void MidiSelectProcessor::processBlock (AudioSampleBuffer &buffer, MidiBuffer &midiMessages)
{
// Do nothing to buffer; just let it pass through
MidiBuffer inputMessages(midiMessages);
midiMessages.clear();
MidiBuffer::Iterator it(inputMessages);
while (true) {
MidiMessage message(0xf0);
int samplePosition;
if (!it.getNextEvent(message, samplePosition)) {
break;
}
if (message.isNoteOn()) {
if (findNextNote()) {
// We know it's safe to add to the list if findNextNote() returns true
transformations[message.getNoteNumber()] = currentNote;
message.setNoteNumber(currentNote);
midiMessages.addEvent(message, samplePosition);
}
else {
// This will just get skipped, but we must make note of that to also skip
// the upcomming note off event
transformations[message.getNoteNumber()] = -1;
}
}
else if (message.isNoteOff()) {
auto transformIt = transformations.find(message.getNoteNumber());
if (transformIt == transformations.end()) {
// I have no recollection of this note
continue;
}
if (transformIt->second == -1) {
// We discarded the note on, discard the note off too
transformations.erase(transformIt);
continue;
}
// Okay, make the note off match the note on, then add
message.setNoteNumber(transformIt->second);
midiMessages.addEvent(message, samplePosition);
transformations.erase(transformIt);
}
else {
// We don't mess with other events (yet), so pass on through
midiMessages.addEvent(message, samplePosition);
}
}
}
void GenericProcessor::addEvent(MidiBuffer& eventBuffer,
uint8 type,
int sampleNum,
uint8 eventId,
uint8 eventChannel,
uint8 numBytes,
uint8* eventData)
{
uint8 *data = new uint8[4+numBytes];
data[0] = type; // event type
data[1] = nodeId; // processor ID automatically added
data[2] = eventId; // event ID
data[3] = eventChannel; // event channel
memcpy(&data[4], eventData, numBytes);
eventBuffer.addEvent(data, // spike data
sizeof(data), // total bytes
sampleNum); // sample index
//if (type == TTL)
// std::cout << "Adding event for channel " << (int) eventChannel << " with ID " << (int) eventId << std::endl;
delete data;
}
void MidiKeyboardState::processNextMidiBuffer (MidiBuffer& buffer,
const int startSample,
const int numSamples,
const bool injectIndirectEvents)
{
MidiBuffer::Iterator i (buffer);
MidiMessage message;
int time;
const ScopedLock sl (lock);
while (i.getNextEvent (message, time))
processNextMidiEvent (message);
if (injectIndirectEvents)
{
MidiBuffer::Iterator i2 (eventsToAdd);
const int firstEventToAdd = eventsToAdd.getFirstEventTime();
const double scaleFactor = numSamples / (double) (eventsToAdd.getLastEventTime() + 1 - firstEventToAdd);
while (i2.getNextEvent (message, time))
{
const int pos = jlimit (0, numSamples - 1, roundToInt ((time - firstEventToAdd) * scaleFactor));
buffer.addEvent (message, startSample + pos);
}
}
eventsToAdd.clear();
}
void addCtrlrMidiMessageToBuffer (MidiBuffer &bufferToAddTo, CtrlrMidiMessage &m)
{
for (int i=0; i<m.getNumMessages(); i++)
{
bufferToAddTo.addEvent (m.getReference(i).m, 1);
}
}
void SoftSynthAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
MidiBuffer processedMidi;
int time;
MidiMessage m;
for (MidiBuffer::Iterator i(midiMessages); i.getNextEvent(m, time);) {
if (m.isNoteOn()) {
m = MidiMessage::noteOn(m.getChannel(), m.getNoteNumber(), m.getVelocity());
synth.keyPressed(m.getNoteNumber(), m.getVelocity());
} else if (m.isNoteOff()) {
m = MidiMessage::noteOff(m.getChannel(), m.getNoteNumber(), m.getVelocity());
synth.keyReleased(m.getNoteNumber());
}
processedMidi.addEvent(m, time);
}
auto synthBuffer = synth.getNextBuffer(buffer.getNumSamples());
float *leftData = buffer.getWritePointer(0);
float *rightData = buffer.getWritePointer(1);
for (int i = 0; i < buffer.getNumSamples(); ++i) {
leftData[i] = synthBuffer[i];
rightData[i] = synthBuffer[i];
}
midiMessages.swapWith(processedMidi);
}
//==============================================================================
void MidiManipulator::processEvents (MidiBuffer& midiMessages, const int blockSize)
{
MidiBuffer midiOutput;
if (! midiMessages.isEmpty ())
{
int timeStamp;
MidiMessage message (0xf4, 0.0);
MidiBuffer::Iterator it (midiMessages);
if (filter)
{
while (it.getNextEvent (message, timeStamp))
{
if (filter->filterEvent (message))
midiOutput.addEvent (message, timeStamp);
}
}
else
{
midiOutput = midiMessages;
}
midiMessages.clear ();
}
if (transform)
{
transform->processEvents (midiOutput, blockSize);
}
midiMessages = midiOutput;
}
void
CfpluginAudioProcessor::MidiPanic(MidiBuffer& midiMessages)
{
for(int i = 1; i <= 16; ++i) {
midiMessages.addEvent(MidiMessage::allNotesOff(i), 0);
midiMessages.addEvent(MidiMessage::allSoundOff(i), 0);
}
}
//==============================================================================
void BeatboxVoxAudioProcessor::processBlock(AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
const auto totalNumInputChannels = getTotalNumInputChannels();
const auto totalNumOutputChannels = getTotalNumOutputChannels();
const auto sampleRate = getSampleRate();
const auto numSamples = buffer.getNumSamples();
//Reset the noise synth if triggered
midiMessages.addEvent(MidiMessage::noteOff(1, noiseNoteNumber), 0);
classifier.processAudioBuffer(buffer.getReadPointer(0), numSamples);
//This is used for configuring the onset detector settings from the GUI
if (classifier.noteOnsetDetected())
{
if (usingOSDTestSound.load())
{
triggerOSDTestSound(midiMessages);
}
else if (classifier.getNumBuffersDelayed() > 0)
{
triggerNoise(midiMessages);
}
}
const auto sound = classifier.classify();
switch (sound)
{
case soundLabel::KickDrum:
triggerKickDrum(midiMessages);
break;
case soundLabel::SnareDrum:
triggerSnareDrum(midiMessages);
break;
case soundLabel::HiHat:
triggerHiHat(midiMessages);
break;
default: break;
}
/** Now classification complete clear the input buffer/signal.
* We only want synth response output, no a blend of input vocal
* signal + synth output.
**/
buffer.clear();
if (usingOSDTestSound.load())
osdTestSynth.renderNextBlock(buffer, midiMessages, 0, buffer.getNumSamples());
else
drumSynth.renderNextBlock(buffer, midiMessages, 0, buffer.getNumSamples());
//Not outputting midi so clear after rendering synths
midiMessages.clear();
}
/**
* Send midi note off message into buffer at given sample position.
*/
void StepSequencer::sendMidiNoteOffMessage(MidiBuffer& midiMessages, int sample)
{
if (currStepOnOff[static_cast<int>(params.seqLastPlayedStep.get())]->getStep() == eOnOffToggle::eOn)
{
MidiMessage m = MidiMessage::noteOff(1, lastPlayedNote);
midiMessages.addEvent(m, sample);
}
seqNoteIsPlaying = false;
}
void Loop::sendCurrentNoteToBuffer(MidiBuffer& buffer, int sample_number, int velocity)
{
MidiMessage m = this->getEventPointer(currentIndex)->message;
//playingNote[m.getNoteNumber()][m.getChannel()-1][0] = jlimit(0,127,m.getNoteNumber()+getTransposition());
m.setNoteNumber(jlimit(0,127,m.getNoteNumber()+getTransposition()));
m.setVelocity((((float)velocity*midiScaler * velocitySensitivity) + (1.f-velocitySensitivity)));
if (outChannel>0) m.setChannel(outChannel);
buffer.addEvent(m,sample_number);
indexOfLastNoteOn=currentIndex;
}
void GenericProcessor::setNumSamples(MidiBuffer& events, int sampleIndex) {
uint8 data[2];
data[0] = BUFFER_SIZE; // most-significant byte
data[1] = nodeId; // least-significant byte
events.addEvent(data, // spike data
2, // total bytes
sampleIndex); // sample index
}
void MidiMessageCollector::removeNextBlockOfMessages (MidiBuffer& destBuffer,
const int numSamples)
{
#if JUCE_DEBUG
jassert (hasCalledReset); // you need to call reset() to set the correct sample rate before using this object
#endif
jassert (numSamples > 0);
auto timeNow = Time::getMillisecondCounterHiRes();
auto msElapsed = timeNow - lastCallbackTime;
const ScopedLock sl (midiCallbackLock);
lastCallbackTime = timeNow;
if (! incomingMessages.isEmpty())
{
int numSourceSamples = jmax (1, roundToInt (msElapsed * 0.001 * sampleRate));
int startSample = 0;
int scale = 1 << 16;
const uint8* midiData;
int numBytes, samplePosition;
MidiBuffer::Iterator iter (incomingMessages);
if (numSourceSamples > numSamples)
{
// if our list of events is longer than the buffer we're being
// asked for, scale them down to squeeze them all in..
const int maxBlockLengthToUse = numSamples << 5;
if (numSourceSamples > maxBlockLengthToUse)
{
startSample = numSourceSamples - maxBlockLengthToUse;
numSourceSamples = maxBlockLengthToUse;
iter.setNextSamplePosition (startSample);
}
scale = (numSamples << 10) / numSourceSamples;
while (iter.getNextEvent (midiData, numBytes, samplePosition))
{
samplePosition = ((samplePosition - startSample) * scale) >> 10;
destBuffer.addEvent (midiData, numBytes,
jlimit (0, numSamples - 1, samplePosition));
}
}
else
{
/**
* Send midi note on message into buffer at given sample position.
*/
void StepSequencer::sendMidiNoteOnMessage(MidiBuffer & midiMessages, int sample)
{
if (currStepOnOff[currSeqNote]->getStep() == eOnOffToggle::eOn)
{
// emphasis on step 1
float seqVelocity = 0.5f;
//if (seqNote == 0)
//{
// seqVelocity = 0.85f;
//}
MidiMessage m = MidiMessage::noteOn(1, static_cast<int>(currMidiStepSeq[currSeqNote]->get()), seqVelocity);
midiMessages.addEvent(m, sample);
}
seqNoteIsPlaying = true;
params.seqLastPlayedStep.set(jmax(0.0f, jmin(static_cast<float>(currSeqNote), 7.0f)));
lastPlayedNote = static_cast<int>(currMidiStepSeq[currSeqNote]->get());
}
void MetronomeProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
const int numSamples = buffer.getNumSamples();
int channel;
// convert special metronome bip and bop (0xf2 00 and 0xf2 01) into note on/off messages
MidiBuffer metronomeMidiBuffer;
MidiBuffer::Iterator iter(midiMessages);
MidiMessage message;
int pos;
while (iter.getNextEvent(message,pos))
{
if (message.getRawDataSize()==2 && message.isSongPositionPointer())
{
char raw[4] = {0,0,0,0};
char* data = (char*)message.getRawData();
if (data[1]==0)
{
// bip
MidiMessage b = MidiMessage::noteOn(1,80,64.0f);
memcpy(raw,b.getRawData(),4);
}
else
{
// bop
MidiMessage b = MidiMessage::noteOn(1,70,64.0f);
memcpy(raw,b.getRawData(),4);
}
if (raw[0])
{
MidiMessage m(raw[0],raw[1],raw[2]);
//printf("m %d %d %d at %d\n",m.getRawData()[0],m.getRawData()[1],m.getRawData()[2],pos);
metronomeMidiBuffer.addEvent(m,pos);
}
}
}
// and now get the synth to process these midi events and generate its output.
synth.renderNextBlock (buffer, metronomeMidiBuffer, 0, numSamples);
}
void Pfm2AudioProcessor::handleIncomingMidiBuffer(MidiBuffer &buffer, int numberOfSamples) {
if (!buffer.isEmpty()) {
MidiBuffer newBuffer;
MidiMessage midiMessage;
int samplePosition;
MidiBuffer::Iterator midiIterator(buffer);
while (midiIterator.getNextEvent(midiMessage, samplePosition)) {
bool copyMessageInNewBuffer = true;
if (midiMessage.isController() && midiMessage.getChannel() == currentMidiChannel) {
switch (midiMessage.getControllerNumber()) {
case 99:
currentNrpn.paramMSB = midiMessage.getControllerValue();
copyMessageInNewBuffer = false;
break;
case 98:
currentNrpn.paramLSB = midiMessage.getControllerValue();
copyMessageInNewBuffer = false;
break;
case 6:
currentNrpn.valueMSB = midiMessage.getControllerValue();
copyMessageInNewBuffer = false;
break;
case 38:
{
currentNrpn.valueLSB = midiMessage.getControllerValue();
copyMessageInNewBuffer = false;
int param = (int)(currentNrpn.paramMSB << 7) + currentNrpn.paramLSB;
int value = (int)(currentNrpn.valueMSB << 7) + currentNrpn.valueLSB;
const MessageManagerLock mmLock;
handleIncomingNrpn(param, value);
break;
}
}
}
if (copyMessageInNewBuffer) {
newBuffer.addEvent(midiMessage, samplePosition);
}
}
buffer.swapWith(newBuffer);
}
}
void getNextAudioBlock (AudioSourceChannelInfo const& bufferToFill)
{
int const numSamples = bufferToFill.numSamples;
// the synth always adds its output
//bufferToFill.clearActiveBufferRegion();
m_midi.clear ();
if (m_active)
{
double const samplesPerBeat = m_sampleRate * 60 / m_tempo;
// Adjust phase so the beat is on or after the beginning of the output
double beat;
if (m_phase > 0)
beat = 1 - m_phase;
else
beat = 0 - m_phase;
// Set notes in midi buffer
for (;;beat += 1)
{
// Calc beat pos
int pos = static_cast <int> (beat * samplesPerBeat);
if (pos < numSamples)
{
m_midi.addEvent (MidiMessage::noteOn (1, 84, 1.f), pos);
}
else
{
break;
}
}
}
m_synth.renderNextBlock (*bufferToFill.buffer,
m_midi,
0,
bufferToFill.numSamples);
}
/**
* If any seqStepNote changed or is muted then send noteOff message for the old note.
*/
void StepSequencer::midiNoteChanged(MidiBuffer & midiMessages)
{
for (int i = 0; i < seqNumSteps; ++i)
{
// compare with current sequence
if ((prevMidiStepSeq[i] != static_cast<int>(currMidiStepSeq[i]->get())) || (prevStepOnOff[i] != currStepOnOff[i]->getStep()))
{
if (i == static_cast<int>(params.seqLastPlayedStep.get()))
{
MidiMessage m = MidiMessage::noteOff(1, prevMidiStepSeq[i]);
midiMessages.addEvent(m, 0);
}
}
}
// save new sequence
for (int i = 0; i < 8; ++i)
{
prevMidiStepSeq[i] = static_cast<int>(currMidiStepSeq[i]->get());
prevStepOnOff[i] = currStepOnOff[i]->getStep();
}
}
void CtrlrPanelProcessor::processBlock(MidiBuffer &midiMessages, MidiBuffer &leftoverBuffer, const AudioPlayHead::CurrentPositionInfo &info)
{
if (owner.getMidiOptionBool(panelMidiInputFromHostCompare))
{
owner.getMIDIInputThread().handleMIDIFromHost(midiMessages);
}
leftoverBuffer.clear();
processLua(midiMessages, info);
MidiBuffer::Iterator i(midiMessages);
MidiMessage m;
int time;
while (i.getNextEvent(m,time))
{
_MIN("VST INPUT", m, time);
if (owner.getMidiOptionBool(panelMidiThruH2D) == true)
{
if (owner.getMidiOptionBool(panelMidiThruH2DChannelize))
{
m.setChannel (owner.getMidiChannel(panelMidiOutputChannelDevice));
}
owner.sendMidi(m);
}
if (owner.getMidiOptionBool(panelMidiThruH2H) == true)
{
if (owner.getMidiOptionBool(panelMidiThruH2HChannelize))
{
m.setChannel (owner.getMidiChannel(panelMidiOutputChannelHost));
}
leftoverBuffer.addEvent (m, time);
}
}
}
/**
* Stop stepSequencer and reset not GUI variables.
*/
void StepSequencer::stopSeq(MidiBuffer& midiMessages)
{
// stop and reset only if not already stopped
if (!seqStopped)
{
params.seqLastPlayedStep.set(0.0f);
currSeqNote = 0;
lastPlayedNote = 0;
seqNoteAdd = 1;
nextPlaySample = 0;
noteOffSample = 0;
seqNextStep = 0.0;
stopNoteTime = 0.0;
lastPlayHeadPosition = 0.0;
seqStopped = true;
seqNoteIsPlaying = false;
// stop all midimessages from sequencer
MidiMessage m = MidiMessage::allNotesOff(1);
midiMessages.addEvent(m, 0);
}
}
void GateTrigger::processBlock (AudioSampleBuffer& buffer,
MidiBuffer& midiMessages)
{
float threshold = library->getThreshold();
int64 releaseTicks = library->getReleaseTicks();
float velocityScale = library->getVelocityScale();
int windowSize = buffer.getNumSamples();
for (int i = 0; i < buffer.getNumSamples(); i += windowSize) {
float rms = 0;
for (int chan = 0; chan < buffer.getNumChannels(); chan++) {
rms += buffer.getMagnitude (chan, i, jmin(windowSize, buffer.getNumSamples() - i));
}
rms = rms / buffer.getNumChannels() * 100;
if (rms - lastRms > threshold) {
if (Time::getHighResolutionTicks() - lastTriggerTick > releaseTicks) {
Pattern* pattern = sequencer->getPattern();
Instrument* instrument = pattern->getActiveInstrument();
// play note
float velocity = (rms - lastRms) / velocityScale;
DBG("RMS: " + String(rms) + " lastRMS: " + String(lastRms) + " velocity: " + String(velocity));
int noteNumber = instrument->getNoteNumber();
MidiMessage m = MidiMessage::noteOn (1, noteNumber, velocity);
midiMessages.addEvent (m, i);
// insert into sequencer pattern
int step = round(sequencer->getPreciseStep());
step = step % pattern->getNumSteps();
Cell* cell = pattern->getCellAt (0, step);
delayedInserterThread->insertNote (cell, velocity, instrument);
// Retrigger the reset timer
resetTimer->retrigger();
lastTriggerTick = Time::getHighResolutionTicks();
}
}
lastRms = rms;
}
}
void CtrlrPanelProcessor::processBlock(MidiBuffer &midiMessages, MidiBuffer &leftoverBuffer)
{
if (owner.getMidiOptionBool(panelMidiInputFromHostCompare))
{
owner.getMIDIInputThread().handleMIDIFromHost(midiMessages);
}
MidiBuffer::Iterator i(midiMessages);
MidiMessage m;
int time;
while (i.getNextEvent(m,time))
{
_MIN("VST INPUT", m);
if (owner.getMidiOptionBool(panelMidiThruH2D) == true)
{
if (owner.getMidiOptionBool(panelMidiThruH2DChannelize))
{
m.setChannel (owner.getMidiChannel(panelMidiOutputChannelDevice));
}
owner.sendMidi(m);
}
if (owner.getMidiOptionBool(panelMidiThruH2H) == true)
{
if (owner.getMidiOptionBool(panelMidiThruH2HChannelize))
{
m.setChannel (owner.getMidiChannel(panelMidiOutputChannelHost));
}
leftoverBuffer.addEvent (m, m.getTimeStamp());
}
}
}
void Sequencer::processBlock (AudioSampleBuffer& buffer,
MidiBuffer& midiMessages)
{
AudioPlayHead::CurrentPositionInfo pos (pluginAudioProcessor->lastPosInfo);
// If we aren't playing...
if (! pos.isPlaying) {
if (noteOffs.size() > 0) {
// Send any upcoming note-off events
Array<NoteOff*> notesToRemove;
for (int i = 0; i < noteOffs.size(); i++) {
int noteNumber = noteOffs[i]->noteNumber;
MidiMessage m2 = MidiMessage::noteOff (1, noteNumber);
midiMessages.addEvent (m2, 0);
playingNotes.set (noteNumber, false);
notesToRemove.add (noteOffs[i]);
}
for (int i = 0; i < notesToRemove.size(); i++) {
// Remove the event from the note-off event list
// (We do this in two steps so that the noteOffs array isn't modified inside a loop)
noteOffs.removeObject (notesToRemove[i], true);
}
}
return;
}
double ppq = pos.ppqPosition;
double timeInSeconds = pos.timeInSeconds;
double bpm = pos.bpm;
//if (primary) {
SharedState::getInstance()->setPpqPosition (ppq);
SharedState::getInstance()->setTimeInSeconds (timeInSeconds);
SharedState::getInstance()->setBpm (bpm);
//}
/*
int numerator = pos.timeSigNumerator;
int denominator = pos.timeSigDenominator;
const int ppqPerBar = (numerator * 4 / denominator); // e.g. 4 if 4/4
const double beats = (fmod (ppq, ppqPerBar) / ppqPerBar) * numerator;
const int bar = ((int) ppq) / ppqPerBar + 1;
const int beat = ((int) beats) + 1;
const int ticks = ((int) (fmod (beats, 1.0) * 960.0));
*/
double tickCountPrecise = fmod (ppq * speed * ticksPerCol, getTotalCols() * ticksPerCol);
tickCount = (int)tickCountPrecise;
lastPlayheadColPrecise = tickCountPrecise / ticksPerCol;
jassert (lastPlayheadColPrecise >= 0.0)
jassert (lastPlayheadColPrecise <= 16.0)
//if (primary) {
SharedState::getInstance()->setPlayheadColPrecise (lastPlayheadColPrecise);
//}
if (tickCount != lastTickCount) {
lastTickCount = tickCount;
// Check if we should be degrading...
int panelIndex = pluginAudioProcessor->getPanelIndex();
int state = SharedState::getInstance()->getState(panelIndex);
if (state == Panel::DEGRADING_SLOW ||
state == Panel::DEGRADING_FAST) {
// degrade at most once per tempo sweep, in column 0
if (getPlayheadCol() == 0) {
if (!columnZeroDegradeUpdate) {
SharedState::getInstance()->degradeStep(panelIndex);
columnZeroDegradeUpdate = true;
}
} else {
columnZeroDegradeUpdate = false;
}
} else {
columnZeroDegradeUpdate = false;
if (state == Panel::ACTIVE) {
if (SharedState::kDegradeAfterInactiveSec > 0 &&
(SharedState::getInstance()->getLastTouchElapsedMs(panelIndex) >=
SharedState::kDegradeAfterInactiveSec * 1000)) {
DBG(String(Time::currentTimeMillis()) + " "
+ "Start degrading panel " + String(panelIndex));
SharedState::getInstance()->startDegrade(panelIndex);
}
}
}
// Update starfield if necessary
if (SharedState::getInstance()->getStarFieldActive()) {
// bug:67 - for production running, we could assert that the following is true:
// SharedState::getInstance()->allAttracting()
// but that won't work if someone turns on the star field by hand via the sequencer GUI
if (pluginAudioProcessor->getPanelIndex() == 0) {
SharedState::getInstance()->updateStarField();
}
}
bool playCol = false; // play the current column of notes?
float velocity = 0.9f;
// Swing...
// If we're on an odd column
if (getPlayheadCol() % 2 != 0) {
// If we've waited for enough ticks for the swing
if (tickCount == (getPlayheadCol() * ticksPerCol) + swingTicks) {
playCol = true;
}
} else {
// Else we're on an even column
// If we're on the first tick of the column...
if (tickCount % ticksPerCol == 0) {
playCol = true;
}
}
// Calculate the latency
double beatsPerSec = bpm * speed * ticksPerCol / 60.0;
double secPerBeat = 1.0 / beatsPerSec;
double tickOffset = tickCountPrecise - tickCount;
int tickOffsetSamples = tickOffset * secPerBeat * sampleRate;
tickOffsetSamples = jmax (buffer.getNumSamples() - tickOffsetSamples - 1, 0);
// Send any upcoming note-off events
Array<NoteOff*> notesToRemove;
for (int i = 0; i < noteOffs.size(); i++) {
if (noteOffs[i]->tick == tickCount) {
int noteNumber = noteOffs[i]->noteNumber;
MidiMessage m2 = MidiMessage::noteOff (1, noteNumber);
midiMessages.addEvent (m2, tickOffsetSamples);
playingNotes.set (noteNumber, false);
notesToRemove.add (noteOffs[i]);
}
}
for (int i = 0; i < notesToRemove.size(); i++) {
// Remove the event from the note-off event list
// (We do this in two steps so that the noteOffs array isn't modified inside a loop)
noteOffs.removeObject (notesToRemove[i], true);
}
// If we should play the current column of notes
if (playCol) {
int panelIndex = pluginAudioProcessor->getPanelIndex();
int tabIndex = pluginAudioProcessor->getTabIndex();
for (int i = 0; i < getTotalRows(); i++) {
Cell* cell = getCellAt (panelIndex, tabIndex, i, getPlayheadCol());
if (cell->isOn()) {
int noteNumber = cell->getNoteNumber();
// If this note is currently playing
if (playingNotes[noteNumber] == true) {
Array<NoteOff*> notesToRemove;
// Remove any pending note-offs
for (int j = 0; j < noteOffs.size(); j++) {
if (noteOffs[j]->noteNumber == noteNumber) {
notesToRemove.add (noteOffs[j]);
}
}
for (int j = 0; j < notesToRemove.size(); j++) {
noteOffs.removeObject (notesToRemove[j], true);
}
// Send a note-off before retriggering
MidiMessage m = MidiMessage::noteOff (1, noteNumber);
midiMessages.addEvent (m, tickOffsetSamples);
playingNotes.set (noteNumber, false);
}
// Play this note
MidiMessage m = MidiMessage::noteOn (1, noteNumber, velocity);
midiMessages.addEvent (m, tickOffsetSamples);
playingNotes.set (noteNumber, true);
// Add an upcoming note-off event
NoteOff* no;
noteOffs.add (no = new NoteOff());
no->tick = (tickCount + noteLength) % (ticksPerCol * getTotalCols());
no->noteNumber = noteNumber;
}
}
}
}
}
//==============================================================================
void MidiSequencePlugin::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
MidiSequencePluginBase::processBlock(buffer, midiMessages);
MidiMessageSequence sourceMidi = *midiSequence;
std::vector<int> doneTheseControllers;
if (transport->isPlaying () && getBoolValue(PROP_SEQENABLED, true))
{
const int blockSize = buffer.getNumSamples ();
MidiBuffer* midiBuffer = midiBuffers.getUnchecked (0);
const int frameCounter = transport->getPositionInFrames ();
const int framesPerBeat = transport->getFramesPerBeat ();
const int nextBlockFrameNumber = frameCounter + blockSize;
const int seqIndex = getLoopRepeatIndex();
const double beatCount = getLoopBeatPosition();
const double frameLenBeatCount = (nextBlockFrameNumber - frameCounter) / (double)framesPerBeat;
double frameEndBeatCount = beatCount + frameLenBeatCount;
if (frameEndBeatCount > getLengthInBeats())
frameEndBeatCount -= getLengthInBeats();
// loop for each controller we need to interpolate
MidiMessage* lastCtrlEvent = NULL;
do
{
lastCtrlEvent = NULL;
// hunt for a controller event before now
int i;
for (i = 0; i < sourceMidi.getNumEvents (); i++)
{
int timeStampInSeq = roundFloatToInt (sourceMidi.getEventTime (i) * framesPerBeat);
int timeStamp = timeStampInSeq + (seqIndex * getLengthInBeats() * framesPerBeat);
MidiMessage* midiMessage = &sourceMidi.getEventPointer (i)->message;
if (timeStamp >= nextBlockFrameNumber || !midiMessage)
break; // event is after now, leave
//if (midiMessage->isController() && (std::find(doneTheseControllers.begin(), doneTheseControllers.end(), midiMessage->getControllerNumber()) == doneTheseControllers.end()))
// lastCtrlEvent = midiMessage;
}
// hunt for a matching event after that one
if (lastCtrlEvent)
{
// store the controller number so we know which controllers we've done
doneTheseControllers.push_back(lastCtrlEvent->getControllerNumber());
MidiMessage* nextCtrlEvent = NULL;
for (; i < sourceMidi.getNumEvents (); i++)
{
MidiMessage* midiMessage = &sourceMidi.getEventPointer (i)->message;
if (midiMessage->isController() && midiMessage->getControllerNumber() == lastCtrlEvent->getControllerNumber())
{
nextCtrlEvent = midiMessage;
break;
}
}
// render an interpolated event!...
if (nextCtrlEvent)
{
double bt = nextCtrlEvent->getTimeStamp();
double at = lastCtrlEvent->getTimeStamp();
double deltaBeats = bt - at;
int a = lastCtrlEvent->getControllerValue();
int b = nextCtrlEvent->getControllerValue();
double now = beatCount + (frameEndBeatCount - beatCount) / 2.0;
double interpRemainBeats = deltaBeats - (now - at);
if (deltaBeats > 0)
{
double nextPart = interpRemainBeats / deltaBeats;
nextPart = 1 - nextPart;
double interpdVal = a + nextPart * (b - a);
MidiMessage interpy = MidiMessage::controllerEvent(lastCtrlEvent->getChannel(), lastCtrlEvent->getControllerNumber(), static_cast<int>(interpdVal));
midiBuffer->addEvent (interpy, (nextBlockFrameNumber - frameCounter) / 2);
}
else
{
DBG ("Negative delta beats when rendering automation!!");
}
}
}
// now we also need to do that again if there are multiple events per frame AND we are interpolating multiple times per frame
// (at the moment only interpolating once per audio frame)
} while (lastCtrlEvent != NULL);
}
}
virtual void menuItemSelected(int menuItemID, int)
{
if (menuItemID == 200)
{
WildcardFileFilter wildcardFilter("*.mid", String::empty, "Midi files");
FileBrowserComponent browser(FileBrowserComponent::canSelectFiles | FileBrowserComponent::openMode,
lastOpenedFile.exists() ? lastOpenedFile : File(String("C:\\Users\\GeorgeKrueger\\Documents")), &wildcardFilter, nullptr);
FileChooserDialogBox dialogBox("Open a midi file",
"Please choose a midi file to open...",
browser,
false,
Colours::lightgrey);
if (dialogBox.show())
{
File selectedFile = browser.getSelectedFile(0);
lastOpenedFile = selectedFile;
FileInputStream fileStream(selectedFile);
juce::MidiFile midiFile;
midiFile.readFrom(fileStream);
int numTracks = midiFile.getNumTracks();
midiFile.convertTimestampTicksToSeconds();
String msg;
msg << "Opened midi file: " << selectedFile.getFileName() << " Tracks: " << numTracks << "\n";
log(msg);
for (int i = 0; i < numTracks; ++i)
{
const MidiMessageSequence* msgSeq = midiFile.getTrack(i);
OwnedArray<PluginDescription> results;
String plugFile = "C:\\VST\\FMMF.dll";
VSTPluginFormat vstFormat;
vstFormat.findAllTypesForFile(results, plugFile);
if (results.size() > 0) {
msg.clear();
msg << "Found " << results.size() << " plugin(s) matching file " << plugFile << "\n";
log(msg);
int secsToRender = 10;
double sampleRate = 44100;
int totalSizeInSamples = static_cast<int>(44100 * secsToRender);
AudioPluginInstance* plugInst = vstFormat.createInstanceFromDescription(*results[0], sampleRate, totalSizeInSamples);
if (!plugInst) {
msg.clear();
msg << "Failed to load plugin " << plugFile << "\n";
log(msg);
continue;
}
int numInputChannels = plugInst->getTotalNumInputChannels();
int numOutputChannels = plugInst->getTotalNumOutputChannels();
msg.clear();
msg << "Plugin input channels: " << numInputChannels << " output channels: " << numOutputChannels
<< " Current program: " << plugInst->getCurrentProgram() << "\n";
log(msg);
int maxChannels = std::max(numInputChannels, numOutputChannels);
AudioBuffer<float> buffer(maxChannels, totalSizeInSamples);
MidiBuffer midiMessages;
for (int j = 0; j < msgSeq->getNumEvents(); ++j)
{
MidiMessageSequence::MidiEventHolder* midiEventHolder = msgSeq->getEventPointer(j);
MidiMessage midiMsg = midiEventHolder->message;
int samplePos = static_cast<int>(midiMsg.getTimeStamp() * sampleRate);
midiMessages.addEvent(midiMsg, samplePos);
}
plugInst->prepareToPlay(sampleRate, totalSizeInSamples);
plugInst->processBlock(buffer, midiMessages);
/*File txtOutFile("C:\\Users\\GeorgeKrueger\\Documents\\GitHub\\soundserver2\\out.txt");
FileOutputStream* txtOutStream = txtOutFile.createOutputStream();
for (int j = 0; j < 44100; ++j)
{
float sample = buffer.getSample(0, j);
txtOutStream->writeFloat(sample);
txtOutStream->writeText(" ", true, false);
}*/
File outputFile("C:\\Users\\GeorgeKrueger\\Documents\\GitHub\\soundserver2\\out.wav");
if (outputFile.exists()) {
outputFile.deleteFile();
}
FileOutputStream* fileOutputStream = outputFile.createOutputStream();
WavAudioFormat wavFormat;
StringPairArray metadataValues;
juce::AudioFormatWriter* wavFormatWriter = wavFormat.createWriterFor(
fileOutputStream, sampleRate, 2, 16, metadataValues, 0);
bool writeAudioDataRet = wavFormatWriter->writeFromAudioSampleBuffer(buffer, 0, buffer.getNumSamples());
wavFormatWriter->flush();
msg.clear();
msg << "Done writing to output file " << outputFile.getFileName() << " . Write return value: "
<< (int)writeAudioDataRet << "\n";
log(msg);
delete wavFormatWriter;
delete plugInst;
}
else {
msg.clear();
msg << "Could not find plugin from file " << plugFile << "\n";
log(msg);
}
}
}
}
}
//==============================================================================
void MidiTransform::processEvents (MidiBuffer& midiMessages, const int blockSize)
{
int timeStamp;
MidiMessage message (0xf4, 0.0);
MidiBuffer::Iterator it (midiMessages);
MidiBuffer midiOutput;
switch (command)
{
case MidiTransform::KeepEvents:
break;
case MidiTransform::DiscardEvents:
{
midiMessages.clear ();
break;
}
case MidiTransform::RemapChannel:
{
while (it.getNextEvent (message, timeStamp))
{
message.setChannel (channelNumber);
midiOutput.addEvent (message, timeStamp);
}
midiMessages = midiOutput;
break;
}
case MidiTransform::ScaleNotes:
{
while (it.getNextEvent (message, timeStamp))
{
if (message.isNoteOnOrOff ())
{
message.setNoteNumber (roundFloatToInt (message.getNoteNumber () * noteScale));
midiOutput.addEvent (message, timeStamp);
}
}
midiMessages = midiOutput;
break;
}
case MidiTransform::InvertNotes:
{
while (it.getNextEvent (message, timeStamp))
{
if (message.isNoteOnOrOff ())
{
message.setNoteNumber (127 - message.getNoteNumber ());
midiOutput.addEvent (message, timeStamp);
}
}
midiMessages = midiOutput;
}
case MidiTransform::TransposeNotes:
{
while (it.getNextEvent (message, timeStamp))
{
if (message.isNoteOnOrOff ())
{
message.setNoteNumber (jmax (0, jmin (127, message.getNoteNumber () - noteTranspose)));
midiOutput.addEvent (message, timeStamp);
}
}
midiMessages = midiOutput;
break;
}
case MidiTransform::ScaleVelocity:
{
while (it.getNextEvent (message, timeStamp))
{
if (message.isNoteOn ())
{
message.setVelocity ((message.getVelocity () / 127.0f) * velocityScale);
midiOutput.addEvent (message, timeStamp);
}
}
midiMessages = midiOutput;
break;
}
case MidiTransform::InvertVelocity:
{
while (it.getNextEvent (message, timeStamp))
{
if (message.isNoteOn ())
{
message.setVelocity ((uint8) (127 - message.getVelocity ()));
midiOutput.addEvent (message, timeStamp);
}
}
midiMessages = midiOutput;
break;
}
case MidiTransform::TransposeVelocity:
{
while (it.getNextEvent (message, timeStamp))
{
if (message.isNoteOn ())
{
message.setVelocity (jmax (0, jmin (127, message.getVelocity () - velocityTranspose)));
midiOutput.addEvent (message, timeStamp);
}
}
midiMessages = midiOutput;
break;
}
case MidiTransform::TriggerCC:
{
break;
}
case MidiTransform::TriggerNote:
{
break;
}
}
}
bool handlePluginRequest(const PluginRequestParameters ¶ms, OutputStream &ostream,
ThreadSafePlugin *plugin = nullptr) {
if (!plugin) {
// It's very possible that all of this was a premature optimization.
// For VSTs at least, code loading and caching is handled by ModuleHandle::findOrCreateModule,
// and each instantiation only requires a couple of disc hits for working directory setting.
// On the other hand, we want to make sure that each audio request has a "fresh" instance.
// The easiest way to do this is by bypassing the instance pool and instantiating on demand.
#if PLUGIN_POOL_SIZE
// Recurse with a plugin from the pool, locking on it.
// Keep trying with a delay until a timeout occurs.
const int TIMEOUT = 5000, WAIT = 200;
int64 startTime = Time::currentTimeMillis();
while (Time::currentTimeMillis() < startTime + TIMEOUT) {
int i = 0;
while ((plugin = pluginPool[i++])) {
const ScopedTryLock pluginTryLock(plugin->crit);
if (pluginTryLock.isLocked()) {
DBG << "Handling with plugin " << i << endl;
return handlePluginRequest(params, ostream, plugin);
}
}
DBG << "Trying again in " << WAIT << endl;
Thread::sleep(WAIT);
}
// If we were unable to obtain a lock, return failure.
DBG << "Timeout" << endl;
return false;
#else
ThreadSafePlugin temporaryPlugin(createSynthInstance());
return handlePluginRequest(params, ostream, &temporaryPlugin);
#endif
}
else {
// Re-acquire or acquire the lock.
const ScopedLock pluginLock(plugin->crit);
AudioPluginInstance *instance = plugin->instance; // unmanaged, for simplicity
// Attempt to reset the plugin in all ways possible.
instance->reset();
// Setting default parameters here causes miniTERA to become unresponsive to parameter settings.
// It's possible that it's effectively pressing some interface buttons that change the editor mode entirely.
// It's not necessary anyways if the plugin instance has been freshly created (see above).
// pluginParametersOldNewFallback(instance, nullptr, &pluginDefaults); // note that the defaults may be empty
instance->setCurrentProgram(0);
// Load preset if specified, before listing or modifying parameters!
if (params.presetNumber >= 0 && params.presetNumber < instance->getNumPrograms()) {
DBG << "Setting program/preset: " << params.presetNumber << endl;
instance->setCurrentProgram(params.presetNumber);
}
int currentProgram = instance->getCurrentProgram();
DBG << "Current program/preset: " << currentProgram << " - " << instance->getProgramName(currentProgram) << endl;
// Set parameters, starting with named, then indexed
pluginParametersSet(instance, params.parameters);
pluginParametersSetIndexed(instance, params.indexedParameters);
// If parameters requested, output them and return
if (params.listParameters) {
DBG << "Rendering parameter list: # parameters " << instance->getNumPrograms() << endl;
// Output each parameter setting in two places:
// an indexed array and a dictionary by name
// All DynamicObjects created will be freed when their var's leave scope.
DynamicObject *outer = new DynamicObject();
DynamicObject *innerParams = new DynamicObject();
var indexedParamVar;
{
for (int i = 0, n = instance->getNumParameters(); i < n; ++i) {
String name = instance->getParameterName(i);
float val = instance->getParameter(i);
innerParams->setProperty(name, val);
DynamicObject *indexedInnerObj = new DynamicObject();
indexedInnerObj->setProperty("index", i);
indexedInnerObj->setProperty("name", name);
indexedInnerObj->setProperty("value", val);
indexedParamVar.append(var(indexedInnerObj)); // frees indexedInnerObj when this scope ends
}
}
outer->setProperty(Identifier("parameters"), var(innerParams));
outer->setProperty(Identifier("indexedParameters"), indexedParamVar);
// List presets/programs.
var progVar;
{
for (int i = 0, n = instance->getNumPrograms(); i < n; ++i) {
progVar.append(var(instance->getProgramName(i)));
}
}
outer->setProperty(Identifier("presets"), progVar);
var outerVar(outer);
JSON::writeToStream(ostream, outerVar);
// DBG << JSON::toString(outerVar, true /* allOnOneLine */) << endl;
return true;
}
// Now attempt to render audio.
AudioFormatManager formatManager;
formatManager.registerBasicFormats();
OptionalScopedPointer<AudioFormat> outputFormat(formatManager.findFormatForFileExtension(params.getFormatName()), false);
if (!outputFormat) return false;
instance->setNonRealtime(true);
instance->prepareToPlay(params.sampleRate, params.blockSize);
instance->setNonRealtime(true);
// The writer takes ownership of the output stream; the writer will delete it when the writer leaves scope.
// Therefore, we pass a special pointer class that does not allow the writer to delete it.
OutputStream *ostreamNonDeleting = new NonDeletingOutputStream(&ostream);
ScopedPointer<AudioFormatWriter> writer(outputFormat->createWriterFor(ostreamNonDeleting,
params.sampleRate, params.nChannels, params.bitDepth,
StringPairArray(), 0));
// Create a MIDI buffer
MidiBuffer midiBuffer;
midiBuffer.addEvent(MidiMessage::noteOn(params.midiChannel, (uint8)params.midiPitch, (uint8)params.midiVelocity),
0 /* time */);
midiBuffer.addEvent(MidiMessage::allNotesOff(params.midiChannel),
params.noteSeconds * params.sampleRate);
AudioSampleBuffer buffer(params.nChannels, params.blockSize);
int numBuffers = (int)(params.renderSeconds * params.sampleRate / params.blockSize);
for (int i = 0; i < numBuffers; ++i) {
// DBG << "Processing block " << i << "..." << flush;
instance->processBlock(buffer, midiBuffer);
// DBG << " left RMS level " << buffer.getRMSLevel(0, 0, params.blockSize) << endl;
writer->writeFromAudioSampleBuffer(buffer, 0 /* offset into buffer */, params.blockSize);
}
instance->reset();
return true;
}
}
void MidiOutFilter::processBlock (AudioSampleBuffer& buffer,
MidiBuffer& midiMessages)
{
for (int i = 0; i < getNumOutputChannels(); ++i)
{
buffer.clear (i, 0, buffer.getNumSamples());
}
const double SR=getSampleRate();
const double iSR=1.0/SR;
AudioPlayHead::CurrentPositionInfo pos;
if (getPlayHead() != 0 && getPlayHead()->getCurrentPosition (pos))
{
if (memcmp (&pos, &lastPosInfo, sizeof (pos)) != 0)
{
if(param[kMTC]>=0.5f) {
double frameRate=24.0;
int mtcFrameRate=0;
const double samplesPerPpq=60.0*SR/pos.bpm;
const double samplesPerClock = SR/(4.0*frameRate);
const long double seconds = (long double)(pos.ppqPosition*60.0f/pos.bpm) /*+ smpteOffset*/;
const long double absSecs = fabs (seconds);
const bool neg = seconds < 0.0;
int hours, mins, secs, frames;
if (frameRate==29.97) {
int64 frameNumber = int64(absSecs*29.97);
frameNumber += 18*(frameNumber/17982) + 2*(((frameNumber%17982) - 2) / 1798);
hours = int((((frameNumber / 30) / 60) / 60) % 24);
mins = int(((frameNumber / 30) / 60) % 60);
secs = int((frameNumber / 30) % 60);
frames = int(frameNumber % 30);
}
else {
hours = (int) (absSecs / (60.0 * 60.0));
mins = ((int) (absSecs / 60.0)) % 60;
secs = ((int) absSecs) % 60;
frames = (int)(int64(absSecs*frameRate) % (int)frameRate);
}
if (pos.isPlaying)
{
double i=0.0;
const double clockppq = fmod(absSecs*frameRate*4.0,(long double)1.0);
samplesToNextMTC = (int)(samplesPerClock * (clockppq+i));
i+=1.0;
if (!wasPlaying) {
//this is so the song position pointer will be sent before any
//other data at the beginning of the song
MidiBuffer temp = midiMessages;
midiMessages.clear();
if (samplesToNextMTC<buffer.getNumSamples()) {
int mtcData;
switch (mtcNumber)
{
case 0: mtcData=frames&0x0f; break;
case 1: mtcData=(frames&0xf0)>>4; break;
case 2: mtcData=secs&0x0f; break;
case 3: mtcData=(secs&0xf0)>>4; break;
case 4: mtcData=mins&0x0f; break;
case 5: mtcData=(mins&0xf0)>>4; break;
case 6: mtcData=hours&0x0f; break;
case 7: mtcData=(hours&0x10)>>4 | mtcFrameRate; break;
}
MidiMessage midiclock(0xf1,(mtcNumber<<4)|(mtcData));
++mtcNumber;
mtcNumber&=0x07;
midiMessages.addEvent(midiclock,samplesToNextMTC);
samplesToNextMTC = (int)(samplesPerClock * (clockppq+i));
i+=1.0;
startMTCAt=-999.0;
sendmtc=true;
}
midiMessages.addEvents(temp,0,buffer.getNumSamples(),0);
}
if (startMTCAt >-999.0 && (int)(samplesPerPpq*(startMTCAt-pos.ppqPosition))<buffer.getNumSamples()) {
samplesToNextMTC = (int)(samplesPerPpq*(startMTCAt-pos.ppqPosition));
int mtcData;
switch (mtcNumber)
{
case 0: mtcData=frames&0x0f; break;
case 1: mtcData=(frames&0xf0)>>4; break;
case 2: mtcData=secs&0x0f; break;
case 3: mtcData=(secs&0xf0)>>4; break;
case 4: mtcData=mins&0x0f; break;
case 5: mtcData=(mins&0xf0)>>4; break;
case 6: mtcData=hours&0x0f; break;
case 7: mtcData=(hours&0x10)>>4 | mtcFrameRate; break;
}
MidiMessage midiclock(0xf1,(mtcNumber<<4)|(mtcData));
++mtcNumber;
mtcNumber&=0x07;
midiMessages.addEvent(midiclock,samplesToNextMTC);
samplesToNextMTC = (int)(samplesPerClock * (clockppq+i));
i+=1.0;
startMTCAt=-999.0;
sendmtc=true;
}
void CfpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
RTContext rt; // Debug RT stuff in debug mode
auto timeAtStartOfBlock = time::now();
{
// This will allocate IFF there is an error.
// Can't put the non-rt specifier within the while...
NonRTSection nonRt;
while (globals_.ErrorBuffer()->dequeue(errorString_)) {
juce::NativeMessageBox::showMessageBoxAsync(
juce::AlertWindow::WarningIcon,
juce::String("Error!"),
juce::String(errorString_.c_str()));
}
}
/************************************************************************************/
// Update UI on every run, TODO make this happen every n times?
changeBroadcaster.sendChangeMessage();
if (resetting_.load()) { return; }
/************************************************************************************/
if (!follower_) { return; }
unsigned trackCount = follower_->StartNewBlock();
if (trackCount == 0) {
MidiPanic(midiMessages);
return;
}
// Track 0 is supposed to be the tempo track
for (unsigned i = 1; i < trackCount; ++i) {
follower_->GetTrackEventsForBlock(i, eventBuffer_);
auto events = eventBuffer_.AllEvents();
events.ForEach([this, &midiMessages, i](samples_t sample, ScoreEventPtr message)
{
assert(sample.value() >= 0);
assert(sample.value() < samplesPerBlock_);
MidiMessage const & msg = midi_event_cast(message)->Message();
// Prohibit all but note on and off for now
if (!msg.isNoteOnOrOff()) {
return;
}
// truncating is probably the right thing to do...
midiMessages.addEvent(msg, static_cast<unsigned>(sample.value()));
});
}
/*****************************************************************************************/
// This is the place where you'd normally do the guts of your plugin's
// audio processing...
/*
for (int channel = 0; channel < getNumInputChannels(); ++channel)
{
float* channelData = buffer.getSampleData (channel);
}
*/
// In case we have more outputs than inputs, we'll clear any output
// channels that didn't contain input data, (because these aren't
// guaranteed to be empty - they may contain garbage).
/*
for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
{
buffer.clear (i, 0, buffer.getNumSamples());
}
*/
// However, we reset all channels
for (int i = 0; i < getNumOutputChannels(); ++i)
{
buffer.clear (i, 0, buffer.getNumSamples());
}
/******************************/
time_quantity elapsedTime = time_cast<time_quantity>(time::now() - timeAtStartOfBlock);
time_quantity blockSize((samplesPerBlock_ * score::samples) / (samplerate_ * score::samples_per_second));
if (elapsedTime > (0.3 * blockSize)) {
LOG("Possible xrun! Process callback took %1% (max: %2%)", elapsedTime, blockSize);
}
}
void LumaPlug::processBlock (AudioSampleBuffer& buffer,
MidiBuffer& midiMessages)
{
// we don't want any midi input events
midiMessages.clear();
bool isRunning = false;
AudioPlayHead* playHead = getPlayHead();
if (playHead)
{
//printf("playhead: 0x%x\n", playHead);
AudioPlayHead::CurrentPositionInfo posInfo;
playHead->getCurrentPosition(posInfo);
isRunning = posInfo.isPlaying || posInfo.isRecording;
luma_->SetBPM(posInfo.bpm);
}
if (isRunning && !luma_->IsRunning())
{
std::string error;
bool setScriptOK = luma_->SetScript(scriptText_.toUTF8(), error);
if (!setScriptOK && getActiveEditor())
{
((LumaPlugEditor*)getActiveEditor())->Log(error.c_str());
}
// else if (getActiveEditor())
// {
// ((LumaPlugEditor*)getActiveEditor())->Log("Play");
// }
luma_->Start();
}
else if (!isRunning && luma_->IsRunning())
{
luma_->Stop();
// if (getActiveEditor())
// {
// ((LumaPlugEditor*)getActiveEditor())->Log("Stop");
// }
}
if (luma_->IsRunning())
{
double sampleRate = getSampleRate();
int numSamples = buffer.getNumSamples();
float elapsed = (float(numSamples) / float(sampleRate)) * 1000.0;
//printf("Elapsed: %f\n", elapsed);
vector<shared_ptr<LumaEvent> > events;
vector<float> offsets;
luma_->Update(elapsed, events, offsets);
if (events.size() > 0)
{
for (unsigned int i = 0; i < events.size(); i++)
{
//printf("New Event. Offset = %d, OffsetSamples = %d, Type = %d\n\n",
// offsets[i], midiEvent->deltaFrames, events[i]->GetType());
int eventOffset = lroundf( ( float(offsets[i]) / 1000.0 ) * sampleRate );
if (events[i]->GetType() == kLumaEvent_NoteOn)
{
NoteOnEvent* noteOn = (NoteOnEvent*)events[i].get();
MidiMessage msg = MidiMessage::noteOn(1, noteOn->GetPitch(), (juce::uint8)noteOn->GetVelocity());
midiMessages.addEvent(msg, eventOffset);
}
else if (events[i]->GetType() == kLumaEvent_NoteOff)
{
NoteOffEvent* noteOff = (NoteOffEvent*)events[i].get();
MidiMessage msg = MidiMessage::noteOff(1, noteOff->GetPitch());
midiMessages.addEvent(msg, eventOffset);
}
else
{
fprintf(stderr, "LumaVST: Unknown event type: %d\n", events[i]->GetType());
}
}
// clear the used luma events
events.clear();
}
}
/*
Simple test of sending midi from the plugin
static int count = 0;
count += buffer.getNumSamples();
if (count >= 20000)
{
//MidiMessage msg = MidiMessage::noteOff(0, 60);
//midiMessages.addEvent(msg, 0);
MidiMessage msg = MidiMessage::noteOn(1, 60, (juce::uint8)100);
midiMessages.addEvent(msg, 0);
count = 0;
}
*/
}
void MiditoOscAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
static float cv[8], shiftcv[8];
static bool _calibMode;
MidiBuffer processedMidi;
MidiMessage m;
int time;
char oscBuffer[IP_MTU_SIZE];
osc::OutboundPacketStream p(oscBuffer, IP_MTU_SIZE);
if (calibMode) // Calibration Mode A440Hz(MIDI number 69)
{
p << osc::BeginBundleImmediate
<< osc::BeginMessage( "/fader1" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader2" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader3" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader4" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader5" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader6" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader7" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/fader8" )
<< calibMap[69] << osc::EndMessage
<< osc::BeginMessage( "/gate1" )
<< 1 << osc::EndMessage
<< osc::BeginMessage( "/gate2" )
<< 1 << osc::EndMessage
<< osc::EndBundle;
sendOSCData(p);
_calibMode = true;
return;
} else {
if (_calibMode)
{
p << osc::BeginBundleImmediate
<< osc::BeginMessage( "/gate1" )
<< 0 << osc::EndMessage
<< osc::BeginMessage( "/gate2" )
<< 0 << osc::EndMessage
<< osc::EndBundle;
sendOSCData(p);
_calibMode = false;
}
}
for (MidiBuffer::Iterator i (midiMessages); i.getNextEvent (m, time);)
{
p.Clear();
usleep(30);
if (m.isNoteOn())
{
if (monoMode) // mono Mode
{
uint32_t midiCh = m.getChannel();
if (midiCh == 0 || midiCh > 7)
{
midiCh = 1;
}
cv[midiCh - 1] = calibMap[m.getNoteNumber()];
switch (midiCh)
{
case 1:
p << osc::BeginMessage("/fader1")
<< cv[0] << osc::EndMessage;
break;
case 2:
p << osc::BeginMessage("/fader2")
<< cv[1] << osc::EndMessage;
break;
case 3:
p << osc::BeginMessage("/fader3")
<< cv[2] << osc::EndMessage;
break;
case 4:
p << osc::BeginMessage("/fader4")
<< cv[3] << osc::EndMessage;
break;
case 5:
p << osc::BeginMessage("/fader5")
<< cv[4] << osc::EndMessage;
break;
case 6:
p << osc::BeginMessage("/fader6")
<< cv[5] << osc::EndMessage;
break;
case 7:
p << osc::BeginMessage("/fader7")
<< cv[6] << osc::EndMessage;
break;
case 8:
p << osc::BeginMessage("/fader8")
<< cv[7] << osc::EndMessage;
break;
default:
break;
}
sendOSCData(p);
} else if (shiftMode) { // shift Mode
cv[0] = calibMap[m.getNoteNumber()];
for (int i = 7; i > 0; i--)
{
shiftcv[i] = shiftcv[i-1];
}
p << osc::BeginBundleImmediate
<< osc::BeginMessage( "/fader1" )
<< cv[0] << osc::EndMessage
<< osc::BeginMessage( "/fader2" )
<< shiftcv[1] << osc::EndMessage
<< osc::BeginMessage( "/fader3" )
<< shiftcv[2] << osc::EndMessage
<< osc::BeginMessage( "/fader4" )
<< shiftcv[3] << osc::EndMessage
<< osc::BeginMessage( "/fader5" )
<< shiftcv[4] << osc::EndMessage
<< osc::BeginMessage( "/fader6" )
<< shiftcv[5] << osc::EndMessage
<< osc::BeginMessage( "/fader7" )
<< shiftcv[6] << osc::EndMessage
<< osc::BeginMessage( "/fader8" )
<< shiftcv[7] << osc::EndMessage
<< osc::BeginMessage( "/gate1" )
<< 1 << osc::EndMessage
<< osc::BeginMessage( "/gate2" )
<< 1 << osc::EndMessage
<< osc::EndBundle;
sendOSCData(p);
shiftcv[0] = cv[0];
} else { // poly Mode
cv[ch] = calibMap[m.getNoteNumber()];
if (currentMaxPoly == 1)
{
cv[1] = cv[0];
}
p << osc::BeginBundleImmediate
<< osc::BeginMessage( "/fader1" )
<< cv[0] << osc::EndMessage
<< osc::BeginMessage( "/fader2" )
<< cv[1] << osc::EndMessage
<< osc::BeginMessage( "/fader3" )
<< cv[2] << osc::EndMessage
<< osc::BeginMessage( "/fader4" )
<< cv[3] << osc::EndMessage
<< osc::BeginMessage( "/fader5" )
<< cv[4] << osc::EndMessage
<< osc::BeginMessage( "/fader6" )
<< cv[5] << osc::EndMessage
<< osc::BeginMessage( "/fader7" )
<< m.getFloatVelocity() << osc::EndMessage
<< osc::BeginMessage( "/gate1" )
<< 1 << osc::EndMessage
<< osc::BeginMessage( "/gate2" )
<< 1 << osc::EndMessage
<< osc::EndBundle;
sendOSCData(p);
ch++;
gateCount++;
if (ch >= currentMaxPoly)
{
ch = 0;
}
}
} else if (m.isNoteOff()) {
if (monoMode)
{
switch (m.getChannel())
{
case 1:
p << osc::BeginMessage( "/gate1" )
<< 0 << osc::EndMessage;
break;
case 2:
p << osc::BeginMessage( "/gate2" )
<< 0 << osc::EndMessage;
break;
case 3:
p << osc::BeginMessage( "/gate3" )
<< 0 << osc::EndMessage;
break;
case 4:
p << osc::BeginMessage( "/gate4" )
<< 0 << osc::EndMessage;
break;
default:
break;
}
sendOSCData(p);
} else if (shiftMode) {
p << osc::BeginBundleImmediate
<< osc::BeginMessage( "/gate1" )
<< 0 << osc::EndMessage
<< osc::BeginMessage( "/gate2" )
<< 0 << osc::EndMessage
<< osc::EndBundle;
sendOSCData(p);
} else {
gateCount --;
if (gateCount <= 0)
{
p << osc::BeginBundleImmediate
<< osc::BeginMessage( "/gate1" )
<< 0 << osc::EndMessage
<< osc::BeginMessage( "/gate2" )
<< 0 << osc::EndMessage
<< osc::EndBundle;
sendOSCData(p);
gateCount = 0;
}
ch--;
if (ch == -1)
{
ch = 0;
}
}
} else if (m.isControllerOfType(1)) { // Modulation Wheel
float modulation = m.getControllerValue();
if (!monoMode && !shiftMode)
{
p << osc::BeginMessage("/fader8")
<< (modulation / 127) << osc::EndMessage;
sendOSCData(p);
}
}
processedMidi.addEvent (m, time);
}
midiMessages.swapWith (processedMidi);
buffer.clear();
for (int channel = 0; channel < getNumInputChannels(); ++channel)
{
float* channelData = 0;
}
}