Mercurial > hg > svapp
changeset 328:28c17ce7a6e9 tony_integration
Merge from tonioni branch
| author | Chris Cannam |
|---|---|
| date | Tue, 28 Jan 2014 15:02:15 +0000 |
| parents | c837368b1faf (current diff) d2c13ec0f148 (diff) |
| children | 1e3af7f4fa86 |
| files | audioio/AudioCoreAudioTarget.cpp audioio/AudioCoreAudioTarget.h audioio/audioio.pro framework/MainWindowBase.cpp framework/framework.pro |
| diffstat | 18 files changed, 1019 insertions(+), 559 deletions(-) [+] |
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line diff
--- a/audioio/AudioCallbackPlaySource.cpp Thu Dec 12 15:20:14 2013 +0000 +++ b/audioio/AudioCallbackPlaySource.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -167,7 +167,7 @@ } #ifdef DEBUG_AUDIO_PLAY_SOURCE - cout << "Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << endl; + cout << "AudioCallbackPlaySource: Adding model with " << modelChannels << " channels at rate " << model->getSampleRate() << endl; #endif if (m_sourceSampleRate == 0) {
--- a/audioio/AudioCoreAudioTarget.cpp Thu Dec 12 15:20:14 2013 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,22 +0,0 @@ -/* -*- 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. -*/ - -#ifdef HAVE_COREAUDIO - -#include "AudioCoreAudioTarget.h" - - - -#endif
--- a/audioio/AudioCoreAudioTarget.h Thu Dec 12 15:20:14 2013 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,64 +0,0 @@ -/* -*- 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. -*/ - -#ifndef _AUDIO_CORE_AUDIO_TARGET_H_ -#define _AUDIO_CORE_AUDIO_TARGET_H_ - -#ifdef HAVE_COREAUDIO - -#include <jack/jack.h> -#include <vector> - -#include <CoreAudio/CoreAudio.h> -#include <CoreAudio/CoreAudioTypes.h> -#include <AudioUnit/AUComponent.h> -#include <AudioUnit/AudioUnitProperties.h> -#include <AudioUnit/AudioUnitParameters.h> -#include <AudioUnit/AudioOutputUnit.h> - -#include "AudioCallbackPlayTarget.h" - -class AudioCallbackPlaySource; - -class AudioCoreAudioTarget : public AudioCallbackPlayTarget -{ - Q_OBJECT - -public: - AudioCoreAudioTarget(AudioCallbackPlaySource *source); - ~AudioCoreAudioTarget(); - - virtual bool isOK() const; - -public slots: - virtual void sourceModelReplaced(); - -protected: - OSStatus process(void *data, - AudioUnitRenderActionFlags *flags, - const AudioTimeStamp *timestamp, - unsigned int inbus, - unsigned int inframes, - AudioBufferList *ioData); - - int m_bufferSize; - int m_sampleRate; - int m_latency; -}; - -#endif /* HAVE_COREAUDIO */ - -#endif -
--- a/audioio/AudioGenerator.cpp Thu Dec 12 15:20:14 2013 +0000 +++ b/audioio/AudioGenerator.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -22,14 +22,14 @@ #include "base/Exceptions.h" #include "data/model/NoteModel.h" +#include "data/model/FlexiNoteModel.h" #include "data/model/DenseTimeValueModel.h" +#include "data/model/SparseTimeValueModel.h" #include "data/model/SparseOneDimensionalModel.h" +#include "data/model/NoteData.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 "ClipMixer.h" +#include "ContinuousSynth.h" #include <iostream> #include <cmath> @@ -38,7 +38,7 @@ #include <QFile> const size_t -AudioGenerator::m_pluginBlockSize = 2048; +AudioGenerator::m_processingBlockSize = 1024; QString AudioGenerator::m_sampleDir = ""; @@ -48,19 +48,15 @@ AudioGenerator::AudioGenerator() : m_sourceSampleRate(0), m_targetChannelCount(1), + m_waveType(0), m_soloing(false) { initialiseSampleDir(); connect(PlayParameterRepository::getInstance(), - SIGNAL(playPluginIdChanged(const Playable *, QString)), + SIGNAL(playClipIdChanged(const Playable *, QString)), this, - SLOT(playPluginIdChanged(const Playable *, QString))); - - connect(PlayParameterRepository::getInstance(), - SIGNAL(playPluginConfigurationChanged(const Playable *, QString)), - this, - SLOT(playPluginConfigurationChanged(const Playable *, QString))); + SLOT(playClipIdChanged(const Playable *, QString))); } AudioGenerator::~AudioGenerator() @@ -125,74 +121,69 @@ } } - RealTimePluginInstance *plugin = loadPluginFor(model); - if (plugin) { - QMutexLocker locker(&m_mutex); - m_synthMap[model] = plugin; - return true; + if (usesClipMixer(model)) { + ClipMixer *mixer = makeClipMixerFor(model); + if (mixer) { + QMutexLocker locker(&m_mutex); + m_clipMixerMap[model] = mixer; + return true; + } + } + + if (usesContinuousSynth(model)) { + ContinuousSynth *synth = makeSynthFor(model); + if (synth) { + QMutexLocker locker(&m_mutex); + m_continuousSynthMap[model] = synth; + return true; + } } return false; } void -AudioGenerator::playPluginIdChanged(const Playable *playable, QString) +AudioGenerator::playClipIdChanged(const Playable *playable, QString) { const Model *model = dynamic_cast<const Model *>(playable); if (!model) { - cerr << "WARNING: AudioGenerator::playPluginIdChanged: playable " + cerr << "WARNING: AudioGenerator::playClipIdChanged: playable " << playable << " is not a supported model type" << endl; return; } - if (m_synthMap.find(model) == m_synthMap.end()) return; - - RealTimePluginInstance *plugin = loadPluginFor(model); - if (plugin) { + if (m_clipMixerMap.find(model) == m_clipMixerMap.end()) return; + + ClipMixer *mixer = makeClipMixerFor(model); + if (mixer) { QMutexLocker locker(&m_mutex); - delete m_synthMap[model]; - m_synthMap[model] = plugin; + m_clipMixerMap[model] = mixer; } } -void -AudioGenerator::playPluginConfigurationChanged(const Playable *playable, - QString configurationXml) +bool +AudioGenerator::usesClipMixer(const Model *model) { -// SVDEBUG << "AudioGenerator::playPluginConfigurationChanged" << endl; - - const Model *model = dynamic_cast<const Model *>(playable); - if (!model) { - cerr << "WARNING: AudioGenerator::playPluginIdChanged: playable " - << playable << " is not a supported model type" - << endl; - return; - } - - if (m_synthMap.find(model) == m_synthMap.end()) { - SVDEBUG << "AudioGenerator::playPluginConfigurationChanged: We don't know about this plugin" << endl; - return; - } - - RealTimePluginInstance *plugin = m_synthMap[model]; - if (plugin) { - PluginXml(plugin).setParametersFromXml(configurationXml); - } + bool clip = + (qobject_cast<const SparseOneDimensionalModel *>(model) || + qobject_cast<const NoteModel *>(model) || + qobject_cast<const FlexiNoteModel *>(model)); + return clip; } -void -AudioGenerator::setSampleDir(RealTimePluginInstance *plugin) +bool +AudioGenerator::usesContinuousSynth(const Model *model) { - if (m_sampleDir != "") { - plugin->configure("sampledir", m_sampleDir.toStdString()); - } -} + bool cont = + (qobject_cast<const SparseTimeValueModel *>(model)); + return cont; +} -RealTimePluginInstance * -AudioGenerator::loadPluginFor(const Model *model) +ClipMixer * +AudioGenerator::makeClipMixerFor(const Model *model) { - QString pluginId, configurationXml; + QString clipId; const Playable *playable = model; if (!playable || !playable->canPlay()) return 0; @@ -200,67 +191,48 @@ PlayParameters *parameters = PlayParameterRepository::getInstance()->getPlayParameters(playable); if (parameters) { - pluginId = parameters->getPlayPluginId(); - configurationXml = parameters->getPlayPluginConfiguration(); + clipId = parameters->getPlayClipId(); } - if (pluginId == "") return 0; + std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): sample id = " << clipId << std::endl; - RealTimePluginInstance *plugin = loadPlugin(pluginId, ""); - if (!plugin) return 0; - - if (configurationXml != "") { - PluginXml(plugin).setParametersFromXml(configurationXml); - setSampleDir(plugin); - } - - configurationXml = PluginXml(plugin).toXmlString(); - - if (parameters) { - parameters->setPlayPluginId(pluginId); - parameters->setPlayPluginConfiguration(configurationXml); - } - - return plugin; -} - -RealTimePluginInstance * -AudioGenerator::loadPlugin(QString pluginId, QString program) -{ - RealTimePluginFactory *factory = - RealTimePluginFactory::instanceFor(pluginId); - - if (!factory) { - cerr << "Failed to get plugin factory" << endl; - return 0; - } - - RealTimePluginInstance *instance = - factory->instantiatePlugin - (pluginId, 0, 0, m_sourceSampleRate, m_pluginBlockSize, m_targetChannelCount); - - if (!instance) { - cerr << "Failed to instantiate plugin " << pluginId << endl; + if (clipId == "") { + SVDEBUG << "AudioGenerator::makeClipMixerFor(" << model << "): no sample, skipping" << endl; return 0; } - setSampleDir(instance); + ClipMixer *mixer = new ClipMixer(m_targetChannelCount, + m_sourceSampleRate, + m_processingBlockSize); - for (unsigned int i = 0; i < instance->getParameterCount(); ++i) { - instance->setParameterValue(i, instance->getParameterDefault(i)); + float clipF0 = Pitch::getFrequencyForPitch(60, 0, 440.0f); // required + + QString clipPath = QString("%1/%2.wav").arg(m_sampleDir).arg(clipId); + + if (!mixer->loadClipData(clipPath, clipF0)) { + delete mixer; + return 0; } - std::string defaultProgram = instance->getProgram(0, 0); - if (defaultProgram != "") { -// cerr << "first selecting default program " << defaultProgram << endl; - instance->selectProgram(defaultProgram); - } - if (program != "") { -// cerr << "now selecting desired program " << program << endl; - instance->selectProgram(program.toStdString()); - } - instance->setIdealChannelCount(m_targetChannelCount); // reset! - return instance; + std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): loaded clip " << clipId << std::endl; + + return mixer; +} + +ContinuousSynth * +AudioGenerator::makeSynthFor(const Model *model) +{ + const Playable *playable = model; + if (!playable || !playable->canPlay()) return 0; + + ContinuousSynth *synth = new ContinuousSynth(m_targetChannelCount, + m_sourceSampleRate, + m_processingBlockSize, + m_waveType); + + std::cerr << "AudioGenerator::makeSynthFor(" << model << "): created synth" << std::endl; + + return synth; } void @@ -272,21 +244,22 @@ QMutexLocker locker(&m_mutex); - if (m_synthMap.find(sodm) == m_synthMap.end()) return; + if (m_clipMixerMap.find(sodm) == m_clipMixerMap.end()) return; - RealTimePluginInstance *instance = m_synthMap[sodm]; - m_synthMap.erase(sodm); - delete instance; + ClipMixer *mixer = m_clipMixerMap[sodm]; + m_clipMixerMap.erase(sodm); + delete mixer; } 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; + + while (!m_clipMixerMap.empty()) { + ClipMixer *mixer = m_clipMixerMap.begin()->second; + m_clipMixerMap.erase(m_clipMixerMap.begin()); + delete mixer; } } @@ -294,10 +267,10 @@ AudioGenerator::reset() { QMutexLocker locker(&m_mutex); - for (PluginMap::iterator i = m_synthMap.begin(); i != m_synthMap.end(); ++i) { + + for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { if (i->second) { - i->second->silence(); - i->second->discardEvents(); + i->second->reset(); } } @@ -314,15 +287,15 @@ 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); + for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { + if (i->second) i->second->setChannelCount(targetChannelCount); } } size_t AudioGenerator::getBlockSize() const { - return m_pluginBlockSize; + return m_processingBlockSize; } void @@ -387,15 +360,18 @@ buffer, gain, pan, fadeIn, fadeOut); } - bool synthetic = - (qobject_cast<SparseOneDimensionalModel *>(model) || - qobject_cast<NoteModel *>(model)); + if (usesClipMixer(model)) { + return mixClipModel(model, startFrame, frameCount, + buffer, gain, pan); + } - if (synthetic) { - return mixSyntheticNoteModel(model, startFrame, frameCount, - buffer, gain, pan, fadeIn, fadeOut); + if (usesContinuousSynth(model)) { + return mixContinuousSynthModel(model, startFrame, frameCount, + buffer, gain, pan); } + std::cerr << "AudioGenerator::mixModel: WARNING: Model " << model << " of type " << model->getTypeName() << " is marked as playable, but I have no mechanism to play it" << std::endl; + return frameCount; } @@ -495,18 +471,17 @@ } size_t -AudioGenerator::mixSyntheticNoteModel(Model *model, - size_t startFrame, size_t frames, - float **buffer, float gain, float pan, - size_t /* fadeIn */, - size_t /* fadeOut */) +AudioGenerator::mixClipModel(Model *model, + size_t startFrame, size_t frames, + float **buffer, float gain, float pan) { - RealTimePluginInstance *plugin = m_synthMap[model]; - if (!plugin) return 0; + ClipMixer *clipMixer = m_clipMixerMap[model]; + if (!clipMixer) return 0; - size_t latency = plugin->getLatency(); - size_t blocks = frames / m_pluginBlockSize; + size_t blocks = frames / m_processingBlockSize; + //!!! todo: the below -- it matters + //!!! 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 @@ -517,197 +492,178 @@ //callback play source has to use that as a multiple for all the //calls to mixModel - size_t got = blocks * m_pluginBlockSize; + size_t got = blocks * m_processingBlockSize; #ifdef DEBUG_AUDIO_GENERATOR - cout << "mixModel [synthetic note]: frames " << frames + cout << "mixModel [clip]: frames " << frames << ", blocks " << blocks << endl; #endif - snd_seq_event_t onEv; - onEv.type = SND_SEQ_EVENT_NOTEON; - onEv.data.note.channel = 0; + ClipMixer::NoteStart on; + ClipMixer::NoteEnd off; - 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[model]; + float **bufferIndexes = new float *[m_targetChannelCount]; + for (size_t i = 0; i < blocks; ++i) { - size_t reqStart = startFrame + i * m_pluginBlockSize; + size_t reqStart = startFrame + i * m_processingBlockSize; - NoteList notes = getNotes(model, - reqStart + latency, - reqStart + latency + m_pluginBlockSize); + NoteList notes; + NoteExportable *exportable = dynamic_cast<NoteExportable *>(model); + if (exportable) { + notes = exportable->getNotes(reqStart, + reqStart + m_processingBlockSize); + } - Vamp::RealTime blockTime = Vamp::RealTime::frame2RealTime - (startFrame + i * m_pluginBlockSize, m_sourceSampleRate); + std::vector<ClipMixer::NoteStart> starts; + std::vector<ClipMixer::NoteEnd> ends; for (NoteList::const_iterator ni = notes.begin(); ni != notes.end(); ++ni) { size_t noteFrame = ni->start; - if (noteFrame >= latency) noteFrame -= latency; - if (noteFrame < reqStart || - noteFrame >= reqStart + m_pluginBlockSize) continue; + noteFrame >= reqStart + m_processingBlockSize) continue; while (noteOffs.begin() != noteOffs.end() && noteOffs.begin()->frame <= noteFrame) { - Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime - (noteOffs.begin()->frame, m_sourceSampleRate); + size_t eventFrame = noteOffs.begin()->frame; + if (eventFrame < reqStart) eventFrame = reqStart; - offEv.data.note.note = noteOffs.begin()->pitch; + off.frameOffset = eventFrame - reqStart; + off.frequency = noteOffs.begin()->frequency; #ifdef DEBUG_AUDIO_GENERATOR - cerr << "mixModel [synthetic]: sending note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << " pitch " << noteOffs.begin()->pitch << endl; + cerr << "mixModel [clip]: adding note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; #endif - plugin->sendEvent(eventTime, &offEv); + ends.push_back(off); noteOffs.erase(noteOffs.begin()); } - Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime - (noteFrame, m_sourceSampleRate); - - if (ni->isMidiPitchQuantized) { - onEv.data.note.note = ni->midiPitch; - } else { -#ifdef DEBUG_AUDIO_GENERATOR - cerr << "mixModel [synthetic]: non-pitch-quantized notes are not supported [yet], quantizing" << endl; -#endif - onEv.data.note.note = Pitch::getPitchForFrequency(ni->frequency); - } - - onEv.data.note.velocity = ni->velocity; - - plugin->sendEvent(eventTime, &onEv); + on.frameOffset = noteFrame - reqStart; + on.frequency = ni->getFrequency(); + on.level = float(ni->velocity) / 127.0; + on.pan = pan; #ifdef DEBUG_AUDIO_GENERATOR - cout << "mixModel [synthetic]: note at frame " << noteFrame << ", block start " << (startFrame + i * m_pluginBlockSize) << ", resulting time " << eventTime << endl; + cout << "mixModel [clip]: adding note at frame " << noteFrame << ", frame offset " << on.frameOffset << " frequency " << on.frequency << endl; #endif + starts.push_back(on); noteOffs.insert - (NoteOff(onEv.data.note.note, noteFrame + ni->duration)); + (NoteOff(on.frequency, noteFrame + ni->duration)); } while (noteOffs.begin() != noteOffs.end() && - noteOffs.begin()->frame <= - startFrame + i * m_pluginBlockSize + m_pluginBlockSize) { + noteOffs.begin()->frame <= reqStart + m_processingBlockSize) { - Vamp::RealTime eventTime = Vamp::RealTime::frame2RealTime - (noteOffs.begin()->frame, m_sourceSampleRate); + size_t eventFrame = noteOffs.begin()->frame; + if (eventFrame < reqStart) eventFrame = reqStart; - offEv.data.note.note = noteOffs.begin()->pitch; + off.frameOffset = eventFrame - reqStart; + off.frequency = noteOffs.begin()->frequency; #ifdef DEBUG_AUDIO_GENERATOR - cerr << "mixModel [synthetic]: sending leftover note-off event at time " << eventTime << " frame " << noteOffs.begin()->frame << " pitch " << noteOffs.begin()->pitch << endl; + cerr << "mixModel [clip]: adding leftover note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; #endif - plugin->sendEvent(eventTime, &offEv); - noteOffs.erase(noteOffs.begin()); + ends.push_back(off); + noteOffs.erase(noteOffs.begin()); } - - plugin->run(blockTime); - float **outs = plugin->getAudioOutputBuffers(); for (size_t c = 0; c < m_targetChannelCount; ++c) { -#ifdef DEBUG_AUDIO_GENERATOR - cout << "mixModel [synthetic]: adding " << m_pluginBlockSize << " samples from plugin output " << c << endl; -#endif + bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; + } - size_t sourceChannel = (c % plugin->getAudioOutputCount()); + clipMixer->mix(bufferIndexes, gain, starts, ends); + } - 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]; - } - } - } + delete[] bufferIndexes; return got; } -AudioGenerator::NoteList -AudioGenerator::getNotes(Model *model, - size_t startFrame, - size_t endFrame) +size_t +AudioGenerator::mixContinuousSynthModel(Model *model, + size_t startFrame, + size_t frames, + float **buffer, + float gain, + float pan) { - NoteList notes; + ContinuousSynth *synth = m_continuousSynthMap[model]; + if (!synth) return 0; - SparseOneDimensionalModel *sodm = - qobject_cast<SparseOneDimensionalModel *>(model); + // only type we support here at the moment + SparseTimeValueModel *stvm = qobject_cast<SparseTimeValueModel *>(model); + if (stvm->getScaleUnits() != "Hz") return 0; - if (sodm) { - - SparseOneDimensionalModel::PointList points = - sodm->getPoints(startFrame, endFrame); + size_t blocks = frames / m_processingBlockSize; - for (SparseOneDimensionalModel::PointList::iterator pli = - points.begin(); pli != points.end(); ++pli) { + //!!! todo: see comment in mixClipModel - notes.push_back - (NoteData(pli->frame, - m_sourceSampleRate / 6, // arbitrary short duration - 64, // default pitch - 100)); // default velocity + size_t got = blocks * m_processingBlockSize; + +#ifdef DEBUG_AUDIO_GENERATOR + cout << "mixModel [synth]: frames " << frames + << ", blocks " << blocks << endl; +#endif + + float **bufferIndexes = new float *[m_targetChannelCount]; + + for (size_t i = 0; i < blocks; ++i) { + + size_t reqStart = startFrame + i * m_processingBlockSize; + + for (size_t c = 0; c < m_targetChannelCount; ++c) { + bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; } - return notes; + SparseTimeValueModel::PointList points = + stvm->getPoints(reqStart, reqStart + m_processingBlockSize); + + // by default, repeat last frequency + float f0 = 0.f; + + // go straight to the last freq that is genuinely in this range + for (SparseTimeValueModel::PointList::const_iterator itr = points.end(); + itr != points.begin(); ) { + --itr; + if (itr->frame >= reqStart && + itr->frame < reqStart + m_processingBlockSize) { + f0 = itr->value; + break; + } + } + + // if we found no such frequency and the next point is further + // away than twice the model resolution, go silent (same + // criterion TimeValueLayer uses for ending a discrete curve + // segment) + if (f0 == 0.f) { + SparseTimeValueModel::PointList nextPoints = + stvm->getNextPoints(reqStart + m_processingBlockSize); + if (nextPoints.empty() || + nextPoints.begin()->frame > reqStart + 2 * stvm->getResolution()) { + f0 = -1.f; + } + } + +// cerr << "f0 = " << f0 << endl; + + synth->mix(bufferIndexes, + gain, + pan, + f0); } - NoteModel *nm = qobject_cast<NoteModel *>(model); + delete[] bufferIndexes; - if (nm) { - - NoteModel::PointList points = - nm->getPoints(startFrame, endFrame); - - for (NoteModel::PointList::iterator pli = - points.begin(); pli != points.end(); ++pli) { - - size_t duration = pli->duration; - if (duration == 0 || duration == 1) { - duration = m_sourceSampleRate / 20; - } - - int pitch = lrintf(pli->value); - - int velocity = 100; - if (pli->level > 0.f && pli->level <= 1.f) { - velocity = lrintf(pli->level * 127); - } - - NoteData note(pli->frame, - duration, - pitch, - velocity); - - if (nm->getScaleUnits() == "Hz") { - note.frequency = pli->value; - note.isMidiPitchQuantized = false; - } - - notes.push_back(note); - } - - return notes; - } - - return notes; + return got; }
--- a/audioio/AudioGenerator.h Thu Dec 12 15:20:14 2013 +0000 +++ b/audioio/AudioGenerator.h Tue Jan 28 15:02:15 2014 +0000 @@ -18,10 +18,12 @@ class Model; class NoteModel; +class FlexiNoteModel; class DenseTimeValueModel; class SparseOneDimensionalModel; -class RealTimePluginInstance; class Playable; +class ClipMixer; +class ContinuousSynth; #include <QObject> #include <QMutex> @@ -57,7 +59,7 @@ virtual void clearModels(); /** - * Reset playback, clearing plugins and the like. + * Reset playback, clearing buffers and the like. */ virtual void reset(); @@ -92,37 +94,21 @@ virtual void clearSoloModelSet(); protected slots: - void playPluginIdChanged(const Playable *, QString); - void playPluginConfigurationChanged(const Playable *, QString); + void playClipIdChanged(const Playable *, QString); protected: - size_t m_sourceSampleRate; - size_t m_targetChannelCount; + size_t m_sourceSampleRate; + size_t m_targetChannelCount; + size_t m_waveType; bool m_soloing; std::set<Model *> m_soloModelSet; - struct NoteData { - - NoteData(size_t _start, size_t _dur, int _mp, int _vel) : - start(_start), duration(_dur), midiPitch(_mp), frequency(0), - isMidiPitchQuantized(true), velocity(_vel) { }; - - size_t start; // audio sample frame - size_t duration; // in audio sample frames - int midiPitch; // 0-127 - int frequency; // Hz, to be used if isMidiPitchQuantized false - bool isMidiPitchQuantized; - int velocity; // MIDI-style 0-127 - }; - - typedef std::vector<NoteData> NoteList; - struct NoteOff { - NoteOff(int _p, size_t _f) : pitch(_p), frame(_f) { } + NoteOff(float _freq, size_t _frame) : frequency(_freq), frame(_frame) { } - int pitch; + float frequency; size_t frame; struct Comparator { @@ -132,32 +118,43 @@ }; }; - typedef std::map<const Model *, RealTimePluginInstance *> PluginMap; + + typedef std::map<const Model *, ClipMixer *> ClipMixerMap; typedef std::multiset<NoteOff, NoteOff::Comparator> NoteOffSet; typedef std::map<const Model *, NoteOffSet> NoteOffMap; + typedef std::map<const Model *, ContinuousSynth *> ContinuousSynthMap; + QMutex m_mutex; - PluginMap m_synthMap; + + ClipMixerMap m_clipMixerMap; NoteOffMap m_noteOffs; static QString m_sampleDir; - virtual RealTimePluginInstance *loadPluginFor(const Model *model); - virtual RealTimePluginInstance *loadPlugin(QString id, QString program); + ContinuousSynthMap m_continuousSynthMap; + + bool usesClipMixer(const Model *); + bool usesContinuousSynth(const Model *); + + ClipMixer *makeClipMixerFor(const Model *model); + ContinuousSynth *makeSynthFor(const Model *model); + static void initialiseSampleDir(); - static void setSampleDir(RealTimePluginInstance *plugin); virtual size_t mixDenseTimeValueModel (DenseTimeValueModel *model, size_t startFrame, size_t frameCount, float **buffer, float gain, float pan, size_t fadeIn, size_t fadeOut); - virtual size_t mixSyntheticNoteModel + virtual size_t mixClipModel (Model *model, size_t startFrame, size_t frameCount, - float **buffer, float gain, float pan, size_t fadeIn, size_t fadeOut); + float **buffer, float gain, float pan); + + virtual size_t mixContinuousSynthModel + (Model *model, size_t startFrame, size_t frameCount, + float **buffer, float gain, float pan); - NoteList getNotes(Model *model, size_t startFrame, size_t endFrame); - - static const size_t m_pluginBlockSize; + static const size_t m_processingBlockSize; }; #endif
--- a/audioio/AudioTargetFactory.cpp Thu Dec 12 15:20:14 2013 +0000 +++ b/audioio/AudioTargetFactory.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -16,7 +16,6 @@ #include "AudioTargetFactory.h" #include "AudioJACKTarget.h" -#include "AudioCoreAudioTarget.h" #include "AudioPortAudioTarget.h" #include "AudioPulseAudioTarget.h" @@ -54,10 +53,6 @@ names.push_back("pulse"); #endif -#ifdef HAVE_COREAUDIO - names.push_back("core"); -#endif - #ifdef HAVE_PORTAUDIO_2_0 names.push_back("port"); #endif @@ -80,10 +75,6 @@ return QCoreApplication::translate("AudioTargetFactory", "PulseAudio Server"); } - if (name == "core") { - return QCoreApplication::translate("AudioTargetFactory", - "Core Audio Device"); - } if (name == "port") { return QCoreApplication::translate("AudioTargetFactory", "Default Soundcard Device"); @@ -126,10 +117,6 @@ if (m_default == "pulse") target = new AudioPulseAudioTarget(source); #endif -#ifdef HAVE_COREAUDIO - if (m_default == "core") target = new AudioCoreAudioTarget(source); -#endif - #ifdef HAVE_PORTAUDIO_2_0 if (m_default == "port") target = new AudioPortAudioTarget(source); #endif @@ -160,15 +147,6 @@ delete target; } #endif - -#ifdef HAVE_COREAUDIO - target = new AudioCoreAudioTarget(source); - if (target->isOK()) return target; - else { - cerr << "WARNING: AudioTargetFactory::createCallbackTarget: Failed to open CoreAudio target" << endl; - delete target; - } -#endif #ifdef HAVE_PORTAUDIO_2_0 target = new AudioPortAudioTarget(source);
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audioio/ClipMixer.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -0,0 +1,238 @@ +/* -*- 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, 2006-2014 QMUL. + + 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 "ClipMixer.h" + +#include <sndfile.h> +#include <cmath> + +#include "base/Debug.h" + +ClipMixer::ClipMixer(int channels, int sampleRate, int blockSize) : + m_channels(channels), + m_sampleRate(sampleRate), + m_blockSize(blockSize), + m_clipData(0) +{ +} + +ClipMixer::~ClipMixer() +{ + if (m_clipData) free(m_clipData); +} + +void +ClipMixer::setChannelCount(int channels) +{ + m_channels = channels; +} + +bool +ClipMixer::loadClipData(QString path, float f0) +{ + if (m_clipData) { + cerr << "ClipMixer::loadClipData: Already have clip loaded" << endl; + return false; + } + + SF_INFO info; + SNDFILE *file; + int sampleCount = 0; + float *tmpFrames; + size_t i; + + info.format = 0; + file = sf_open(path.toLocal8Bit().data(), SFM_READ, &info); + if (!file) { + cerr << "ClipMixer::loadClipData: Failed to open file path \"" + << path << "\": " << sf_strerror(file) << endl; + return false; + } + + tmpFrames = (float *)malloc(info.frames * info.channels * sizeof(float)); + if (!tmpFrames) { + cerr << "ClipMixer::loadClipData: malloc(" << info.frames * info.channels * sizeof(float) << ") failed" << endl; + return false; + } + + sf_readf_float(file, tmpFrames, info.frames); + sf_close(file); + + m_clipData = (float *)malloc(info.frames * sizeof(float)); + if (!m_clipData) { + cerr << "ClipMixer::loadClipData: malloc(" << info.frames * sizeof(float) << ") failed" << endl; + free(tmpFrames); + return false; + } + + for (i = 0; i < info.frames; ++i) { + int j; + m_clipData[i] = 0.0f; + for (j = 0; j < info.channels; ++j) { + m_clipData[i] += tmpFrames[i * info.channels + j]; + } + } + + free(tmpFrames); + + m_clipLength = info.frames; + m_clipF0 = f0; + m_clipRate = info.samplerate; + + return true; +} + +void +ClipMixer::reset() +{ + m_playing.clear(); +} + +float +ClipMixer::getResampleRatioFor(float frequency) +{ + if (!m_clipData) return 1.0; + float pitchRatio = m_clipF0 / frequency; + float resampleRatio = m_sampleRate / m_clipRate; + return pitchRatio * resampleRatio; +} + +int +ClipMixer::getResampledClipDuration(float frequency) +{ + return int(ceil(m_clipLength * getResampleRatioFor(frequency))); +} + +void +ClipMixer::mix(float **toBuffers, + float gain, + std::vector<NoteStart> newNotes, + std::vector<NoteEnd> endingNotes) +{ + foreach (NoteStart note, newNotes) { + if (note.frequency > 20 && + note.frequency < 5000) { + m_playing.push_back(note); + } + } + + std::vector<NoteStart> remaining; + + float *levels = new float[m_channels]; + + foreach (NoteStart note, m_playing) { + + for (int c = 0; c < m_channels; ++c) { + levels[c] = gain; + } + if (note.pan != 0.0 && m_channels == 2) { + levels[0] *= 1.0 - note.pan; + levels[1] *= note.pan + 1.0; + } + + int start = note.frameOffset; + int durationHere = m_blockSize; + if (start > 0) durationHere = m_blockSize - start; + + bool ending = false; + + foreach (NoteEnd end, endingNotes) { + if (end.frequency == note.frequency && + end.frameOffset >= start && + end.frameOffset <= m_blockSize) { + ending = true; + durationHere = end.frameOffset; + if (start > 0) durationHere = end.frameOffset - start; + break; + } + } + + int clipDuration = getResampledClipDuration(note.frequency); + if (start + clipDuration > 0) { + if (start < 0 && start + clipDuration < durationHere) { + durationHere = start + clipDuration; + } + if (durationHere > 0) { + mixNote(toBuffers, + levels, + note.frequency, + start < 0 ? -start : 0, + start > 0 ? start : 0, + durationHere, + ending); + } + } + + if (!ending) { + NoteStart adjusted = note; + adjusted.frameOffset -= m_blockSize; + remaining.push_back(adjusted); + } + } + + delete[] levels; + + m_playing = remaining; +} + +void +ClipMixer::mixNote(float **toBuffers, + float *levels, + float frequency, + int sourceOffset, + int targetOffset, + int sampleCount, + bool isEnd) +{ + if (!m_clipData) return; + + float ratio = getResampleRatioFor(frequency); + + float releaseTime = 0.01; + int releaseSampleCount = round(releaseTime * m_sampleRate); + if (releaseSampleCount > sampleCount) { + releaseSampleCount = sampleCount; + } + float releaseFraction = 1.f/releaseSampleCount; + + for (int i = 0; i < sampleCount; ++i) { + + int s = sourceOffset + i; + + float os = s / ratio; + int osi = int(floor(os)); + + //!!! just linear interpolation for now (same as SV's sample + //!!! player). a small sinc kernel would be better and + //!!! probably "good enough" + float value = 0.f; + if (osi < m_clipLength) { + value += m_clipData[osi]; + } + if (osi + 1 < m_clipLength) { + value += (m_clipData[osi + 1] - m_clipData[osi]) * (os - osi); + } + + if (isEnd && i + releaseSampleCount > sampleCount) { + value *= releaseFraction * (sampleCount - i); // linear ramp for release + } + + for (int c = 0; c < m_channels; ++c) { + toBuffers[c][targetOffset + i] += levels[c] * value; + } + } +} + +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audioio/ClipMixer.h Tue Jan 28 15:02:15 2014 +0000 @@ -0,0 +1,89 @@ +/* -*- 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, 2006-2014 QMUL. + + 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. +*/ + +#ifndef CLIP_MIXER_H +#define CLIP_MIXER_H + +#include <QString> +#include <vector> + +/** + * Mix in synthetic notes produced by resampling a prerecorded + * clip. (i.e. this is an implementation of a digital sampler in the + * musician's sense.) This can mix any number of notes of arbitrary + * frequency, so long as they all use the same sample clip. + */ + +class ClipMixer +{ +public: + ClipMixer(int channels, int sampleRate, int blockSize); + ~ClipMixer(); + + void setChannelCount(int channels); + + /** + * Load a sample clip from a wav file. This can only happen once: + * construct a new ClipMixer if you want a different clip. + */ + bool loadClipData(QString clipFilePath, float clipF0); + + void reset(); // discarding any playing notes + + struct NoteStart { + int frameOffset; // within current processing block + float frequency; // Hz + float level; // volume in range (0,1] + float pan; // range [-1,1] + }; + + struct NoteEnd { + int frameOffset; // in current processing block + float frequency; // matching note start + }; + + void mix(float **toBuffers, + float gain, + std::vector<NoteStart> newNotes, + std::vector<NoteEnd> endingNotes); + +private: + int m_channels; + int m_sampleRate; + int m_blockSize; + + QString m_clipPath; + + float *m_clipData; + int m_clipLength; + float m_clipF0; + float m_clipRate; + + std::vector<NoteStart> m_playing; + + float getResampleRatioFor(float frequency); + int getResampledClipDuration(float frequency); + + void mixNote(float **toBuffers, + float *levels, + float frequency, + int sourceOffset, // within resampled note + int targetOffset, // within target buffer + int sampleCount, + bool isEnd); +}; + + +#endif
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audioio/ContinuousSynth.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -0,0 +1,149 @@ +/* -*- 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 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 "ContinuousSynth.h" + +#include "base/Debug.h" +#include "system/System.h" + +#include <cmath> + +ContinuousSynth::ContinuousSynth(int channels, int sampleRate, int blockSize, int waveType) : + m_channels(channels), + m_sampleRate(sampleRate), + m_blockSize(blockSize), + m_prevF0(-1.f), + m_phase(0.0), + m_wavetype(waveType) // 0: 3 sinusoids, 1: 1 sinusoid, 2: sawtooth, 3: square +{ +} + +ContinuousSynth::~ContinuousSynth() +{ +} + +void +ContinuousSynth::reset() +{ + m_phase = 0; +} + +void +ContinuousSynth::mix(float **toBuffers, float gain, float pan, float f0) +{ + if (f0 == 0.f) f0 = m_prevF0; + + bool wasOn = (m_prevF0 > 0.f); + bool nowOn = (f0 > 0.f); + + if (!nowOn && !wasOn) { + m_phase = 0; + return; + } + + int fadeLength = 100; // samples + + float *levels = new float[m_channels]; + + for (int c = 0; c < m_channels; ++c) { + levels[c] = gain * 0.5; // scale gain otherwise too loud compared to source + } + if (pan != 0.0 && m_channels == 2) { + levels[0] *= 1.0 - pan; + levels[1] *= pan + 1.0; + } + +// cerr << "ContinuousSynth::mix: f0 = " << f0 << " (from " << m_prevF0 << "), phase = " << m_phase << endl; + + for (int i = 0; i < m_blockSize; ++i) { + + double fHere = (nowOn ? f0 : m_prevF0); + + if (wasOn && nowOn && (f0 != m_prevF0) && (i < fadeLength)) { + // interpolate the frequency shift + fHere = m_prevF0 + ((f0 - m_prevF0) * i) / fadeLength; + } + + double phasor = (fHere * 2 * M_PI) / m_sampleRate; + + m_phase = m_phase + phasor; + + int harmonics = (m_sampleRate / 4) / fHere - 1; + if (harmonics < 1) harmonics = 1; + + switch (m_wavetype) { + case 1: + harmonics = 1; + break; + case 2: + break; + case 3: + break; + default: + harmonics = 3; + break; + } + + + for (int h = 0; h < harmonics; ++h) { + + double v = 0; + double hn = 0; + double hp = 0; + + switch (m_wavetype) { + case 1: // single sinusoid + v = sin(m_phase); + break; + case 2: // sawtooth + if (h != 0) { + hn = h + 1; + hp = m_phase * hn; + v = -(1.0 / M_PI) * sin(hp) / hn; + } else { + v = 0.5; + } + break; + case 3: // square + hn = h*2 + 1; + hp = m_phase * hn; + v = sin(hp) / hn; + break; + default: // 3 sinusoids + hn = h + 1; + hp = m_phase * hn; + v = sin(hp) / hn; + break; + } + + if (!wasOn && i < fadeLength) { + // fade in + v = v * (i / double(fadeLength)); + } else if (!nowOn) { + // fade out + if (i > fadeLength) v = 0; + else v = v * (1.0 - (i / double(fadeLength))); + } + + for (int c = 0; c < m_channels; ++c) { + toBuffers[c][i] += levels[c] * v; + } + } + } + + m_prevF0 = f0; + + delete[] levels; +} +
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audioio/ContinuousSynth.h Tue Jan 28 15:02:15 2014 +0000 @@ -0,0 +1,63 @@ +/* -*- 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 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. +*/ + +#ifndef CONTINUOUS_SYNTH_H +#define CONTINUOUS_SYNTH_H + +/** + * Mix into a target buffer a signal synthesised so as to sound at a + * specific frequency. The frequency may change with each processing + * block, or may be switched on or off. + */ + +class ContinuousSynth +{ +public: + ContinuousSynth(int channels, int sampleRate, int blockSize, int waveType); + ~ContinuousSynth(); + + void setChannelCount(int channels); + + void reset(); + + /** + * Mix in a signal to be heard at the given fundamental + * frequency. Any oscillator state will be maintained between + * process calls so as to provide a continuous sound. The f0 value + * may vary between calls. + * + * Supply f0 equal to 0 if you want to maintain the f0 from the + * previous block (without having to remember what it was). + * + * Supply f0 less than 0 for silence. You should continue to call + * this even when the signal is silent if you want to ensure the + * sound switches on and off cleanly. + */ + void mix(float **toBuffers, + float gain, + float pan, + float f0); + +private: + int m_channels; + int m_sampleRate; + int m_blockSize; + + double m_prevF0; + double m_phase; + + int m_wavetype; +}; + +#endif
--- a/audioio/audioio.pro Thu Dec 12 15:20:14 2013 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,33 +0,0 @@ -TEMPLATE = lib - -SV_UNIT_PACKAGES = fftw3f samplerate jack portaudio-2.0 libpulse rubberband -load(../prf/sv.prf) - -CONFIG += sv staticlib qt thread warn_on stl rtti exceptions -QT -= gui - -TARGET = svaudioio - -DEPENDPATH += .. -INCLUDEPATH += . .. -OBJECTS_DIR = tmp_obj -MOC_DIR = tmp_moc - -HEADERS += AudioCallbackPlaySource.h \ - AudioCallbackPlayTarget.h \ - AudioCoreAudioTarget.h \ - AudioGenerator.h \ - AudioJACKTarget.h \ - AudioPortAudioTarget.h \ - AudioPulseAudioTarget.h \ - AudioTargetFactory.h \ - PlaySpeedRangeMapper.h -SOURCES += AudioCallbackPlaySource.cpp \ - AudioCallbackPlayTarget.cpp \ - AudioCoreAudioTarget.cpp \ - AudioGenerator.cpp \ - AudioJACKTarget.cpp \ - AudioPortAudioTarget.cpp \ - AudioPulseAudioTarget.cpp \ - AudioTargetFactory.cpp \ - PlaySpeedRangeMapper.cpp
--- a/framework/Document.cpp Thu Dec 12 15:20:14 2013 +0000 +++ b/framework/Document.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -19,6 +19,8 @@ #include "data/model/WritableWaveFileModel.h" #include "data/model/DenseThreeDimensionalModel.h" #include "data/model/DenseTimeValueModel.h" +#include "data/model/FlexiNoteModel.h" + #include "layer/Layer.h" #include "widgets/CommandHistory.h" #include "base/Command.h" @@ -27,6 +29,7 @@ #include "base/PlayParameters.h" #include "transform/TransformFactory.h" #include "transform/ModelTransformerFactory.h" +#include "transform/FeatureExtractionModelTransformer.h" #include <QApplication> #include <QTextStream> #include <QSettings> @@ -38,6 +41,8 @@ #include "data/model/SparseTimeValueModel.h" #include "data/model/AlignmentModel.h" +using std::vector; + //#define DEBUG_DOCUMENT 1 //!!! still need to handle command history, documentRestored/documentModified @@ -207,56 +212,80 @@ Document::createDerivedLayer(const Transform &transform, const ModelTransformer::Input &input) { + Transforms transforms; + transforms.push_back(transform); + vector<Layer *> layers = createDerivedLayers(transforms, input); + if (layers.empty()) return 0; + else return layers[0]; +} + +vector<Layer *> +Document::createDerivedLayers(const Transforms &transforms, + const ModelTransformer::Input &input) +{ QString message; - Model *newModel = addDerivedModel(transform, input, message); - if (!newModel) { - emit modelGenerationFailed(transform.getIdentifier(), message); - return 0; + vector<Model *> newModels = addDerivedModels(transforms, input, message); + + if (newModels.empty()) { + //!!! This identifier may be wrong! + emit modelGenerationFailed(transforms[0].getIdentifier(), message); + return vector<Layer *>(); } else if (message != "") { - emit modelGenerationWarning(transform.getIdentifier(), message); + //!!! This identifier may be wrong! + emit modelGenerationWarning(transforms[0].getIdentifier(), message); } - LayerFactory::LayerTypeSet types = - LayerFactory::getInstance()->getValidLayerTypes(newModel); + vector<Layer *> layers; - if (types.empty()) { - cerr << "WARNING: Document::createLayerForTransformer: no valid display layer for output of transform " << transform.getIdentifier() << endl; - newModel->aboutToDelete(); - emit modelAboutToBeDeleted(newModel); - m_models.erase(newModel); - delete newModel; - return 0; + for (int i = 0; i < (int)newModels.size(); ++i) { + + Model *newModel = newModels[i]; + + LayerFactory::LayerTypeSet types = + LayerFactory::getInstance()->getValidLayerTypes(newModel); + + if (types.empty()) { + cerr << "WARNING: Document::createLayerForTransformer: no valid display layer for output of transform " << transforms[i].getIdentifier() << endl; + //!!! inadequate cleanup: + newModel->aboutToDelete(); + emit modelAboutToBeDeleted(newModel); + m_models.erase(newModel); + delete newModel; + return vector<Layer *>(); + } + + //!!! for now, just use the first suitable layer type + + Layer *newLayer = createLayer(*types.begin()); + setModel(newLayer, newModel); + + //!!! We need to clone the model when adding the layer, so that it + //can be edited without affecting other layers that are based on + //the same model. Unfortunately we can't just clone it now, + //because it probably hasn't been completed yet -- the transform + //runs in the background. Maybe the transform has to handle + //cloning and cacheing models itself. + // + // Once we do clone models here, of course, we'll have to avoid + // leaking them too. + // + // We want the user to be able to add a model to a second layer + // _while it's still being calculated in the first_ and have it + // work quickly. That means we need to put the same physical + // model pointer in both layers, so they can't actually be cloned. + + if (newLayer) { + newLayer->setObjectName(getUniqueLayerName + (TransformFactory::getInstance()-> + getTransformFriendlyName + (transforms[i].getIdentifier()))); + } + + emit layerAdded(newLayer); + layers.push_back(newLayer); } - //!!! for now, just use the first suitable layer type - - Layer *newLayer = createLayer(*types.begin()); - setModel(newLayer, newModel); - - //!!! We need to clone the model when adding the layer, so that it - //can be edited without affecting other layers that are based on - //the same model. Unfortunately we can't just clone it now, - //because it probably hasn't been completed yet -- the transform - //runs in the background. Maybe the transform has to handle - //cloning and cacheing models itself. - // - // Once we do clone models here, of course, we'll have to avoid - // leaking them too. - // - // We want the user to be able to add a model to a second layer - // _while it's still being calculated in the first_ and have it - // work quickly. That means we need to put the same physical - // model pointer in both layers, so they can't actually be cloned. - - if (newLayer) { - newLayer->setObjectName(getUniqueLayerName - (TransformFactory::getInstance()-> - getTransformFriendlyName - (transform.getIdentifier()))); - } - - emit layerAdded(newLayer); - return newLayer; + return layers; } void @@ -503,40 +532,58 @@ const ModelTransformer::Input &input, QString &message) { - Model *model = 0; - for (ModelMap::iterator i = m_models.begin(); i != m_models.end(); ++i) { - if (i->second.transform == transform && - i->second.source == input.getModel() && + if (i->second.transform == transform && + i->second.source == input.getModel() && i->second.channel == input.getChannel()) { - return i->first; - } + std::cerr << "derived model taken from map " << std::endl; + return i->first; + } } - model = ModelTransformerFactory::getInstance()->transform - (transform, input, message); + Transforms tt; + tt.push_back(transform); + vector<Model *> mm = addDerivedModels(tt, input, message); + if (mm.empty()) return 0; + else return mm[0]; +} - // The transform we actually used was presumably identical to the - // one asked for, except that the version of the plugin may - // differ. It's possible that the returned message contains a - // warning about this; that doesn't concern us here, but we do - // need to ensure that the transform we remember is correct for - // what was actually applied, with the current plugin version. +vector<Model *> +Document::addDerivedModels(const Transforms &transforms, + const ModelTransformer::Input &input, + QString &message) +{ + vector<Model *> mm = + ModelTransformerFactory::getInstance()->transformMultiple + (transforms, input, message); - Transform applied = transform; - applied.setPluginVersion - (TransformFactory::getInstance()-> - getDefaultTransformFor(transform.getIdentifier(), - lrintf(transform.getSampleRate())) - .getPluginVersion()); + for (int j = 0; j < (int)mm.size(); ++j) { - if (!model) { - cerr << "WARNING: Document::addDerivedModel: no output model for transform " << transform.getIdentifier() << endl; - } else { - addDerivedModel(applied, input, model); + Model *model = mm[j]; + + // The transform we actually used was presumably identical to + // the one asked for, except that the version of the plugin + // may differ. It's possible that the returned message + // contains a warning about this; that doesn't concern us + // here, but we do need to ensure that the transform we + // remember is correct for what was actually applied, with the + // current plugin version. + + Transform applied = transforms[j]; + applied.setPluginVersion + (TransformFactory::getInstance()-> + getDefaultTransformFor(applied.getIdentifier(), + lrintf(applied.getSampleRate())) + .getPluginVersion()); + + if (!model) { + cerr << "WARNING: Document::addDerivedModel: no output model for transform " << applied.getIdentifier() << endl; + } else { + addDerivedModel(applied, input, model); + } } - - return model; + + return mm; } void @@ -690,6 +737,7 @@ } LayerFactory::getInstance()->setModel(layer, model); + // std::cerr << "layer type: " << LayerFactory::getInstance()->getLayerTypeName(LayerFactory::getInstance()->getLayerType(layer)) << std::endl; if (previousModel) { releaseModel(previousModel);
--- a/framework/Document.h Thu Dec 12 15:20:14 2013 +0000 +++ b/framework/Document.h Tue Jan 28 15:02:15 2014 +0000 @@ -19,6 +19,7 @@ #include "layer/LayerFactory.h" #include "transform/Transform.h" #include "transform/ModelTransformer.h" +#include "transform/FeatureExtractionModelTransformer.h" #include "base/Command.h" #include <map> @@ -117,6 +118,15 @@ const ModelTransformer::Input &); /** + * Create and return suitable layers for the given transforms, + * which must be identical apart from the output (i.e. must use + * the same plugin and configuration). The layers are returned in + * the same order as the transformed are supplied. + */ + std::vector<Layer *> createDerivedLayers(const Transforms &, + const ModelTransformer::Input &); + + /** * Delete the given layer, and also its associated model if no * longer used by any other layer. In general, this should be the * only method used to delete layers -- doing so directly is a bit @@ -154,6 +164,15 @@ QString &returnedMessage); /** + * Add derived models associated with the given set of related + * transforms, running the transforms and returning the resulting + * models. + */ + std::vector<Model *> addDerivedModels(const Transforms &transforms, + const ModelTransformer::Input &input, + QString &returnedMessage); + + /** * Add a derived model associated with the given transform. This * is necessary to register any derived model that was not created * by the document using createDerivedModel or createDerivedLayer.
--- a/framework/MainWindowBase.cpp Thu Dec 12 15:20:14 2013 +0000 +++ b/framework/MainWindowBase.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -22,6 +22,7 @@ #include "data/model/WaveFileModel.h" #include "data/model/SparseOneDimensionalModel.h" #include "data/model/NoteModel.h" +#include "data/model/FlexiNoteModel.h" #include "data/model/Labeller.h" #include "data/model/TabularModel.h" #include "view/ViewManager.h" @@ -35,6 +36,7 @@ #include "layer/SliceableLayer.h" #include "layer/ImageLayer.h" #include "layer/NoteLayer.h" +#include "layer/FlexiNoteLayer.h" #include "layer/RegionLayer.h" #include "widgets/ListInputDialog.h" @@ -309,6 +311,10 @@ return ff->getOpenFileName(type, m_sessionFile); case FileFinder::LayerFileNoMidi: return ff->getOpenFileName(type, m_sessionFile); + case FileFinder::LayerFileNonSV: + return ff->getOpenFileName(type, m_sessionFile); + case FileFinder::LayerFileNoMidiNonSV: + return ff->getOpenFileName(type, m_sessionFile); case FileFinder::SessionOrAudioFile: return ff->getOpenFileName(type, m_sessionFile); case FileFinder::ImageFile: @@ -341,6 +347,10 @@ return ff->getSaveFileName(type, m_sessionFile); case FileFinder::LayerFileNoMidi: return ff->getSaveFileName(type, m_sessionFile); + case FileFinder::LayerFileNonSV: + return ff->getSaveFileName(type, m_sessionFile); + case FileFinder::LayerFileNoMidiNonSV: + return ff->getSaveFileName(type, m_sessionFile); case FileFinder::SessionOrAudioFile: return ff->getSaveFileName(type, m_sessionFile); case FileFinder::ImageFile: @@ -430,6 +440,7 @@ bool haveCurrentDurationLayer = (haveCurrentLayer && (dynamic_cast<NoteLayer *>(currentLayer) || + dynamic_cast<FlexiNoteLayer *>(currentLayer) || dynamic_cast<RegionLayer *>(currentLayer))); bool haveCurrentColour3DPlot = (haveCurrentLayer && @@ -1015,6 +1026,25 @@ CommandHistory::getInstance()->addCommand(c, false); return; } + + FlexiNoteModel *fnm = dynamic_cast<FlexiNoteModel *>(layer->getModel()); + if (fnm) { + FlexiNoteModel::Point point(alignedStart, + rm->getValueMinimum(), + alignedDuration, + 1.f, + ""); + FlexiNoteModel::EditCommand *command = + new FlexiNoteModel::EditCommand(fnm, tr("Add Point")); + command->addPoint(point); + command->setName(name); + c = command->finish(); + } + + if (c) { + CommandHistory::getInstance()->addCommand(c, false); + return; + } } void @@ -2229,7 +2259,11 @@ MainWindowBase::zoomDefault() { Pane *currentPane = m_paneStack->getCurrentPane(); - if (currentPane) currentPane->setZoomLevel(1024); + QSettings settings; + settings.beginGroup("MainWindow"); + int zoom = settings.value("zoom-default", 1024).toInt(); + settings.endGroup(); + if (currentPane) currentPane->setZoomLevel(zoom); } void @@ -3091,6 +3125,7 @@ MainWindowBase::modelAdded(Model *model) { // SVDEBUG << "MainWindowBase::modelAdded(" << model << ")" << endl; + std::cerr << "\nAdding model " << model->getTypeName() << " to playsource " << std::endl; m_playSource->addModel(model); }
--- a/framework/SVFileReader.cpp Thu Dec 12 15:20:14 2013 +0000 +++ b/framework/SVFileReader.cpp Tue Jan 28 15:02:15 2014 +0000 @@ -1271,8 +1271,8 @@ float gain = attributes.value("gain").toFloat(&ok); if (ok) parameters->setPlayGain(gain); - QString pluginId = attributes.value("pluginId"); - if (pluginId != "") parameters->setPlayPluginId(pluginId); + QString clipId = attributes.value("clipId"); + if (clipId != "") parameters->setPlayClipId(clipId); m_currentPlayParameters = parameters; @@ -1291,17 +1291,26 @@ bool SVFileReader::readPlugin(const QXmlAttributes &attributes) { - if (m_currentDerivedModelId < 0 && !m_currentPlayParameters) { + if (m_currentDerivedModelId >= 0) { + return readPluginForTransform(attributes); + } else if (m_currentPlayParameters) { + return readPluginForPlayback(attributes); + } else { cerr << "WARNING: SV-XML: Plugin found outside derivation or play parameters" << endl; return false; } +} - if (!m_currentPlayParameters && m_currentTransformIsNewStyle) { +bool +SVFileReader::readPluginForTransform(const QXmlAttributes &attributes) +{ + if (m_currentTransformIsNewStyle) { + // Not needed, we have the transform element instead return true; } QString configurationXml = "<plugin"; - + for (int i = 0; i < attributes.length(); ++i) { configurationXml += QString(" %1=\"%2\"") .arg(attributes.qName(i)) @@ -1310,12 +1319,21 @@ configurationXml += "/>"; - if (m_currentPlayParameters) { - m_currentPlayParameters->setPlayPluginConfiguration(configurationXml); - } else { - TransformFactory::getInstance()-> - setParametersFromPluginConfigurationXml(m_currentTransform, - configurationXml); + TransformFactory::getInstance()-> + setParametersFromPluginConfigurationXml(m_currentTransform, + configurationXml); + return true; +} + +bool +SVFileReader::readPluginForPlayback(const QXmlAttributes &attributes) +{ + // Obsolete but supported for compatibility + + QString ident = attributes.value("identifier"); + if (ident == "sample_player") { + QString clipId = attributes.value("program"); + if (clipId != "") m_currentPlayParameters->setPlayClipId(clipId); } return true;
--- a/framework/SVFileReader.h Thu Dec 12 15:20:14 2013 +0000 +++ b/framework/SVFileReader.h Tue Jan 28 15:02:15 2014 +0000 @@ -222,6 +222,8 @@ bool readDerivation(const QXmlAttributes &); bool readPlayParameters(const QXmlAttributes &); bool readPlugin(const QXmlAttributes &); + bool readPluginForTransform(const QXmlAttributes &); + bool readPluginForPlayback(const QXmlAttributes &); bool readTransform(const QXmlAttributes &); bool readParameter(const QXmlAttributes &); bool readSelection(const QXmlAttributes &);
--- a/framework/framework.pro Thu Dec 12 15:20:14 2013 +0000 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,25 +0,0 @@ -TEMPLATE = lib - -SV_UNIT_PACKAGES = vamp vamp-hostsdk # required because we use transform headers -load(../prf/sv.prf) - -CONFIG += sv staticlib qt thread warn_on stl rtti exceptions -QT += xml network - -TARGET = svframework - -DEPENDPATH += .. -INCLUDEPATH += . .. -OBJECTS_DIR = tmp_obj -MOC_DIR = tmp_moc - -HEADERS += Document.h \ - MainWindowBase.h \ - SVFileReader.h \ - VersionTester.h - -SOURCES += Document.cpp \ - MainWindowBase.cpp \ - SVFileReader.cpp \ - VersionTester.cpp -
--- a/svapp.pro Thu Dec 12 15:20:14 2013 +0000 +++ b/svapp.pro Tue Jan 28 15:02:15 2014 +0000 @@ -4,10 +4,26 @@ exists(config.pri) { include(config.pri) } -win* { - !exists(config.pri) { +!exists(config.pri) { + win32-g++ { + INCLUDEPATH += ../sv-dependency-builds/win32-mingw/include + LIBS += -L../sv-dependency-builds/win32-mingw/lib + } + win32-msvc* { + INCLUDEPATH += ../sv-dependency-builds/win32-msvc/include + LIBS += -L../sv-dependency-builds/win32-msvc/lib + } + macx* { + INCLUDEPATH += ../sv-dependency-builds/osx/include + LIBS += -L../sv-dependency-builds/osx/lib + } + + win* { DEFINES += HAVE_PORTAUDIO_2_0 } + macx* { + DEFINES += HAVE_COREAUDIO HAVE_PORTAUDIO_2_0 + } } CONFIG += staticlib qt thread warn_on stl rtti exceptions @@ -20,30 +36,26 @@ OBJECTS_DIR = o MOC_DIR = o -win32-g++ { - INCLUDEPATH += ../sv-dependency-builds/win32-mingw/include -} -win32-msvc* { - INCLUDEPATH += ../sv-dependency-builds/win32-msvc/include -} - HEADERS += audioio/AudioCallbackPlaySource.h \ audioio/AudioCallbackPlayTarget.h \ - audioio/AudioCoreAudioTarget.h \ audioio/AudioGenerator.h \ audioio/AudioJACKTarget.h \ audioio/AudioPortAudioTarget.h \ audioio/AudioPulseAudioTarget.h \ audioio/AudioTargetFactory.h \ + audioio/ClipMixer.h \ + audioio/ContinuousSynth.h \ audioio/PlaySpeedRangeMapper.h + SOURCES += audioio/AudioCallbackPlaySource.cpp \ audioio/AudioCallbackPlayTarget.cpp \ - audioio/AudioCoreAudioTarget.cpp \ audioio/AudioGenerator.cpp \ audioio/AudioJACKTarget.cpp \ audioio/AudioPortAudioTarget.cpp \ audioio/AudioPulseAudioTarget.cpp \ audioio/AudioTargetFactory.cpp \ + audioio/ClipMixer.cpp \ + audioio/ContinuousSynth.cpp \ audioio/PlaySpeedRangeMapper.cpp HEADERS += framework/Document.h \