void RadiumCompressorAudioProcessor::setParameter (int index, float newValue)
{
bool updated = false;
switch(index){
case radium_compressor::COMP_EFF_RATIO:
updated=wrapper->update_parameter_if_different(index,get_ratio_from_slider(newValue));
break;
case radium_compressor::COMP_EFF_THRESHOLD:
updated=wrapper->update_parameter_if_different(index,get_threshold_from_slider(newValue));
break;
case radium_compressor::COMP_EFF_ATTACK:
updated=wrapper->update_parameter_if_different(index,get_attack_release_from_slider(newValue));
break;
case radium_compressor::COMP_EFF_RELEASE:
updated=wrapper->update_parameter_if_different(index,get_attack_release_from_slider(newValue));
break;
case radium_compressor::COMP_EFF_OUTPUT_VOLUME:
updated=wrapper->update_parameter_if_different(index,get_makeup_gain_from_slider(newValue));
break;
case radium_compressor::COMP_EFF_BYPASS:
{
bool old=is_bypassing;
is_bypassing = newValue>=0.5 ? true : false;
updated = old!=is_bypassing;
break;
}
}
if(updated==true){
RadiumCompressorAudioProcessorEditor *editor = dynamic_cast<RadiumCompressorAudioProcessorEditor *>(getActiveEditor()); // I guess we can use static_cast instead of dynamic_cast though.
if(editor!=NULL)
editor->update_gui(); // (update_gui() is RT safe.)
}
}
void PitchTuneAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
AudioPlayHead::CurrentPositionInfo posInfo;
bool isHostGoing_ = false;
if (getPlayHead() != 0 && getPlayHead()->getCurrentPosition(posInfo)) {
isHostGoing_ = posInfo.isPlaying;//(posInfo.isPlaying || posInfo.isRecording);
updateBpm(posInfo.bpm, posInfo.timeSigDenominator);
}
PitchTuneAudioProcessorEditor *e = (PitchTuneAudioProcessorEditor*)getActiveEditor();
if (e) {
if (e->isVisible()) {
e->ppq = posInfo.ppqPosition;
e->type = 1;
e->triggerAsyncUpdate();
}
}
if (isRecording) {
if (isHostGoing_) {
//record when host is playing
if (sampleBeingRecorded && sampleBeingRecorded->getCursor() == 0) {
// first time recording, store ppq
sampleBeingRecorded->startPpq = posInfo.ppqPosition;
//set recording flag
sampleBeingRecorded->startRecording();
}
float* channelData = buffer.getSampleData (0);
sampleBeingRecorded->record(channelData, buffer.getNumSamples());
}
else {
if (sampleBeingRecorded && sampleBeingRecorded->isRecording) {
//store stop ppq
sampleBeingRecorded->stopPpq = posInfo.ppqPosition;
//daw has stopped
stopTransferring();
//process pitch
processPitch();
}
}
}
else {
//playback the processed
float* channelData = buffer.getSampleData (0);
if (isHostGoing_) {
int nClips = (int)samples.size();
for (int i = 0; i < nClips; ++i) {
Sample *curSample = samples[i];
if (curSample ->startPpq >= posInfo.ppqPosition && !curSample ->isPlaying) {
//reach the start ppq
curSample ->startPlay();
}
if (posInfo.ppqPosition >= curSample ->stopPpq && curSample ->isPlaying) {
//reach the end ppq
curSample->stopPlay();
}
if (curSample ->isPlaying) {
curSample ->play(channelData, buffer.getNumSamples(), pitchProcessor->psola, (long)(posInfo.ppqPosition / Utility::numBeatsPerSample));
}
}
}
}
for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
{
buffer.clear (i, 0, buffer.getNumSamples());
}
lastBlockPpq = posInfo.ppqPosition;
}
void DynamicEditor::buttonUp(Vec2i mouse, unsigned char button)
{
getActiveEditor()->buttonUp(mouse,button);
}
void DynamicEditor::mouseMove(Vec2i mouse)
{
getActiveEditor()->mouseMove(mouse);
}
void MLPluginProcessor::getStateAsXML (XmlElement& xml)
{
if( !(mEngine.getCompileStatus() == MLProc::OK)) return;
#if DEMO
xml.setAttribute ("pluginVersion", JucePlugin_VersionCode);
xml.setAttribute ("presetName", String("----"));
#else
const unsigned numParams = getNumParameters();
// TODO use string properties of model instead of these JUCE strings.
// also move to JSON.
xml.setAttribute ("pluginVersion", JucePlugin_VersionCode);
xml.setAttribute ("presetName", String(getStringProperty("preset").c_str()));
xml.setAttribute ("scaleName", String(getStringProperty("key_scale").c_str()));
// store parameter values to xml as a bunch of attributes.
// not XML best practice in general but takes fewer characters.
for(unsigned i=0; i<numParams; ++i)
{
const String paramName = symbolToXMLAttr(getParameterAlias(i));
const float defaultVal = getParameterDefault(i);
const float paramVal = getParameter(i);
if (paramVal != defaultVal)
{
xml.setAttribute(paramName, paramVal);
//debug() << "setting XML param " << paramName << " to " << paramVal << "\n";
}
}
// store patcher info to xml
{
MLProcList patchers = getPatcherList();
if (!patchers.empty())
{
MLProcPatcher& firstPatcher = static_cast<MLProcPatcher&>(**patchers.begin());
const int inputs = firstPatcher.getParam("inputs");
const int outputs = firstPatcher.getParam("outputs");
String outStr;
String patcherInput = "patcher_input_";
for(unsigned i=1; i<=inputs; ++i)
{
bool differentFromDefault = false;
outStr = "";
for(unsigned j=1; j<=outputs; ++j)
{
if (firstPatcher.getConnection(i, j))
{
outStr += "1";
differentFromDefault = true;
}
else
{
outStr += "0";
}
}
if(differentFromDefault)
{
String outNum (i);
xml.setAttribute(patcherInput + outNum, outStr);
}
}
}
}
// store editor state to XML if one exists
MLPluginEditor* pEditor = static_cast<MLPluginEditor*>(getActiveEditor());
if(pEditor)
{
MLRect r = pEditor->getWindowBounds();
xml.setAttribute("editor_x", r.x());
xml.setAttribute("editor_y", r.y());
xml.setAttribute("editor_width", r.getWidth());
xml.setAttribute("editor_height", r.getHeight());
xml.setAttribute("editor_num", getFloatProperty("patch_num"));
xml.setAttribute("editor_anim", getFloatProperty("patch_anim"));
}
// save blob as most recently saved state
mpLatestStateLoaded = XmlElementPtr(new XmlElement(xml));
#endif
}
void AdmvAudioProcessor::processBlock(AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
double cp[2];
int channelCount = 0;
size_t sampleRate = getSampleRate();
for (int channel = 0; channel < (getNumInputChannels() - 1); channel += 2)
{
// No need to process signal if editor is closed
if (getActiveEditor() == NULL)
{
break;
}
// TODO: investigate how to get number of input channels really connected to the plugin ATM.
// It seems that getNumInputChannels() will always return max possible defined by JucePlugin_MaxNumInputChannels
// This solution is bad, because it iterates through all input buffers.
if (!isBlockInformative(buffer, channel / 2))
{
mGonioSegments[channel / 2] = GonioPoints<double>();
mSpectroSegments[channel / 2] = tomatl::dsp::SpectrumBlock();
continue;
}
channelCount += 2;
float* l = buffer.getWritePointer(channel + 0);
float* r = buffer.getWritePointer(channel + 1);
for (int i = 0; i < buffer.getNumSamples(); ++i)
{
std::pair<double, double>* res = mGonioCalcs[channel / 2]->handlePoint(l[i], r[i], sampleRate);
cp[0] = l[i];
cp[1] = r[i];
mSpectroCalcs[channel / 2]->checkSampleRate(getSampleRate());
tomatl::dsp::SpectrumBlock spectroResult = mSpectroCalcs[channel / 2]->process((double*)&cp);
if (res != NULL)
{
mGonioSegments[channel / 2] = GonioPoints<double>(res, mGonioCalcs[channel / 2]->getSegmentLength(), channel / 2, sampleRate);
mLastGonioScale = mGonioCalcs[channel / 2]->getCurrentScaleValue();
}
if (spectroResult.mLength > 0)
{
mSpectroSegments[channel / 2] = spectroResult;
}
}
}
mCurrentInputCount = channelCount;
if (getState().mOutputMode == AdmvPluginState::outputMute)
{
buffer.clear();
}
else
{
// 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());
}
}
}
