Mercurial > hg > sonic-visualiser
view audioio/AudioGenerator.cpp @ 46:47bff9f88882
* Add --no-audio option to avoid opening any audio output device
author | Chris Cannam |
---|---|
date | Thu, 05 Oct 2006 11:01:36 +0000 |
parents | 40116f709d3b |
children | b4110b17bca8 |
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/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ /* Sonic Visualiser An audio file viewer and annotation editor. Centre for Digital Music, Queen Mary, University of London. This file copyright 2006 Chris Cannam. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. See the file COPYING included with this distribution for more information. */ #include "AudioGenerator.h" #include "base/TempDirectory.h" #include "base/PlayParameters.h" #include "base/PlayParameterRepository.h" #include "base/Pitch.h" #include "base/Exceptions.h" #include "data/model/NoteModel.h" #include "data/model/DenseTimeValueModel.h" #include "data/model/SparseOneDimensionalModel.h" #include "plugin/RealTimePluginFactory.h" #include "plugin/RealTimePluginInstance.h" #include "plugin/PluginIdentifier.h" #include "plugin/PluginXml.h" #include "plugin/api/alsa/seq_event.h" #include <iostream> #include <math.h> #include <QDir> #include <QFile> const size_t AudioGenerator::m_pluginBlockSize = 2048; QString AudioGenerator::m_sampleDir = ""; //#define DEBUG_AUDIO_GENERATOR 1 AudioGenerator::AudioGenerator() : m_sourceSampleRate(0), m_targetChannelCount(1) { connect(PlayParameterRepository::getInstance(), SIGNAL(playPluginIdChanged(const Model *, QString)), this, SLOT(playPluginIdChanged(const Model *, QString))); connect(PlayParameterRepository::getInstance(), SIGNAL(playPluginConfigurationChanged(const Model *, QString)), this, SLOT(playPluginConfigurationChanged(const Model *, QString))); } AudioGenerator::~AudioGenerator() { } bool AudioGenerator::canPlay(const Model *model) { if (dynamic_cast<const DenseTimeValueModel *>(model) || dynamic_cast<const SparseOneDimensionalModel *>(model) || dynamic_cast<const NoteModel *>(model)) { return true; } else { return false; } } bool 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(); return true; } } RealTimePluginInstance *plugin = loadPluginFor(model); if (plugin) { QMutexLocker locker(&m_mutex); m_synthMap[model] = plugin; return true; } return false; } void AudioGenerator::playPluginIdChanged(const Model *model, QString) { if (m_synthMap.find(model) == m_synthMap.end()) return; RealTimePluginInstance *plugin = loadPluginFor(model); if (plugin) { QMutexLocker locker(&m_mutex); delete m_synthMap[model]; m_synthMap[model] = plugin; } } void AudioGenerator::playPluginConfigurationChanged(const Model *model, QString configurationXml) { // std::cerr << "AudioGenerator::playPluginConfigurationChanged" << std::endl; if (m_synthMap.find(model) == m_synthMap.end()) { std::cerr << "AudioGenerator::playPluginConfigurationChanged: We don't know about this plugin" << std::endl; return; } RealTimePluginInstance *plugin = m_synthMap[model]; if (plugin) { PluginXml(plugin).setParametersFromXml(configurationXml); } } QString AudioGenerator::getDefaultPlayPluginId(const Model *model) { const SparseOneDimensionalModel *sodm = dynamic_cast<const SparseOneDimensionalModel *>(model); if (sodm) { return QString("dssi:%1:sample_player"). arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME); } const NoteModel *nm = dynamic_cast<const NoteModel *>(model); if (nm) { return QString("dssi:%1:sample_player"). arg(PluginIdentifier::BUILTIN_PLUGIN_SONAME); } return ""; } QString AudioGenerator::getDefaultPlayPluginConfiguration(const Model *model) { QString program = ""; const SparseOneDimensionalModel *sodm = dynamic_cast<const SparseOneDimensionalModel *>(model); if (sodm) { program = "tap"; } const NoteModel *nm = dynamic_cast<const NoteModel *>(model); if (nm) { program = "piano"; } if (program == "") return ""; return QString("<plugin configuration=\"%1\" program=\"%2\"/>") .arg(XmlExportable::encodeEntities (QString("sampledir=%1") .arg(PluginXml::encodeConfigurationChars(getSampleDir())))) .arg(XmlExportable::encodeEntities(program)); } QString AudioGenerator::getSampleDir() { if (m_sampleDir != "") return m_sampleDir; try { m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples"); } catch (DirectoryCreationFailed f) { std::cerr << "WARNING: AudioGenerator::getSampleDir: Failed to create " << "temporary sample directory" << std::endl; m_sampleDir = ""; return ""; } QDir sampleResourceDir(":/samples", "*.wav"); for (unsigned int i = 0; i < sampleResourceDir.count(); ++i) { QString fileName(sampleResourceDir[i]); QFile file(sampleResourceDir.filePath(fileName)); if (!file.copy(QDir(m_sampleDir).filePath(fileName))) { std::cerr << "WARNING: AudioGenerator::getSampleDir: " << "Unable to copy " << fileName.toStdString() << " into temporary directory \"" << m_sampleDir.toStdString() << "\"" << std::endl; } } return m_sampleDir; } void AudioGenerator::setSampleDir(RealTimePluginInstance *plugin) { plugin->configure("sampledir", getSampleDir().toStdString()); } RealTimePluginInstance * AudioGenerator::loadPluginFor(const Model *model) { QString pluginId, configurationXml; PlayParameters *parameters = PlayParameterRepository::getInstance()->getPlayParameters(model); if (parameters) { pluginId = parameters->getPlayPluginId(); configurationXml = parameters->getPlayPluginConfiguration(); } if (pluginId == "") { pluginId = getDefaultPlayPluginId(model); configurationXml = getDefaultPlayPluginConfiguration(model); } if (pluginId == "") return 0; RealTimePluginInstance *plugin = loadPlugin(pluginId, ""); if (!plugin) return 0; if (configurationXml != "") { PluginXml(plugin).setParametersFromXml(configurationXml); } if (parameters) { parameters->setPlayPluginId(pluginId); parameters->setPlayPluginConfiguration(configurationXml); } return plugin; } RealTimePluginInstance * AudioGenerator::loadPlugin(QString pluginId, QString program) { RealTimePluginFactory *factory = RealTimePluginFactory::instanceFor(pluginId); if (!factory) { std::cerr << "Failed to get plugin factory" << std::endl; return false; } RealTimePluginInstance *instance = factory->instantiatePlugin (pluginId, 0, 0, m_sourceSampleRate, m_pluginBlockSize, m_targetChannelCount); if (!instance) { std::cerr << "Failed to instantiate plugin " << pluginId.toStdString() << std::endl; return 0; } setSampleDir(instance); for (unsigned int i = 0; i < instance->getParameterCount(); ++i) { instance->setParameterValue(i, instance->getParameterDefault(i)); } std::string defaultProgram = instance->getProgram(0, 0); if (defaultProgram != "") { // std::cerr << "first selecting default program " << defaultProgram << std::endl; instance->selectProgram(defaultProgram); } if (program != "") { // std::cerr << "now selecting desired program " << program.toStdString() << std::endl; instance->selectProgram(program.toStdString()); } instance->setIdealChannelCount(m_targetChannelCount); // reset! return instance; } void AudioGenerator::removeModel(Model *model) { SparseOneDimensionalModel *sodm = dynamic_cast<SparseOneDimensionalModel *>(model); if (!sodm) return; // nothing to do QMutexLocker locker(&m_mutex); if (m_synthMap.find(sodm) == m_synthMap.end()) return; RealTimePluginInstance *instance = m_synthMap[sodm]; m_synthMap.erase(sodm); delete instance; } void AudioGenerator::clearModels() { QMutexLocker locker(&m_mutex); while (!m_synthMap.empty()) { RealTimePluginInstance *instance = m_synthMap.begin()->second; m_synthMap.erase(m_synthMap.begin()); delete instance; } } void AudioGenerator::reset() { QMutexLocker locker(&m_mutex); 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) { if (m_targetChannelCount == targetChannelCount) return; // std::cerr << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << std::endl; QMutexLocker locker(&m_mutex); 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, size_t fadeIn, size_t fadeOut) { if (m_sourceSampleRate == 0) { std::cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << std::endl; return frameCount; } QMutexLocker locker(&m_mutex); PlayParameters *parameters = PlayParameterRepository::getInstance()->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, fadeIn, fadeOut); } SparseOneDimensionalModel *sodm = dynamic_cast<SparseOneDimensionalModel *> (model); if (sodm) { return mixSparseOneDimensionalModel(sodm, startFrame, frameCount, buffer, gain, pan, fadeIn, fadeOut); } NoteModel *nm = dynamic_cast<NoteModel *>(model); if (nm) { return mixNoteModel(nm, startFrame, frameCount, buffer, gain, pan, fadeIn, fadeOut); } return frameCount; } size_t AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm, size_t startFrame, size_t frames, float **buffer, float gain, float pan, size_t fadeIn, size_t fadeOut) { static float *channelBuffer = 0; static size_t channelBufSiz = 0; size_t totalFrames = frames + fadeIn/2 + fadeOut/2; if (channelBufSiz < totalFrames) { delete[] channelBuffer; channelBuffer = new float[totalFrames]; channelBufSiz = totalFrames; } size_t got = 0; size_t prevChannel = 999; for (size_t c = 0; c < m_targetChannelCount; ++c) { size_t sourceChannel = (c % dtvm->getChannelCount()); // std::cerr << "mixing channel " << c << " from source channel " << sourceChannel << std::endl; float channelGain = gain; if (pan != 0.0) { if (c == 0) { if (pan > 0.0) channelGain *= 1.0 - pan; } else { if (pan < 0.0) channelGain *= pan + 1.0; } } if (prevChannel != sourceChannel) { if (startFrame >= fadeIn/2) { got = dtvm->getValues (sourceChannel, startFrame - fadeIn/2, startFrame + frames + fadeOut/2, channelBuffer); } else { size_t missing = fadeIn/2 - startFrame; got = dtvm->getValues (sourceChannel, 0, startFrame + frames + fadeOut/2, channelBuffer + missing); } } prevChannel = sourceChannel; for (size_t i = 0; i < fadeIn/2; ++i) { float *back = buffer[c]; back -= fadeIn/2; back[i] += (channelGain * channelBuffer[i] * i) / fadeIn; } for (size_t i = 0; i < frames + fadeOut/2; ++i) { float mult = channelGain; if (i < fadeIn/2) { mult = (mult * i) / fadeIn; } if (i > frames - fadeOut/2) { mult = (mult * ((frames + fadeOut/2) - i)) / fadeOut; } buffer[c][i] += mult * channelBuffer[i]; } } return got; } size_t AudioGenerator::mixSparseOneDimensionalModel(SparseOneDimensionalModel *sodm, size_t startFrame, size_t frames, float **buffer, float gain, float pan, size_t /* fadeIn */, size_t /* fadeOut */) { 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 ¬eOffs = m_noteOffs[sodm]; for (size_t i = 0; i < blocks; ++i) { size_t reqStart = startFrame + i * m_pluginBlockSize; SparseOneDimensionalModel::PointList points = sodm->getPoints(reqStart + latency, reqStart + latency + m_pluginBlockSize); Vamp::RealTime blockTime = Vamp::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; if (pliFrame < reqStart || pliFrame >= reqStart + m_pluginBlockSize) continue; while (noteOffs.begin() != noteOffs.end() && noteOffs.begin()->frame <= pliFrame) { Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime (noteOffs.begin()->frame, m_sourceSampleRate); offEv.data.note.note = noteOffs.begin()->pitch; #ifdef DEBUG_AUDIO_GENERATOR std::cerr << "mixModel [sparse]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl; #endif plugin->sendEvent(eventTime, &offEv); noteOffs.erase(noteOffs.begin()); } Vamp::RealTime eventTime = Vamp::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) { Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime (noteOffs.begin()->frame, m_sourceSampleRate); offEv.data.note.note = noteOffs.begin()->pitch; #ifdef DEBUG_AUDIO_GENERATOR std::cerr << "mixModel [sparse]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl; #endif plugin->sendEvent(eventTime, &offEv); noteOffs.erase(noteOffs.begin()); } plugin->run(blockTime); float **outs = plugin->getAudioOutputBuffers(); for (size_t c = 0; c < m_targetChannelCount; ++c) { #ifdef DEBUG_AUDIO_GENERATOR std::cout << "mixModel [sparse]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl; #endif size_t sourceChannel = (c % plugin->getAudioOutputCount()); float channelGain = gain; if (pan != 0.0) { if (c == 0) { if (pan > 0.0) channelGain *= 1.0 - pan; } else { if (pan < 0.0) channelGain *= pan + 1.0; } } for (size_t j = 0; j < m_pluginBlockSize; ++j) { buffer[c][i * m_pluginBlockSize + j] += channelGain * outs[sourceChannel][j]; } } } return got; } //!!! mucho duplication with above -- refactor size_t AudioGenerator::mixNoteModel(NoteModel *nm, size_t startFrame, size_t frames, float **buffer, float gain, float pan, size_t /* fadeIn */, size_t /* fadeOut */) { RealTimePluginInstance *plugin = m_synthMap[nm]; 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 [note]: 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 ¬eOffs = m_noteOffs[nm]; for (size_t i = 0; i < blocks; ++i) { size_t reqStart = startFrame + i * m_pluginBlockSize; NoteModel::PointList points = nm->getPoints(reqStart + latency, reqStart + latency + m_pluginBlockSize); Vamp::RealTime blockTime = Vamp::RealTime::frame2RealTime (startFrame + i * m_pluginBlockSize, m_sourceSampleRate); for (NoteModel::PointList::iterator pli = points.begin(); pli != points.end(); ++pli) { size_t pliFrame = pli->frame; if (pliFrame >= latency) pliFrame -= latency; if (pliFrame < reqStart || pliFrame >= reqStart + m_pluginBlockSize) continue; while (noteOffs.begin() != noteOffs.end() && noteOffs.begin()->frame <= pliFrame) { Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime (noteOffs.begin()->frame, m_sourceSampleRate); offEv.data.note.note = noteOffs.begin()->pitch; #ifdef DEBUG_AUDIO_GENERATOR std::cerr << "mixModel [note]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl; #endif plugin->sendEvent(eventTime, &offEv); noteOffs.erase(noteOffs.begin()); } Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime (pliFrame, m_sourceSampleRate); if (nm->getScaleUnits() == "Hz") { onEv.data.note.note = Pitch::getPitchForFrequency(pli->value); } else { onEv.data.note.note = lrintf(pli->value); } plugin->sendEvent(eventTime, &onEv); #ifdef DEBUG_AUDIO_GENERATOR std::cout << "mixModel [note]: point at frame " << pliFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << std::endl; #endif size_t duration = pli->duration; if (duration == 0 || duration == 1) { duration = m_sourceSampleRate / 20; } 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) { Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime (noteOffs.begin()->frame, m_sourceSampleRate); offEv.data.note.note = noteOffs.begin()->pitch; #ifdef DEBUG_AUDIO_GENERATOR std::cerr << "mixModel [note]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << std::endl; #endif plugin->sendEvent(eventTime, &offEv); noteOffs.erase(noteOffs.begin()); } plugin->run(blockTime); float **outs = plugin->getAudioOutputBuffers(); for (size_t c = 0; c < m_targetChannelCount; ++c) { #ifdef DEBUG_AUDIO_GENERATOR std::cout << "mixModel [note]: adding " << m_pluginBlockSize << " samples from plugin output " << c << std::endl; #endif size_t sourceChannel = (c % plugin->getAudioOutputCount()); float channelGain = gain; if (pan != 0.0) { if (c == 0) { if (pan > 0.0) channelGain *= 1.0 - pan; } else { if (pan < 0.0) channelGain *= pan + 1.0; } } for (size_t j = 0; j < m_pluginBlockSize; ++j) { buffer[c][i * m_pluginBlockSize + j] += channelGain * outs[sourceChannel][j]; } } } return got; }