diff audioio/AudioCallbackPlaySource.cpp @ 0:db6fcbd4405c

initial import

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/audioio/AudioCallbackPlaySource.cpp	Tue Jan 10 16:33:16 2006 +0000
@@ -0,0 +1,759 @@
+/* -*- c-basic-offset: 4 -*-  vi:set ts=8 sts=4 sw=4: */
+
+/*
+    A waveform viewer and audio annotation editor.
+    Chris Cannam, Queen Mary University of London, 2005
+    
+    This is experimental software.  Not for distribution.
+*/
+
+#include "AudioCallbackPlaySource.h"
+
+#include "AudioGenerator.h"
+
+#include "base/Model.h"
+#include "base/ViewManager.h"
+#include "model/DenseTimeValueModel.h"
+#include "model/SparseOneDimensionalModel.h"
+#include "dsp/timestretching/IntegerTimeStretcher.h"
+
+#include <iostream>
+
+//#define DEBUG_AUDIO_PLAY_SOURCE 1
+
+//const size_t AudioCallbackPlaySource::m_ringBufferSize = 102400;
+const size_t AudioCallbackPlaySource::m_ringBufferSize = 131071;
+
+AudioCallbackPlaySource::AudioCallbackPlaySource(ViewManager *manager) :
+    m_viewManager(manager),
+    m_audioGenerator(new AudioGenerator(manager)),
+    m_bufferCount(0),
+    m_blockSize(1024),
+    m_sourceSampleRate(0),
+    m_targetSampleRate(0),
+    m_playLatency(0),
+    m_playing(false),
+    m_exiting(false),
+    m_bufferedToFrame(0),
+    m_outputLeft(0.0),
+    m_outputRight(0.0),
+    m_slowdownCounter(0),
+    m_timeStretcher(0),
+    m_fillThread(0),
+    m_converter(0)
+{
+    // preallocate some slots, to avoid reallocation in an
+    // un-thread-safe manner later
+    while (m_buffers.size() < 20) m_buffers.push_back(0);
+
+    m_viewManager->setAudioPlaySource(this);
+}
+
+AudioCallbackPlaySource::~AudioCallbackPlaySource()
+{
+    m_exiting = true;
+
+    if (m_fillThread) {
+	m_condition.wakeAll();
+	m_fillThread->wait();
+	delete m_fillThread;
+    }
+
+    clearModels();
+}
+
+void
+AudioCallbackPlaySource::addModel(Model *model)
+{
+    m_mutex.lock();
+
+    m_models.insert(model);
+
+    bool buffersChanged = false, srChanged = false;
+
+    if (m_sourceSampleRate == 0) {
+
+	m_sourceSampleRate = model->getSampleRate();
+	srChanged = true;
+
+    } else if (model->getSampleRate() != m_sourceSampleRate) {
+	std::cerr << "AudioCallbackPlaySource::addModel: ERROR: "
+		  << "New model sample rate does not match" << std::endl
+		  << "existing model(s) (new " << model->getSampleRate()
+		  << " vs " << m_sourceSampleRate
+		  << "), playback will be wrong"
+		  << std::endl;
+    }
+
+    size_t sz = m_ringBufferSize;
+    if (m_bufferCount > 0) {
+	sz = m_buffers[0]->getSize();
+    }
+
+    size_t modelChannels = 1;
+    DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
+    if (dtvm) modelChannels = dtvm->getChannelCount();
+
+    while (m_bufferCount < modelChannels) {
+
+	if (m_buffers.size() < modelChannels) {
+	    // This is a hideously chancy operation -- the RT thread
+	    // could be using this vector.  We allocated several slots
+	    // in the ctor to avoid exactly this, but if we ever end
+	    // up with more channels than that (!) then we're just
+	    // going to have to risk it
+	    m_buffers.push_back(new RingBuffer<float>(sz));
+
+	} else {
+	    // The usual case
+	    m_buffers[m_bufferCount] = new RingBuffer<float>(sz);
+	}
+
+	++m_bufferCount;
+	buffersChanged = true;
+    }
+
+    if (buffersChanged) {
+	m_audioGenerator->setTargetChannelCount(m_bufferCount);
+    }
+
+    if (buffersChanged || srChanged) {
+
+	if (m_converter) {
+	    src_delete(m_converter);
+	    m_converter = 0;
+	}
+
+	if (getSourceSampleRate() != getTargetSampleRate()) {
+
+	    int err = 0;
+	    m_converter = src_new(SRC_SINC_FASTEST, m_bufferCount, &err);
+	    if (!m_converter) {
+		std::cerr
+		    << "AudioCallbackPlaySource::setModel: ERROR in creating samplerate converter: "
+		    << src_strerror(err) << std::endl;
+	    }
+	}
+    }
+
+    m_audioGenerator->addModel(model);
+
+    m_mutex.unlock();
+
+    if (!m_fillThread) {
+	m_fillThread = new AudioCallbackPlaySourceFillThread(*this);
+	m_fillThread->start();
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    std::cerr << "AudioCallbackPlaySource::addModel: emitting modelReplaced" << std::endl;
+#endif
+    emit modelReplaced();
+
+    if (srChanged && (getSourceSampleRate() != getTargetSampleRate())) {
+	emit sampleRateMismatch(getSourceSampleRate(), getTargetSampleRate());
+    }
+}
+
+void
+AudioCallbackPlaySource::removeModel(Model *model)
+{
+    m_mutex.lock();
+
+    m_models.erase(model);
+
+    if (m_models.empty()) {
+	if (m_converter) {
+	    src_delete(m_converter);
+	    m_converter = 0;
+	}
+	m_sourceSampleRate = 0;
+    }
+
+    m_audioGenerator->removeModel(model);
+
+    m_mutex.unlock();
+}
+
+void
+AudioCallbackPlaySource::clearModels()
+{
+    m_mutex.lock();
+
+    m_models.clear();
+
+    if (m_converter) {
+	src_delete(m_converter);
+	m_converter = 0;
+    }
+
+    m_audioGenerator->clearModels();
+
+    m_sourceSampleRate = 0;
+
+    m_mutex.unlock();
+}    
+
+void
+AudioCallbackPlaySource::play(size_t startFrame)
+{
+    // The fill thread will automatically empty its buffers before
+    // starting again if we have not so far been playing, but not if
+    // we're just re-seeking.
+
+    if (m_playing) {
+	m_mutex.lock();
+	m_bufferedToFrame = startFrame;
+	for (size_t c = 0; c < m_bufferCount; ++c) {
+	    getRingBuffer(c).reset();
+	    if (m_converter) src_reset(m_converter);
+	}
+	m_mutex.unlock();
+    } else {
+	m_bufferedToFrame = startFrame;
+    }
+
+    m_audioGenerator->reset();
+
+    m_playing = true;
+    m_condition.wakeAll();
+}
+
+void
+AudioCallbackPlaySource::stop()
+{
+    m_playing = false;
+    m_condition.wakeAll();
+}
+
+void
+AudioCallbackPlaySource::setTargetBlockSize(size_t size)
+{
+    std::cerr << "AudioCallbackPlaySource::setTargetBlockSize() -> " << size << std::endl;
+    m_blockSize = size;
+    for (size_t i = 0; i < m_bufferCount; ++i) {
+	getRingBuffer(i).resize(m_ringBufferSize);
+    }
+}
+
+size_t
+AudioCallbackPlaySource::getTargetBlockSize() const
+{
+    std::cerr << "AudioCallbackPlaySource::getTargetBlockSize() -> " << m_blockSize << std::endl;
+    return m_blockSize;
+}
+
+void
+AudioCallbackPlaySource::setTargetPlayLatency(size_t latency)
+{
+    m_playLatency = latency;
+}
+
+size_t
+AudioCallbackPlaySource::getTargetPlayLatency() const
+{
+    return m_playLatency;
+}
+
+size_t
+AudioCallbackPlaySource::getCurrentPlayingFrame()
+{
+    bool resample = false;
+    double ratio = 1.0;
+
+    if (getSourceSampleRate() != getTargetSampleRate()) {
+	resample = true;
+	ratio = double(getSourceSampleRate()) / double(getTargetSampleRate());
+    }
+
+    size_t readSpace = 0;
+    for (size_t c = 0; c < getSourceChannelCount(); ++c) {
+	size_t spaceHere = getRingBuffer(c).getReadSpace();
+	if (c == 0 || spaceHere < readSpace) readSpace = spaceHere;
+    }
+
+    if (resample) {
+	readSpace = size_t(readSpace * ratio + 0.1);
+    }
+
+    size_t lastRequestedFrame = 0;
+    if (m_bufferedToFrame > readSpace) {
+	lastRequestedFrame = m_bufferedToFrame - readSpace;
+    }
+
+    size_t framePlaying = lastRequestedFrame;
+
+    size_t latency = m_playLatency;
+    if (resample) latency = size_t(m_playLatency * ratio + 0.1);
+    
+    TimeStretcherData *timeStretcher = m_timeStretcher;
+    if (timeStretcher) {
+	latency += timeStretcher->getStretcher(0)->getProcessingLatency();
+    }
+
+    if (framePlaying > latency) {
+	framePlaying = framePlaying - latency;
+    } else {
+	framePlaying = 0;
+    }
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    std::cout << "getCurrentPlayingFrame: readSpace " << readSpace << ", lastRequestedFrame " << lastRequestedFrame << ", framePlaying " << framePlaying << ", latency " << latency << std::endl;
+#endif
+
+    return framePlaying;
+}
+
+void
+AudioCallbackPlaySource::setOutputLevels(float left, float right)
+{
+    m_outputLeft = left;
+    m_outputRight = right;
+}
+
+bool
+AudioCallbackPlaySource::getOutputLevels(float &left, float &right)
+{
+    left = m_outputLeft;
+    right = m_outputRight;
+    return true;
+}
+
+void
+AudioCallbackPlaySource::setTargetSampleRate(size_t sr)
+{
+    m_targetSampleRate = sr;
+}
+
+size_t
+AudioCallbackPlaySource::getTargetSampleRate() const
+{
+    if (m_targetSampleRate) return m_targetSampleRate;
+    else return getSourceSampleRate();
+}
+
+size_t
+AudioCallbackPlaySource::getSourceChannelCount() const
+{
+    return m_bufferCount;
+}
+
+size_t
+AudioCallbackPlaySource::getSourceSampleRate() const
+{
+    return m_sourceSampleRate;
+}
+
+AudioCallbackPlaySource::TimeStretcherData::TimeStretcherData(size_t channels,
+							      size_t factor,
+							      size_t blockSize) :
+    m_factor(factor),
+    m_blockSize(blockSize)
+{
+    std::cerr << "TimeStretcherData::TimeStretcherData(" << channels << ", " << factor << ", " << blockSize << ")" << std::endl;
+
+    for (size_t ch = 0; ch < channels; ++ch) {
+	m_stretcher[ch] = StretcherBuffer
+	    //!!! We really need to measure performance and work out
+	    //what sort of quality level to use -- or at least to
+	    //allow the user to configure it
+	    (new IntegerTimeStretcher(factor, blockSize, 128),
+	     new double[blockSize * factor]);
+    }
+    m_stretchInputBuffer = new double[blockSize];
+}
+
+AudioCallbackPlaySource::TimeStretcherData::~TimeStretcherData()
+{
+    std::cerr << "IntegerTimeStretcher::~IntegerTimeStretcher" << std::endl;
+
+    while (!m_stretcher.empty()) {
+	delete m_stretcher.begin()->second.first;
+	delete[] m_stretcher.begin()->second.second;
+	m_stretcher.erase(m_stretcher.begin());
+    }
+    delete m_stretchInputBuffer;
+}
+
+IntegerTimeStretcher *
+AudioCallbackPlaySource::TimeStretcherData::getStretcher(size_t channel)
+{
+    return m_stretcher[channel].first;
+}
+
+double *
+AudioCallbackPlaySource::TimeStretcherData::getOutputBuffer(size_t channel)
+{
+    return m_stretcher[channel].second;
+}
+
+double *
+AudioCallbackPlaySource::TimeStretcherData::getInputBuffer()
+{
+    return m_stretchInputBuffer;
+}
+
+void
+AudioCallbackPlaySource::TimeStretcherData::run(size_t channel)
+{
+    getStretcher(channel)->process(getInputBuffer(),
+				   getOutputBuffer(channel),
+				   m_blockSize);
+}
+
+void
+AudioCallbackPlaySource::setSlowdownFactor(size_t factor)
+{
+    // Avoid locks -- create, assign, mark old one for scavenging
+    // later (as a call to getSourceSamples may still be using it)
+
+    TimeStretcherData *existingStretcher = m_timeStretcher;
+
+    if (existingStretcher && existingStretcher->getFactor() == factor) {
+	return;
+    }
+
+    if (factor > 1) {
+	TimeStretcherData *newStretcher = new TimeStretcherData
+	    (getSourceChannelCount(), factor, getTargetBlockSize());
+	m_slowdownCounter = 0;
+	m_timeStretcher = newStretcher;
+    } else {
+	m_timeStretcher = 0;
+    }
+
+    if (existingStretcher) {
+	m_timeStretcherScavenger.claim(existingStretcher);
+    }
+}
+	    
+size_t
+AudioCallbackPlaySource::getSourceSamples(size_t count, float **buffer)
+{
+    if (!m_playing) {
+	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
+	    for (size_t i = 0; i < count; ++i) {
+		buffer[ch][i] = 0.0;
+	    }
+	}
+	return 0;
+    }
+
+    TimeStretcherData *timeStretcher = m_timeStretcher;
+
+    if (!timeStretcher || timeStretcher->getFactor() == 1) {
+
+	size_t got = 0;
+
+	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
+
+	    RingBuffer<float> &rb = *m_buffers[ch];
+
+	    // this is marginally more likely to leave our channels in
+	    // sync after a processing failure than just passing "count":
+	    size_t request = count;
+	    if (ch > 0) request = got;
+
+	    got = rb.read(buffer[ch], request);
+	    
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " samples on channel " << ch << ", signalling for more (possibly)" << std::endl;
+#endif
+	}
+
+	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
+	    for (size_t i = got; i < count; ++i) {
+		buffer[ch][i] = 0.0;
+	    }
+	}
+
+        m_condition.wakeAll();
+	return got;
+    }
+
+    if (m_slowdownCounter == 0) {
+
+	size_t got = 0;
+	double *ib = timeStretcher->getInputBuffer();
+
+	for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
+
+	    RingBuffer<float> &rb = *m_buffers[ch];
+	    size_t request = count;
+	    if (ch > 0) request = got; // see above
+	    got = rb.read(buffer[ch], request);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    std::cout << "AudioCallbackPlaySource::getSamples: got " << got << " samples on channel " << ch << ", running time stretcher" << std::endl;
+#endif
+
+	    for (size_t i = 0; i < count; ++i) {
+		ib[i] = buffer[ch][i];
+	    }
+	    
+	    timeStretcher->run(ch);
+	}
+
+    } else if (m_slowdownCounter >= timeStretcher->getFactor()) {
+	// reset this in case the factor has changed leaving the
+	// counter out of range
+	m_slowdownCounter = 0;
+    }
+
+    for (size_t ch = 0; ch < getSourceChannelCount(); ++ch) {
+
+	double *ob = timeStretcher->getOutputBuffer(ch);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	std::cerr << "AudioCallbackPlaySource::getSamples: Copying from (" << (m_slowdownCounter * count) << "," << count << ") to buffer" << std::endl;
+#endif
+
+	for (size_t i = 0; i < count; ++i) {
+	    buffer[ch][i] = ob[m_slowdownCounter * count + i];
+	}
+    }
+
+    if (m_slowdownCounter == 0) m_condition.wakeAll();
+    m_slowdownCounter = (m_slowdownCounter + 1) % timeStretcher->getFactor();
+    return count;
+}
+
+void
+AudioCallbackPlaySource::fillBuffers()
+{
+    static float *tmp = 0;
+    static size_t tmpSize = 0;
+
+    size_t space = 0;
+    for (size_t c = 0; c < m_bufferCount; ++c) {
+	size_t spaceHere = getRingBuffer(c).getWriteSpace();
+	if (c == 0 || spaceHere < space) space = spaceHere;
+    }
+    
+    if (space == 0) return;
+    
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    std::cout << "AudioCallbackPlaySourceFillThread: filling " << space << " frames" << std::endl;
+#endif
+
+    size_t f = m_bufferedToFrame;
+	
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    std::cout << "buffered to " << f << " already" << std::endl;
+#endif
+
+    bool resample = (getSourceSampleRate() != getTargetSampleRate());
+    size_t channels = getSourceChannelCount();
+    size_t orig = space;
+    size_t got = 0;
+
+    static float **bufferPtrs = 0;
+    static size_t bufferPtrCount = 0;
+
+    if (bufferPtrCount < channels) {
+	if (bufferPtrs) delete[] bufferPtrs;
+	bufferPtrs = new float *[channels];
+	bufferPtrCount = channels;
+    }
+
+    size_t generatorBlockSize = m_audioGenerator->getBlockSize();
+
+    if (resample && m_converter) {
+
+	double ratio =
+	    double(getTargetSampleRate()) / double(getSourceSampleRate());
+	orig = size_t(orig / ratio + 0.1);
+
+	// orig must be a multiple of generatorBlockSize
+	orig = (orig / generatorBlockSize) * generatorBlockSize;
+	if (orig == 0) return;
+
+	size_t work = std::max(orig, space);
+
+	// We only allocate one buffer, but we use it in two halves.
+	// We place the non-interleaved values in the second half of
+	// the buffer (orig samples for channel 0, orig samples for
+	// channel 1 etc), and then interleave them into the first
+	// half of the buffer.  Then we resample back into the second
+	// half (interleaved) and de-interleave the results back to
+	// the start of the buffer for insertion into the ringbuffers.
+	// What a faff -- especially as we've already de-interleaved
+	// the audio data from the source file elsewhere before we
+	// even reach this point.
+	
+	if (tmpSize < channels * work * 2) {
+	    delete[] tmp;
+	    tmp = new float[channels * work * 2];
+	    tmpSize = channels * work * 2;
+	}
+
+	float *nonintlv = tmp + channels * work;
+	float *intlv = tmp;
+	float *srcout = tmp + channels * work;
+	
+	for (size_t c = 0; c < channels; ++c) {
+	    for (size_t i = 0; i < orig; ++i) {
+		nonintlv[channels * i + c] = 0.0f;
+	    }
+	}
+
+	for (std::set<Model *>::iterator mi = m_models.begin();
+	     mi != m_models.end(); ++mi) {
+
+	    for (size_t c = 0; c < channels; ++c) {
+		bufferPtrs[c] = nonintlv + c * orig;
+	    }
+	    
+	    size_t gotHere = m_audioGenerator->mixModel
+		(*mi, f, orig, bufferPtrs);
+
+	    got = std::max(got, gotHere);
+	}
+
+	// and interleave into first half
+	for (size_t c = 0; c < channels; ++c) {
+	    for (size_t i = 0; i < orig; ++i) {
+		float sample = 0;
+		if (i < got) {
+		    sample = nonintlv[c * orig + i];
+		}
+		intlv[channels * i + c] = sample;
+	    }
+	}
+		
+	SRC_DATA data;
+	data.data_in = intlv;
+	data.data_out = srcout;
+	data.input_frames = orig;
+	data.output_frames = work;
+	data.src_ratio = ratio;
+	data.end_of_input = 0;
+	
+	int err = src_process(m_converter, &data);
+	size_t toCopy = size_t(work * ratio + 0.1);
+	
+	if (err) {
+	    std::cerr
+		<< "AudioCallbackPlaySourceFillThread: ERROR in samplerate conversion: "
+		<< src_strerror(err) << std::endl;
+	    //!!! Then what?
+	} else {
+	    got = data.input_frames_used;
+	    toCopy = data.output_frames_gen;
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    std::cerr << "Resampled " << got << " frames to " << toCopy << " frames" << std::endl;
+#endif
+	}
+	
+	for (size_t c = 0; c < channels; ++c) {
+	    for (size_t i = 0; i < toCopy; ++i) {
+		tmp[i] = srcout[channels * i + c];
+	    }
+	    getRingBuffer(c).write(tmp, toCopy);
+	}
+	
+    } else {
+
+	// space must be a multiple of generatorBlockSize
+	space = (space / generatorBlockSize) * generatorBlockSize;
+	if (space == 0) return;
+
+	if (tmpSize < channels * space) {
+	    delete[] tmp;
+	    tmp = new float[channels * space];
+	    tmpSize = channels * space;
+	}
+
+	for (size_t c = 0; c < channels; ++c) {
+
+	    bufferPtrs[c] = tmp + c * space;
+
+	    for (size_t i = 0; i < space; ++i) {
+		tmp[c * space + i] = 0.0f;
+	    }
+	}
+
+	for (std::set<Model *>::iterator mi = m_models.begin();
+	     mi != m_models.end(); ++mi) {
+
+	    got = m_audioGenerator->mixModel
+		(*mi, f, space, bufferPtrs);
+	}
+
+	for (size_t c = 0; c < channels; ++c) {
+
+	    got = getRingBuffer(c).write(bufferPtrs[c], space);
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    std::cerr << "Wrote " << got << " frames for ch " << c << ", now "
+		      << getRingBuffer(c).getReadSpace() << " to read" 
+		      << std::endl;
+#endif
+	}
+    }
+    
+    m_bufferedToFrame = f + got;
+}    
+
+void
+AudioCallbackPlaySource::AudioCallbackPlaySourceFillThread::run()
+{
+    AudioCallbackPlaySource &s(m_source);
+    
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+    std::cerr << "AudioCallbackPlaySourceFillThread starting" << std::endl;
+#endif
+
+    s.m_mutex.lock();
+
+    bool previouslyPlaying = s.m_playing;
+
+    while (!s.m_exiting) {
+
+	s.m_timeStretcherScavenger.scavenge();
+
+	float ms = 100;
+	if (s.getSourceSampleRate() > 0) {
+	    ms = float(m_ringBufferSize) / float(s.getSourceSampleRate()) * 1000.0;
+	}
+
+	if (!s.m_playing) ms *= 10;
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	std::cout << "AudioCallbackPlaySourceFillThread: waiting for " << ms/4 << "ms..." << std::endl;
+#endif
+
+	s.m_condition.wait(&s.m_mutex, size_t(ms / 4));
+
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	std::cout << "AudioCallbackPlaySourceFillThread: awoken" << std::endl;
+#endif
+
+	if (!s.getSourceSampleRate()) continue;
+
+	bool playing = s.m_playing;
+
+	if (playing && !previouslyPlaying) {
+#ifdef DEBUG_AUDIO_PLAY_SOURCE
+	    std::cout << "AudioCallbackPlaySourceFillThread: playback state changed, resetting" << std::endl;
+#endif
+	    for (size_t c = 0; c < s.getSourceChannelCount(); ++c) {
+		s.getRingBuffer(c).reset();
+	    }
+	}
+	previouslyPlaying = playing;
+
+	if (!playing) continue;
+
+	s.fillBuffers();
+    }
+
+    s.m_mutex.unlock();
+}
+
+
+
+#ifdef INCLUDE_MOCFILES
+#include "AudioCallbackPlaySource.moc.cpp"
+#endif
+