Mercurial > hg > svapp
diff audio/AudioGenerator.cpp @ 468:56acd9368532 bqaudioio
Initial work toward switching to bqaudioio library (so as to get I/O, not just O)
author | Chris Cannam |
---|---|
date | Tue, 04 Aug 2015 13:27:42 +0100 |
parents | audioio/AudioGenerator.cpp@3485d324c172 |
children | 4480b031fe38 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/audio/AudioGenerator.cpp Tue Aug 04 13:27:42 2015 +0100 @@ -0,0 +1,709 @@ +/* -*- 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/FlexiNoteModel.h" +#include "data/model/DenseTimeValueModel.h" +#include "data/model/SparseTimeValueModel.h" +#include "data/model/SparseOneDimensionalModel.h" +#include "data/model/NoteData.h" + +#include "ClipMixer.h" +#include "ContinuousSynth.h" + +#include <iostream> +#include <cmath> + +#include <QDir> +#include <QFile> + +const sv_frame_t +AudioGenerator::m_processingBlockSize = 1024; + +QString +AudioGenerator::m_sampleDir = ""; + +//#define DEBUG_AUDIO_GENERATOR 1 + +AudioGenerator::AudioGenerator() : + m_sourceSampleRate(0), + m_targetChannelCount(1), + m_waveType(0), + m_soloing(false), + m_channelBuffer(0), + m_channelBufSiz(0), + m_channelBufCount(0) +{ + initialiseSampleDir(); + + connect(PlayParameterRepository::getInstance(), + SIGNAL(playClipIdChanged(const Playable *, QString)), + this, + SLOT(playClipIdChanged(const Playable *, QString))); +} + +AudioGenerator::~AudioGenerator() +{ +#ifdef DEBUG_AUDIO_GENERATOR + SVDEBUG << "AudioGenerator::~AudioGenerator" << endl; +#endif +} + +void +AudioGenerator::initialiseSampleDir() +{ + if (m_sampleDir != "") return; + + try { + m_sampleDir = TempDirectory::getInstance()->getSubDirectoryPath("samples"); + } catch (DirectoryCreationFailed f) { + cerr << "WARNING: AudioGenerator::initialiseSampleDir:" + << " Failed to create temporary sample directory" + << 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)); + QString target = QDir(m_sampleDir).filePath(fileName); + + if (!file.copy(target)) { + cerr << "WARNING: AudioGenerator::getSampleDir: " + << "Unable to copy " << fileName + << " into temporary directory \"" + << m_sampleDir << "\"" << endl; + } else { + QFile tf(target); + tf.setPermissions(tf.permissions() | + QFile::WriteOwner | + QFile::WriteUser); + } + } +} + +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; + } + } + + const Playable *playable = model; + if (!playable || !playable->canPlay()) return 0; + + PlayParameters *parameters = + PlayParameterRepository::getInstance()->getPlayParameters(playable); + + bool willPlay = !parameters->isPlayMuted(); + + if (usesClipMixer(model)) { + ClipMixer *mixer = makeClipMixerFor(model); + if (mixer) { + QMutexLocker locker(&m_mutex); + m_clipMixerMap[model] = mixer; + return willPlay; + } + } + + if (usesContinuousSynth(model)) { + ContinuousSynth *synth = makeSynthFor(model); + if (synth) { + QMutexLocker locker(&m_mutex); + m_continuousSynthMap[model] = synth; + return willPlay; + } + } + + return false; +} + +void +AudioGenerator::playClipIdChanged(const Playable *playable, QString) +{ + const Model *model = dynamic_cast<const Model *>(playable); + if (!model) { + cerr << "WARNING: AudioGenerator::playClipIdChanged: playable " + << playable << " is not a supported model type" + << endl; + return; + } + + if (m_clipMixerMap.find(model) == m_clipMixerMap.end()) return; + + ClipMixer *mixer = makeClipMixerFor(model); + if (mixer) { + QMutexLocker locker(&m_mutex); + m_clipMixerMap[model] = mixer; + } +} + +bool +AudioGenerator::usesClipMixer(const Model *model) +{ + bool clip = + (qobject_cast<const SparseOneDimensionalModel *>(model) || + qobject_cast<const NoteModel *>(model) || + qobject_cast<const FlexiNoteModel *>(model)); + return clip; +} + +bool +AudioGenerator::wantsQuieterClips(const Model *model) +{ + // basically, anything that usually has sustain (like notes) or + // often has multiple sounds at once (like notes) wants to use a + // quieter level than simple click tracks + bool does = + (qobject_cast<const NoteModel *>(model) || + qobject_cast<const FlexiNoteModel *>(model)); + return does; +} + +bool +AudioGenerator::usesContinuousSynth(const Model *model) +{ + bool cont = + (qobject_cast<const SparseTimeValueModel *>(model)); + return cont; +} + +ClipMixer * +AudioGenerator::makeClipMixerFor(const Model *model) +{ + QString clipId; + + const Playable *playable = model; + if (!playable || !playable->canPlay()) return 0; + + PlayParameters *parameters = + PlayParameterRepository::getInstance()->getPlayParameters(playable); + if (parameters) { + clipId = parameters->getPlayClipId(); + } + +#ifdef DEBUG_AUDIO_GENERATOR + std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): sample id = " << clipId << std::endl; +#endif + + if (clipId == "") { + SVDEBUG << "AudioGenerator::makeClipMixerFor(" << model << "): no sample, skipping" << endl; + return 0; + } + + ClipMixer *mixer = new ClipMixer(m_targetChannelCount, + m_sourceSampleRate, + m_processingBlockSize); + + double clipF0 = Pitch::getFrequencyForPitch(60, 0, 440.0); // required + + QString clipPath = QString("%1/%2.wav").arg(m_sampleDir).arg(clipId); + + double level = wantsQuieterClips(model) ? 0.5 : 1.0; + if (!mixer->loadClipData(clipPath, clipF0, level)) { + delete mixer; + return 0; + } + +#ifdef DEBUG_AUDIO_GENERATOR + std::cerr << "AudioGenerator::makeClipMixerFor(" << model << "): loaded clip " << clipId << std::endl; +#endif + + 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); + +#ifdef DEBUG_AUDIO_GENERATOR + std::cerr << "AudioGenerator::makeSynthFor(" << model << "): created synth" << std::endl; +#endif + + return synth; +} + +void +AudioGenerator::removeModel(Model *model) +{ + SparseOneDimensionalModel *sodm = + dynamic_cast<SparseOneDimensionalModel *>(model); + if (!sodm) return; // nothing to do + + QMutexLocker locker(&m_mutex); + + if (m_clipMixerMap.find(sodm) == m_clipMixerMap.end()) return; + + ClipMixer *mixer = m_clipMixerMap[sodm]; + m_clipMixerMap.erase(sodm); + delete mixer; +} + +void +AudioGenerator::clearModels() +{ + QMutexLocker locker(&m_mutex); + + while (!m_clipMixerMap.empty()) { + ClipMixer *mixer = m_clipMixerMap.begin()->second; + m_clipMixerMap.erase(m_clipMixerMap.begin()); + delete mixer; + } +} + +void +AudioGenerator::reset() +{ + QMutexLocker locker(&m_mutex); + +#ifdef DEBUG_AUDIO_GENERATOR + cerr << "AudioGenerator::reset()" << endl; +#endif + + for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { + if (i->second) { + i->second->reset(); + } + } + + m_noteOffs.clear(); +} + +void +AudioGenerator::setTargetChannelCount(int targetChannelCount) +{ + if (m_targetChannelCount == targetChannelCount) return; + +// SVDEBUG << "AudioGenerator::setTargetChannelCount(" << targetChannelCount << ")" << endl; + + QMutexLocker locker(&m_mutex); + m_targetChannelCount = targetChannelCount; + + for (ClipMixerMap::iterator i = m_clipMixerMap.begin(); i != m_clipMixerMap.end(); ++i) { + if (i->second) i->second->setChannelCount(targetChannelCount); + } +} + +sv_frame_t +AudioGenerator::getBlockSize() const +{ + return m_processingBlockSize; +} + +void +AudioGenerator::setSoloModelSet(std::set<Model *> s) +{ + QMutexLocker locker(&m_mutex); + + m_soloModelSet = s; + m_soloing = true; +} + +void +AudioGenerator::clearSoloModelSet() +{ + QMutexLocker locker(&m_mutex); + + m_soloModelSet.clear(); + m_soloing = false; +} + +sv_frame_t +AudioGenerator::mixModel(Model *model, sv_frame_t startFrame, sv_frame_t frameCount, + float **buffer, sv_frame_t fadeIn, sv_frame_t fadeOut) +{ + if (m_sourceSampleRate == 0) { + cerr << "WARNING: AudioGenerator::mixModel: No base source sample rate available" << endl; + return frameCount; + } + + QMutexLocker locker(&m_mutex); + + Playable *playable = model; + if (!playable || !playable->canPlay()) return frameCount; + + PlayParameters *parameters = + PlayParameterRepository::getInstance()->getPlayParameters(playable); + if (!parameters) return frameCount; + + bool playing = !parameters->isPlayMuted(); + if (!playing) { +#ifdef DEBUG_AUDIO_GENERATOR + cout << "AudioGenerator::mixModel(" << model << "): muted" << endl; +#endif + return frameCount; + } + + if (m_soloing) { + if (m_soloModelSet.find(model) == m_soloModelSet.end()) { +#ifdef DEBUG_AUDIO_GENERATOR + cout << "AudioGenerator::mixModel(" << model << "): not one of the solo'd models" << endl; +#endif + 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); + } + + if (usesClipMixer(model)) { + return mixClipModel(model, startFrame, frameCount, + buffer, gain, pan); + } + + 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; +} + +sv_frame_t +AudioGenerator::mixDenseTimeValueModel(DenseTimeValueModel *dtvm, + sv_frame_t startFrame, sv_frame_t frames, + float **buffer, float gain, float pan, + sv_frame_t fadeIn, sv_frame_t fadeOut) +{ + sv_frame_t maxFrames = frames + std::max(fadeIn, fadeOut); + + int modelChannels = dtvm->getChannelCount(); + + if (m_channelBufSiz < maxFrames || m_channelBufCount < modelChannels) { + + for (int c = 0; c < m_channelBufCount; ++c) { + delete[] m_channelBuffer[c]; + } + + delete[] m_channelBuffer; + m_channelBuffer = new float *[modelChannels]; + + for (int c = 0; c < modelChannels; ++c) { + m_channelBuffer[c] = new float[maxFrames]; + } + + m_channelBufCount = modelChannels; + m_channelBufSiz = maxFrames; + } + + sv_frame_t got = 0; + + if (startFrame >= fadeIn/2) { + got = dtvm->getMultiChannelData(0, modelChannels - 1, + startFrame - fadeIn/2, + frames + fadeOut/2 + fadeIn/2, + m_channelBuffer); + } else { + sv_frame_t missing = fadeIn/2 - startFrame; + + for (int c = 0; c < modelChannels; ++c) { + m_channelBuffer[c] += missing; + } + + if (missing > 0) { + cerr << "note: channelBufSiz = " << m_channelBufSiz + << ", frames + fadeOut/2 = " << frames + fadeOut/2 + << ", startFrame = " << startFrame + << ", missing = " << missing << endl; + } + + got = dtvm->getMultiChannelData(0, modelChannels - 1, + startFrame, + frames + fadeOut/2, + m_channelBuffer); + + for (int c = 0; c < modelChannels; ++c) { + m_channelBuffer[c] -= missing; + } + + got += missing; + } + + for (int c = 0; c < m_targetChannelCount; ++c) { + + int sourceChannel = (c % modelChannels); + +// SVDEBUG << "mixing channel " << c << " from source channel " << sourceChannel << endl; + + float channelGain = gain; + if (pan != 0.0) { + if (c == 0) { + if (pan > 0.0) channelGain *= 1.0f - pan; + } else { + if (pan < 0.0) channelGain *= pan + 1.0f; + } + } + + for (sv_frame_t i = 0; i < fadeIn/2; ++i) { + float *back = buffer[c]; + back -= fadeIn/2; + back[i] += + (channelGain * m_channelBuffer[sourceChannel][i] * float(i)) + / float(fadeIn); + } + + for (sv_frame_t i = 0; i < frames + fadeOut/2; ++i) { + float mult = channelGain; + if (i < fadeIn/2) { + mult = (mult * float(i)) / float(fadeIn); + } + if (i > frames - fadeOut/2) { + mult = (mult * float((frames + fadeOut/2) - i)) / float(fadeOut); + } + float val = m_channelBuffer[sourceChannel][i]; + if (i >= got) val = 0.f; + buffer[c][i] += mult * val; + } + } + + return got; +} + +sv_frame_t +AudioGenerator::mixClipModel(Model *model, + sv_frame_t startFrame, sv_frame_t frames, + float **buffer, float gain, float pan) +{ + ClipMixer *clipMixer = m_clipMixerMap[model]; + if (!clipMixer) return 0; + + int blocks = int(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 + //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 + + sv_frame_t got = blocks * m_processingBlockSize; + +#ifdef DEBUG_AUDIO_GENERATOR + cout << "mixModel [clip]: start " << startFrame << ", frames " << frames + << ", blocks " << blocks << ", have " << m_noteOffs.size() + << " note-offs" << endl; +#endif + + ClipMixer::NoteStart on; + ClipMixer::NoteEnd off; + + NoteOffSet ¬eOffs = m_noteOffs[model]; + + float **bufferIndexes = new float *[m_targetChannelCount]; + + for (int i = 0; i < blocks; ++i) { + + sv_frame_t reqStart = startFrame + i * m_processingBlockSize; + + NoteList notes; + NoteExportable *exportable = dynamic_cast<NoteExportable *>(model); + if (exportable) { + notes = exportable->getNotesWithin(reqStart, + reqStart + m_processingBlockSize); + } + + std::vector<ClipMixer::NoteStart> starts; + std::vector<ClipMixer::NoteEnd> ends; + + for (NoteList::const_iterator ni = notes.begin(); + ni != notes.end(); ++ni) { + + sv_frame_t noteFrame = ni->start; + + if (noteFrame < reqStart || + noteFrame >= reqStart + m_processingBlockSize) continue; + + while (noteOffs.begin() != noteOffs.end() && + noteOffs.begin()->frame <= noteFrame) { + + sv_frame_t eventFrame = noteOffs.begin()->frame; + if (eventFrame < reqStart) eventFrame = reqStart; + + off.frameOffset = eventFrame - reqStart; + off.frequency = noteOffs.begin()->frequency; + +#ifdef DEBUG_AUDIO_GENERATOR + cerr << "mixModel [clip]: adding note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; +#endif + + ends.push_back(off); + noteOffs.erase(noteOffs.begin()); + } + + on.frameOffset = noteFrame - reqStart; + on.frequency = ni->getFrequency(); + on.level = float(ni->velocity) / 127.0f; + on.pan = pan; + +#ifdef DEBUG_AUDIO_GENERATOR + cout << "mixModel [clip]: adding note at frame " << noteFrame << ", frame offset " << on.frameOffset << " frequency " << on.frequency << ", level " << on.level << endl; +#endif + + starts.push_back(on); + noteOffs.insert + (NoteOff(on.frequency, noteFrame + ni->duration)); + } + + while (noteOffs.begin() != noteOffs.end() && + noteOffs.begin()->frame <= reqStart + m_processingBlockSize) { + + sv_frame_t eventFrame = noteOffs.begin()->frame; + if (eventFrame < reqStart) eventFrame = reqStart; + + off.frameOffset = eventFrame - reqStart; + off.frequency = noteOffs.begin()->frequency; + +#ifdef DEBUG_AUDIO_GENERATOR + cerr << "mixModel [clip]: adding leftover note-off at frame " << eventFrame << " frame offset " << off.frameOffset << " frequency " << off.frequency << endl; +#endif + + ends.push_back(off); + noteOffs.erase(noteOffs.begin()); + } + + for (int c = 0; c < m_targetChannelCount; ++c) { + bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; + } + + clipMixer->mix(bufferIndexes, gain, starts, ends); + } + + delete[] bufferIndexes; + + return got; +} + +sv_frame_t +AudioGenerator::mixContinuousSynthModel(Model *model, + sv_frame_t startFrame, + sv_frame_t frames, + float **buffer, + float gain, + float pan) +{ + ContinuousSynth *synth = m_continuousSynthMap[model]; + if (!synth) return 0; + + // only type we support here at the moment + SparseTimeValueModel *stvm = qobject_cast<SparseTimeValueModel *>(model); + if (stvm->getScaleUnits() != "Hz") return 0; + + int blocks = int(frames / m_processingBlockSize); + + //!!! todo: see comment in mixClipModel + + sv_frame_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 (int i = 0; i < blocks; ++i) { + + sv_frame_t reqStart = startFrame + i * m_processingBlockSize; + + for (int c = 0; c < m_targetChannelCount; ++c) { + bufferIndexes[c] = buffer[c] + i * m_processingBlockSize; + } + + 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); + } + + delete[] bufferIndexes; + + return got; +} +