diff audioio/AudioGenerator.cpp @ 0:db6fcbd4405c

initial import

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/audioio/AudioGenerator.cpp	Tue Jan 10 16:33:16 2006 +0000
@@ -0,0 +1,324 @@
+/* -*- 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 "AudioGenerator.h"
+
+#include "base/ViewManager.h"
+#include "base/PlayParameters.h"
+
+#include "model/DenseTimeValueModel.h"
+#include "model/SparseOneDimensionalModel.h"
+
+#include "plugin/RealTimePluginFactory.h"
+#include "plugin/RealTimePluginInstance.h"
+#include "plugin/PluginIdentifier.h"
+#include "plugin/api/alsa/seq_event.h"
+
+#include <iostream>
+
+const size_t
+AudioGenerator::m_pluginBlockSize = 2048;
+
+// #define DEBUG_AUDIO_GENERATOR 1
+
+AudioGenerator::AudioGenerator(ViewManager *manager) :
+    m_viewManager(manager),
+    m_sourceSampleRate(0),
+    m_targetChannelCount(1)
+{
+}
+
+AudioGenerator::~AudioGenerator()
+{
+}
+
+void
+AudioGenerator::addModel(Model *model)
+{
+    if (m_sourceSampleRate == 0) {
+
+	m_sourceSampleRate = model->getSampleRate();
+
+    } else {
+
+	DenseTimeValueModel *dtvm =
+	    dynamic_cast<DenseTimeValueModel *>(model);
+
+	if (dtvm) {
+	    m_sourceSampleRate = model->getSampleRate();
+	}
+    }
+
+    SparseOneDimensionalModel *sodm =
+	dynamic_cast<SparseOneDimensionalModel *>(model);
+    if (!sodm) return; // nothing else to initialise
+
+//	QString pluginId = "dssi:/usr/lib/dssi/dssi-vst.so:FEARkILLERrev1.dll";
+//	QString pluginId = "dssi:/usr/lib/dssi/hexter.so:hexter";
+//	QString pluginId = "dssi:/usr/lib/dssi/sineshaper.so:sineshaper";
+//	QString pluginId = "dssi:/usr/local/lib/dssi/xsynth-dssi.so:Xsynth";
+//	QString pluginId = "dssi:/usr/local/lib/dssi/trivial_synth.so:TS";
+    QString pluginId = QString("dssi:%1:sample_player").
+	arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME);
+    RealTimePluginFactory *factory =
+	RealTimePluginFactory::instanceFor(pluginId);
+    
+    if (!factory) {
+	std::cerr << "Failed to get plugin factory" << std::endl;
+	return;
+    }
+	
+    RealTimePluginInstance *instance =
+	factory->instantiatePlugin
+	(pluginId, 0, 0, m_sourceSampleRate, m_pluginBlockSize, m_targetChannelCount);
+
+    if (instance) {
+	m_synthMap[sodm] = instance;
+	for (unsigned int i = 0; i < instance->getParameterCount(); ++i) {
+	    instance->setParameterValue(i, instance->getParameterDefault(i));
+	}
+	QString program = instance->getProgram(0, 0);
+	if (program != "") {
+	    std::cerr << "selecting program " << program.toLocal8Bit().data() << std::endl;
+	    instance->selectProgram(program);
+	}
+	instance->selectProgram("cowbell"); //!!!
+	instance->setIdealChannelCount(m_targetChannelCount); // reset!
+    } else {
+	std::cerr << "Failed to instantiate plugin" << std::endl;
+    }
+}
+
+void
+AudioGenerator::removeModel(Model *model)
+{
+    SparseOneDimensionalModel *sodm =
+	dynamic_cast<SparseOneDimensionalModel *>(model);
+    if (!sodm) return; // nothing to do
+
+    if (m_synthMap.find(sodm) == m_synthMap.end()) return;
+
+    RealTimePluginInstance *instance = m_synthMap[sodm];
+    m_synthMap.erase(sodm);
+    delete instance;
+}
+
+void
+AudioGenerator::clearModels()
+{
+    while (!m_synthMap.empty()) {
+	RealTimePluginInstance *instance = m_synthMap.begin()->second;
+	m_synthMap.erase(m_synthMap.begin());
+	delete instance;
+    }
+}    
+
+void
+AudioGenerator::reset()
+{
+    for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
+	if (i->second) {
+	    i->second->silence();
+	    i->second->discardEvents();
+	}
+    }
+
+    m_noteOffs.clear();
+}
+
+void
+AudioGenerator::setTargetChannelCount(size_t targetChannelCount)
+{
+    m_targetChannelCount = targetChannelCount;
+
+    for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) {
+	if (i->second) i->second->setIdealChannelCount(targetChannelCount);
+    }
+}
+
+size_t
+AudioGenerator::getBlockSize() const
+{
+    return m_pluginBlockSize;
+}
+
+size_t
+AudioGenerator::mixModel(Model *model, size_t startFrame, size_t frameCount,
+			 float **buffer)
+{
+    if (m_sourceSampleRate == 0) {
+	std::cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << std::endl;
+	return frameCount;
+    }
+
+    PlayParameters *parameters = m_viewManager->getPlayParameters(model);
+    if (!parameters) return frameCount;
+
+    bool playing = !parameters->isPlayMuted();
+    if (!playing) return frameCount;
+
+    float gain = parameters->getPlayGain();
+    float pan = parameters->getPlayPan();
+
+    DenseTimeValueModel *dtvm = dynamic_cast<DenseTimeValueModel *>(model);
+    if (dtvm) {
+	return mixDenseTimeValueModel(dtvm, startFrame, frameCount,
+				      buffer, gain, pan);
+    }
+
+    SparseOneDimensionalModel *sodm = dynamic_cast<SparseOneDimensionalModel *>
+	(model);
+    if (sodm) {
+	return mixSparseOneDimensionalModel(sodm, startFrame, frameCount,
+					    buffer, gain, pan);
+    }
+
+    return frameCount;
+}
+
+size_t
+AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm,
+				       size_t startFrame, size_t frames,
+				       float **buffer, float gain, float pan)
+{
+    static float *channelBuffer = 0;
+    static size_t channelBufSiz = 0;
+    
+    if (channelBufSiz < frames) {
+	delete[] channelBuffer;
+	channelBuffer = new float[frames];
+	channelBufSiz = frames;
+    }
+    
+    size_t got = 0;
+
+    for (size_t c = 0; c < m_targetChannelCount && c < dtvm->getChannelCount(); ++c) {
+	got = dtvm->getValues(c, startFrame, startFrame + frames, channelBuffer);
+	for (size_t i = 0; i < frames; ++i) {
+	    buffer[c][i] += gain * channelBuffer[i];
+	}
+    }
+
+    return got;
+}
+    
+size_t
+AudioGenerator::mixSparseOneDimensionalModel(SparseOneDimensionalModel *sodm,
+					     size_t startFrame, size_t frames,
+					     float **buffer, float gain, float pan)
+{
+    RealTimePluginInstance *plugin = m_synthMap[sodm];
+    if (!plugin) return 0;
+
+    size_t latency = plugin->getLatency();
+    size_t blocks = frames / m_pluginBlockSize;
+    
+    //!!! hang on -- the fact that the audio callback play source's
+    //buffer is a multiple of the plugin's buffer size doesn't mean
+    //that we always get called for a multiple of it here (because it
+    //also depends on the JACK block size).  how should we ensure that
+    //all models write the same amount in to the mix, and that we
+    //always have a multiple of the plugin buffer size?  I guess this
+    //class has to be queryable for the plugin buffer size & the
+    //callback play source has to use that as a multiple for all the
+    //calls to mixModel
+
+    size_t got = blocks * m_pluginBlockSize;
+
+#ifdef DEBUG_AUDIO_GENERATOR
+    std::cout << "mixModel [sparse]: frames " << frames
+	      << ", blocks " << blocks << std::endl;
+#endif
+
+    snd_seq_event_t onEv;
+    onEv.type = SND_SEQ_EVENT_NOTEON;
+    onEv.data.note.channel = 0;
+    onEv.data.note.note = 64;
+    onEv.data.note.velocity = 127;
+
+    snd_seq_event_t offEv;
+    offEv.type = SND_SEQ_EVENT_NOTEOFF;
+    offEv.data.note.channel = 0;
+    offEv.data.note.velocity = 0;
+    
+    NoteOffSet &noteOffs = m_noteOffs[sodm];
+
+    for (size_t i = 0; i < blocks; ++i) {
+
+	size_t reqStart = startFrame + i * m_pluginBlockSize;
+
+	SparseOneDimensionalModel::PointList points =
+	    sodm->getPoints(reqStart > 0 ? reqStart + latency : reqStart,
+			    reqStart + latency + m_pluginBlockSize);
+
+	RealTime blockTime = RealTime::frame2RealTime
+	    (startFrame + i * m_pluginBlockSize, m_sourceSampleRate);
+
+	for (SparseOneDimensionalModel::PointList::iterator pli =
+		 points.begin(); pli != points.end(); ++pli) {
+
+	    size_t pliFrame = pli->frame;
+	    if (pliFrame >= latency) pliFrame -= latency;
+
+	    while (noteOffs.begin() != noteOffs.end() &&
+		   noteOffs.begin()->frame <= pliFrame) {
+
+		RealTime eventTime = RealTime::frame2RealTime
+		    (noteOffs.begin()->frame, m_sourceSampleRate);
+
+		offEv.data.note.note = noteOffs.begin()->pitch;
+		plugin->sendEvent(eventTime, &offEv);
+		noteOffs.erase(noteOffs.begin());
+	    }
+
+	    RealTime eventTime = RealTime::frame2RealTime
+		(pliFrame, m_sourceSampleRate);
+	    
+	    plugin->sendEvent(eventTime, &onEv);
+
+#ifdef DEBUG_AUDIO_GENERATOR
+	    std::cout << "mixModel [sparse]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl;
+#endif
+	    
+	    size_t duration = 7000; // frames [for now]
+	    NoteOff noff;
+	    noff.pitch = onEv.data.note.note;
+	    noff.frame = pliFrame + duration;
+	    noteOffs.insert(noff);
+	}
+
+	while (noteOffs.begin() != noteOffs.end() &&
+	       noteOffs.begin()->frame <=
+	       startFrame + i * m_pluginBlockSize + m_pluginBlockSize) {
+
+	    RealTime eventTime = RealTime::frame2RealTime
+		(noteOffs.begin()->frame, m_sourceSampleRate);
+
+	    offEv.data.note.note = noteOffs.begin()->pitch;
+	    plugin->sendEvent(eventTime, &offEv);
+	    noteOffs.erase(noteOffs.begin());
+	}
+	
+	plugin->run(blockTime);
+	float **outs = plugin->getAudioOutputBuffers();
+
+	for (size_t c = 0; c < m_targetChannelCount && c < plugin->getAudioOutputCount(); ++c) {
+#ifdef DEBUG_AUDIO_GENERATOR
+	    std::cout << "mixModel [sparse]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl;
+#endif
+
+	    for (size_t j = 0; j < m_pluginBlockSize; ++j) {
+		buffer[c][i * m_pluginBlockSize + j] += gain * outs[c][j];
+	    }
+	}
+    }
+
+    return got;
+}
+