:     if (factor > 1) {
Chris@0: 	TimeStretcherData *newStretcher = new TimeStretcherData
Chris@0: 	    (getSourceChannelCount(), factor, getTargetBlockSize());
Chris@0: 	m_slowdownCounter = 0;
Chris@0: 	m_timeStretcher = newStretcher;
Chris@0:     } else {
Chris@0: 	m_timeStretcher = 0;
Chris@0:     }
Chris@0: 
Chris@0:     if (existingStretcher) {
Chris@0: 	m_timeStretcherScavenger.claim(existingStretcher);
Chris@0:     }
Chris@0: }
Chris@0: 	    
Chris@0: size_t
Chris@0: AudioCallbackPlaySource::getSourceSamples(size_t count, float **buffer)
Chris@0: {
Chris@0:     if (!m_playing) {
Chris@0: 	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
Chris@0: 	    for (size_t i = 0; i < count; ++i) {
Chris@0: 		buffer[ch][i] = 0.0;
Chris@0: 	    }
Chris@0: 	}
Chris@0: 	return 0;
Chris@0:     }
Chris@0: 
Chris@0:     TimeStretcherData *timeStretcher = m_timeStretcher;
Chris@0: 
Chris@0:     if (!timeStretcher || timeStretcher->getFactor() == 1) {
Chris@0: 
Chris@0: 	size_t got = 0;
Chris@0: 
Chris@0: 	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
Chris@0: 
Chris@0: 	    RingBuffer<float> &rb = *m_buffers[ch];
Chris@0: 
Chris@0: 	    // this is marginally more likely to leave our channels in
Chris@0: 	    // sync after a processing failure than just passing "count":
Chris@0: 	    size_t request = count;
Chris@0: 	    if (ch > 0) request = got;
Chris@0: 
Chris@0: 	    got = rb.read(buffer[ch], request);
Chris@0: 	    
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	    std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " samples on channel " << ch << ", signalling for more (possibly)" << std::endl;
Chris@0: #endif
Chris@0: 	}
Chris@0: 
Chris@0: 	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
Chris@0: 	    for (size_t i = got; i < count; ++i) {
Chris@0: 		buffer[ch][i] = 0.0;
Chris@0: 	    }
Chris@0: 	}
Chris@0: 
Chris@0:         m_condition.wakeAll();
Chris@0: 	return got;
Chris@0:     }
Chris@0: 
Chris@0:     if (m_slowdownCounter == 0) {
Chris@0: 
Chris@0: 	size_t got = 0;
Chris@0: 	double *ib = timeStretcher->getInputBuffer();
Chris@0: 
Chris@0: 	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
Chris@0: 
Chris@0: 	    RingBuffer<float> &rb = *m_buffers[ch];
Chris@0: 	    size_t request = count;
Chris@0: 	    if (ch > 0) request = got; // see above
Chris@0: 	    got = rb.read(buffer[ch], request);
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	    std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " samples on channel " << ch << ", running time stretcher" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0: 	    for (size_t i = 0; i < count; ++i) {
Chris@0: 		ib[i] = buffer[ch][i];
Chris@0: 	    }
Chris@0: 	    
Chris@0: 	    timeStretcher->run(ch);
Chris@0: 	}
Chris@0: 
Chris@0:     } else if (m_slowdownCounter >= timeStretcher->getFactor()) {
Chris@0: 	// reset this in case the factor has changed leaving the
Chris@0: 	// counter out of range
Chris@0: 	m_slowdownCounter = 0;
Chris@0:     }
Chris@0: 
Chris@0:     for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
Chris@0: 
Chris@0: 	double *ob = timeStretcher->getOutputBuffer(ch);
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	std::cerr << "AudioCallbackPlaySource::getSamples: Copying from (" << (m_slowdownCounter * count) << "," << count << ") to buffer" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0: 	for (size_t i = 0; i < count; ++i) {
Chris@0: 	    buffer[ch][i] = ob[m_slowdownCounter * count + i];
Chris@0: 	}
Chris@0:     }
Chris@0: 
Chris@0:     if (m_slowdownCounter == 0) m_condition.wakeAll();
Chris@0:     m_slowdownCounter = (m_slowdownCounter + 1) % timeStretcher->getFactor();
Chris@0:     return count;
Chris@0: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::fillBuffers()
Chris@0: {
Chris@0:     static float *tmp = 0;
Chris@0:     static size_t tmpSize = 0;
Chris@0: 
Chris@0:     size_t space = 0;
Chris@0:     for (size_t c = 0; c < m_bufferCount; ++c) {
Chris@0: 	size_t spaceHere = getRingBuffer(c).getWriteSpace();
Chris@0: 	if (c == 0 || spaceHere < space) space = spaceHere;
Chris@0:     }
Chris@0:     
Chris@0:     if (space == 0) return;
Chris@0:     
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0:     std::cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0:     size_t f = m_bufferedToFrame;
Chris@0: 	
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0:     std::cout << "buffered to " << f << " already" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0:     bool resample = (getSourceSampleRate() != getTargetSampleRate());
Chris@1: 
Chris@1: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@1:     std::cout << (resample ? "" : "not ") << "resampling (source " << getSourceSampleRate() << ", target " << getTargetSampleRate() << ")" << std::endl;
Chris@1: #endif
Chris@1: 
Chris@0:     size_t channels = getSourceChannelCount();
Chris@0:     size_t orig = space;
Chris@0:     size_t got = 0;
Chris@0: 
Chris@0:     static float **bufferPtrs = 0;
Chris@0:     static size_t bufferPtrCount = 0;
Chris@0: 
Chris@0:     if (bufferPtrCount < channels) {
Chris@0: 	if (bufferPtrs) delete[] bufferPtrs;
Chris@0: 	bufferPtrs = new float *[channels];
Chris@0: 	bufferPtrCount = channels;
Chris@0:     }
Chris@0: 
Chris@0:     size_t generatorBlockSize = m_audioGenerator->getBlockSize();
Chris@0: 
Chris@1:     if (resample && !m_converter) {
Chris@1: 	static bool warned = false;
Chris@1: 	if (!warned) {
Chris@1: 	    std::cerr << "WARNING: sample rates differ, but no converter available!" << std::endl;
Chris@1: 	    warned = true;
Chris@1: 	}
Chris@1:     }
Chris@1: 
Chris@0:     if (resample && m_converter) {
Chris@0: 
Chris@0: 	double ratio =
Chris@0: 	    double(getTargetSampleRate()) / double(getSourceSampleRate());
Chris@0: 	orig = size_t(orig / ratio + 0.1);
Chris@0: 
Chris@0: 	// orig must be a multiple of generatorBlockSize
Chris@0: 	orig = (orig / generatorBlockSize) * generatorBlockSize;
Chris@0: 	if (orig == 0) return;
Chris@0: 
Chris@0: 	size_t work = std::max(orig, space);
Chris@0: 
Chris@0: 	// We only allocate one buffer, but we use it in two halves.
Chris@0: 	// We place the non-interleaved values in the second half of
Chris@0: 	// the buffer (orig samples for channel 0, orig samples for
Chris@0: 	// channel 1 etc), and then interleave them into the first
Chris@0: 	// half of the buffer.  Then we resample back into the second
Chris@0: 	// half (interleaved) and de-interleave the results back to
Chris@0: 	// the start of the buffer for insertion into the ringbuffers.
Chris@0: 	// What a faff -- especially as we've already de-interleaved
Chris@0: 	// the audio data from the source file elsewhere before we
Chris@0: 	// even reach this point.
Chris@0: 	
Chris@0: 	if (tmpSize < channels * work * 2) {
Chris@0: 	    delete[] tmp;
Chris@0: 	    tmp = new float[channels * work * 2];
Chris@0: 	    tmpSize = channels * work * 2;
Chris@0: 	}
Chris@0: 
Chris@0: 	float *nonintlv = tmp + channels * work;
Chris@0: 	float *intlv = tmp;
Chris@0: 	float *srcout = tmp + channels * work;
Chris@0: 	
Chris@0: 	for (size_t c = 0; c < channels; ++c) {
Chris@0: 	    for (size_t i = 0; i < orig; ++i) {
Chris@0: 		nonintlv[channels * i + c] = 0.0f;
Chris@0: 	    }
Chris@0: 	}
Chris@0: 
Chris@3: 	for (size_t c = 0; c < channels; ++c) {
Chris@3: 	    bufferPtrs[c] = nonintlv + c * orig;
Chris@3: 	}
Chris@0: 
Chris@3: 	bool ended = !mixModels(f, orig, bufferPtrs);
Chris@3: 	got = orig;
Chris@0: 
Chris@0: 	// and interleave into first half
Chris@0: 	for (size_t c = 0; c < channels; ++c) {
Chris@0: 	    for (size_t i = 0; i < orig; ++i) {
Chris@0: 		float sample = 0;
Chris@0: 		if (i < got) {
Chris@0: 		    sample = nonintlv[c * orig + i];
Chris@0: 		}
Chris@0: 		intlv[channels * i + c] = sample;
Chris@0: 	    }
Chris@0: 	}
Chris@0: 		
Chris@0: 	SRC_DATA data;
Chris@0: 	data.data_in = intlv;
Chris@0: 	data.data_out = srcout;
Chris@0: 	data.input_frames = orig;
Chris@0: 	data.output_frames = work;
Chris@0: 	data.src_ratio = ratio;
Chris@0: 	data.end_of_input = 0;
Chris@0: 	
Chris@0: 	int err = src_process(m_converter, &data);
Chris@0: 	size_t toCopy = size_t(work * ratio + 0.1);
Chris@0: 	
Chris@0: 	if (err) {
Chris@0: 	    std::cerr
Chris@0: 		<< "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: "
Chris@0: 		<< src_strerror(err) << std::endl;
Chris@0: 	    //!!! Then what?
Chris@0: 	} else {
Chris@0: 	    got = data.input_frames_used;
Chris@0: 	    toCopy = data.output_frames_gen;
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	    std::cerr << "Resampled " << got << " frames to " << toCopy << " frames" << std::endl;
Chris@0: #endif
Chris@0: 	}
Chris@0: 	
Chris@0: 	for (size_t c = 0; c < channels; ++c) {
Chris@0: 	    for (size_t i = 0; i < toCopy; ++i) {
Chris@0: 		tmp[i] = srcout[channels * i + c];
Chris@0: 	    }
Chris@0: 	    getRingBuffer(c).write(tmp, toCopy);
Chris@0: 	}
Chris@3: 
Chris@3: 	m_bufferedToFrame = f;
Chris@0: 	
Chris@0:     } else {
Chris@0: 
Chris@0: 	// space must be a multiple of generatorBlockSize
Chris@0: 	space = (space / generatorBlockSize) * generatorBlockSize;
Chris@0: 	if (space == 0) return;
Chris@0: 
Chris@0: 	if (tmpSize < channels * space) {
Chris@0: 	    delete[] tmp;
Chris@0: 	    tmp = new float[channels * space];
Chris@0: 	    tmpSize = channels * space;
Chris@0: 	}
Chris@0: 
Chris@0: 	for (size_t c = 0; c < channels; ++c) {
Chris@0: 
Chris@0: 	    bufferPtrs[c] = tmp + c * space;
Chris@3: 	    
Chris@0: 	    for (size_t i = 0; i < space; ++i) {
Chris@0: 		tmp[c * space + i] = 0.0f;
Chris@0: 	    }
Chris@0: 	}
Chris@0: 
Chris@3: 	bool ended = !mixModels(f, space, bufferPtrs);
Chris@0: 
Chris@0: 	for (size_t c = 0; c < channels; ++c) {
Chris@0: 
Chris@3: 	    getRingBuffer(c).write(bufferPtrs[c], space);
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	    std::cerr << "Wrote " << got << " frames for ch " << c << ", now "
Chris@0: 		      << getRingBuffer(c).getReadSpace() << " to read" 
Chris@0: 		      << std::endl;
Chris@0: #endif
Chris@0: 	}
Chris@3: 
Chris@3: 	m_bufferedToFrame = f;
Chris@3: 	//!!! 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@0:     }
Chris@3: }    
Chris@3: 
Chris@3: bool
Chris@3: AudioCallbackPlaySource::mixModels(size_t &frame, size_t count, float **buffers)
Chris@3: {
Chris@3:     size_t processed = 0;
Chris@3:     size_t chunkStart = frame;
Chris@3:     size_t chunkSize = count;
Chris@3:     size_t nextChunkStart = chunkStart + chunkSize;
Chris@0:     
Chris@3:     bool useSelection = (m_viewManager->getPlaySelectionMode() &&
Chris@3: 			 !m_viewManager->getSelections().empty());
Chris@3: 
Chris@3:     static float **chunkBufferPtrs = 0;
Chris@3:     static size_t chunkBufferPtrCount = 0;
Chris@3:     size_t channels = getSourceChannelCount();
Chris@3: 
Chris@3: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@3:     std::cerr << "Selection playback: start " << frame << ", size " << count <<", channels " << channels << std::endl;
Chris@3: #endif
Chris@3: 
Chris@3:     if (chunkBufferPtrCount < channels) {
Chris@3: 	if (chunkBufferPtrs) delete[] chunkBufferPtrs;
Chris@3: 	chunkBufferPtrs = new float *[channels];
Chris@3: 	chunkBufferPtrCount = channels;
Chris@3:     }
Chris@3: 
Chris@3:     for (size_t c = 0; c < channels; ++c) {
Chris@3: 	chunkBufferPtrs[c] = buffers[c];
Chris@3:     }
Chris@3: 
Chris@3:     while (processed < count) {
Chris@3: 	
Chris@3: 	chunkSize = count - processed;
Chris@3: 	nextChunkStart = chunkStart + chunkSize;
Chris@3: 
Chris@3: 	if (useSelection) {
Chris@3: 	    
Chris@3: 	    Selection selection =
Chris@3: 		m_viewManager->getContainingSelection(chunkStart, true);
Chris@3: 	    
Chris@3: 	    if (selection.isEmpty()) {
Chris@3: 		if (m_viewManager->getPlayLoopMode()) {
Chris@3: 		    selection = *m_viewManager->getSelections().begin();
Chris@3: 		    chunkStart = selection.getStartFrame();
Chris@3: 		}
Chris@3: 	    }
Chris@3: 
Chris@3: 	    if (selection.isEmpty()) {
Chris@3: 
Chris@3: 		chunkSize = 0;
Chris@3: 		nextChunkStart = chunkStart;
Chris@3: 
Chris@3: 	    } else {
Chris@3: 
Chris@3: 		if (chunkStart < selection.getStartFrame()) {
Chris@3: 		    chunkStart = selection.getStartFrame();
Chris@3: 		}
Chris@3: 
Chris@3: 		nextChunkStart = std::min(chunkStart + chunkSize,
Chris@3: 					  selection.getEndFrame());
Chris@3: 
Chris@3: 		chunkSize = nextChunkStart - chunkStart;
Chris@3: 	    }
Chris@3: 	}
Chris@3: 	
Chris@3: 	if (!chunkSize) {
Chris@3: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@3: 	    std::cerr << "Ending selection playback at " << nextChunkStart << std::endl;
Chris@3: #endif
Chris@3: 	    // We need to maintain full buffers so that the other
Chris@3: 	    // thread can tell where it's got to in the playback -- so
Chris@3: 	    // return the full amount here
Chris@3: 	    frame = frame + count;
Chris@3: 	    return false;
Chris@3: 	}
Chris@3: 
Chris@3: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@3: 	std::cerr << "Selection playback: chunk at " << chunkStart << " -> " << nextChunkStart << " (size " << chunkSize << ")" << std::endl;
Chris@3: #endif
Chris@3: 
Chris@3: 	size_t got = 0;
Chris@3: 
Chris@3: 	for (std::set<Model *>::iterator mi = m_models.begin();
Chris@3: 	     mi != m_models.end(); ++mi) {
Chris@3: 	    
Chris@3: 	    got = m_audioGenerator->mixModel(*mi, chunkStart, 
Chris@3: 					     chunkSize, chunkBufferPtrs);
Chris@3: 	}
Chris@3: 
Chris@3: 	for (size_t c = 0; c < channels; ++c) {
Chris@3: 	    chunkBufferPtrs[c] += chunkSize;
Chris@3: 	}
Chris@3: 
Chris@3: 	processed += chunkSize;
Chris@3: 	chunkStart = nextChunkStart;
Chris@3:     }
Chris@3: 
Chris@3: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@3:     std::cerr << "Returning selection playback at " << nextChunkStart << std::endl;
Chris@3: #endif
Chris@3: 
Chris@3:     frame = nextChunkStart;
Chris@3:     return true;
Chris@3: }
Chris@0: 
Chris@0: void
Chris@0: AudioCallbackPlaySource::AudioCallbackPlaySourceFillThread::run()
Chris@0: {
Chris@0:     AudioCallbackPlaySource &s(m_source);
Chris@0:     
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0:     std::cerr << "AudioCallbackPlaySourceFillThread starting" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0:     s.m_mutex.lock();
Chris@0: 
Chris@0:     bool previouslyPlaying = s.m_playing;
Chris@0: 
Chris@0:     while (!s.m_exiting) {
Chris@0: 
Chris@0: 	s.m_timeStretcherScavenger.scavenge();
Chris@0: 
Chris@0: 	float ms = 100;
Chris@0: 	if (s.getSourceSampleRate() > 0) {
Chris@0: 	    ms = float(m_ringBufferSize) / float(s.getSourceSampleRate()) * 1000.0;
Chris@0: 	}
Chris@0: 
Chris@0: 	if (!s.m_playing) ms *= 10;
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	std::cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms/4 << "ms..." << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0: 	s.m_condition.wait(&s.m_mutex, size_t(ms / 4));
Chris@0: 
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	std::cout << "AudioCallbackPlaySourceFillThread: awoken" << std::endl;
Chris@0: #endif
Chris@0: 
Chris@0: 	if (!s.getSourceSampleRate()) continue;
Chris@0: 
Chris@0: 	bool playing = s.m_playing;
Chris@0: 
Chris@0: 	if (playing && !previouslyPlaying) {
Chris@0: #ifdef DEBUG_AUDIO_PLAY_SOURCE
Chris@0: 	    std::cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << std::endl;
Chris@0: #endif
Chris@0: 	    for (size_t c = 0; c < s.getSourceChannelCount(); ++c) {
Chris@0: 		s.getRingBuffer(c).reset();
Chris@0: 	    }
Chris@0: 	}
Chris@0: 	previouslyPlaying = playing;
Chris@0: 
Chris@0: 	if (!playing) continue;
Chris@0: 
Chris@0: 	s.fillBuffers();
Chris@0:     }
Chris@0: 
Chris@0:     s.m_mutex.unlock();
Chris@0: }
Chris@0: 
Chris@0: 
Chris@0: 
Chris@0: #ifdef INCLUDE_MOCFILES
Chris@0: #include "AudioCallbackPlaySource.moc.cpp"
Chris@0: #endif
Chris@0: