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paulxstretch/Source/PluginProcessor.cpp

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/*
Copyright (C) 2006-2011 Nasca Octavian Paul
Author: Nasca Octavian Paul
Copyright (C) 2017 Xenakios
This program is free software; you can redistribute it and/or modify
it under the terms of version 2 of the GNU General Public License
as published by the Free Software Foundation.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License (version 2) for more details.
You should have received a copy of the GNU General Public License (version 2)
along with this program; if not, write to the Free Software Foundation,
Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "PluginProcessor.h"
#include "PluginEditor.h"
#include <set>
#ifdef WIN32
#undef min
#undef max
#endif
std::set<PaulstretchpluginAudioProcessor*> g_activeprocessors;
template<typename F>
void callGUI(AudioProcessor* ap, F&& f, bool async)
{
auto ed = dynamic_cast<PaulstretchpluginAudioProcessorEditor*>(ap->getActiveEditor());
if (ed)
{
if (async == false)
f(ed);
else
MessageManager::callAsync([ed,f]() { f(ed); });
}
}
int get_optimized_updown(int n, bool up) {
int orig_n = n;
while (true) {
n = orig_n;
while (!(n % 11)) n /= 11;
while (!(n % 7)) n /= 7;
while (!(n % 5)) n /= 5;
while (!(n % 3)) n /= 3;
while (!(n % 2)) n /= 2;
if (n<2) break;
if (up) orig_n++;
else orig_n--;
if (orig_n<4) return 4;
};
return orig_n;
};
int optimizebufsize(int n) {
int n1 = get_optimized_updown(n, false);
int n2 = get_optimized_updown(n, true);
if ((n - n1)<(n2 - n)) return n1;
else return n2;
};
//==============================================================================
PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
: m_bufferingthread("pspluginprebufferthread")
#ifndef JucePlugin_PreferredChannelConfigurations
: AudioProcessor (BusesProperties()
#if ! JucePlugin_IsMidiEffect
#if ! JucePlugin_IsSynth
.withInput ("Input", AudioChannelSet::stereo(), true)
#endif
.withOutput ("Output", AudioChannelSet::stereo(), true)
#endif
)
#endif
{
g_activeprocessors.insert(this);
m_recbuffer.setSize(2, 44100);
m_recbuffer.clear();
if (m_afm->getNumKnownFormats()==0)
m_afm->registerBasicFormats();
m_stretch_source = std::make_unique<StretchAudioSource>(2, m_afm);
setPreBufferAmount(2);
m_ppar.pitch_shift.enabled = true;
m_ppar.freq_shift.enabled = true;
m_ppar.filter.enabled = true;
m_ppar.compressor.enabled = true;
m_stretch_source->setOnsetDetection(0.0);
m_stretch_source->setLoopingEnabled(true);
m_stretch_source->setFFTWindowingType(1);
addParameter(new AudioParameterFloat("mainvolume0", "Main volume", -24.0f, 12.0f, -3.0f)); // 0
addParameter(new AudioParameterFloat("stretchamount0", "Stretch amount",
NormalisableRange<float>(0.1f, 1024.0f, 0.01f, 0.25),1.0f)); // 1
addParameter(new AudioParameterFloat("fftsize0", "FFT size", 0.0f, 1.0f, 0.7f)); // 2
addParameter(new AudioParameterFloat("pitchshift0", "Pitch shift", -24.0f, 24.0f, 0.0f)); // 3
addParameter(new AudioParameterFloat("freqshift0", "Frequency shift", -1000.0f, 1000.0f, 0.0f)); // 4
addParameter(new AudioParameterFloat("playrange_start0", "Sound start", 0.0f, 1.0f, 0.0f)); // 5
addParameter(new AudioParameterFloat("playrange_end0", "Sound end", 0.0f, 1.0f, 1.0f)); // 6
addParameter(new AudioParameterBool("freeze0", "Freeze", false)); // 7
addParameter(new AudioParameterFloat("spread0", "Frequency spread", 0.0f, 1.0f, 0.0f)); // 8
addParameter(new AudioParameterFloat("compress0", "Compress", 0.0f, 1.0f, 0.0f)); // 9
addParameter(new AudioParameterFloat("loopxfadelen0", "Loop xfade length", 0.0f, 1.0f, 0.0f)); // 10
addParameter(new AudioParameterFloat("numharmonics0", "Num harmonics", 0.0f, 100.0f, 0.0f)); // 11
addParameter(new AudioParameterFloat("harmonicsfreq0", "Harmonics base freq",
NormalisableRange<float>(1.0f, 5000.0f, 1.00f, 0.5), 128.0f)); // 12
addParameter(new AudioParameterFloat("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f)); // 13
addParameter(new AudioParameterBool("harmonicsgauss0", "Gaussian harmonics", false)); // 14
addParameter(new AudioParameterFloat("octavemixm2_0", "2 octaves down level", 0.0f, 1.0f, 0.0f)); // 15
addParameter(new AudioParameterFloat("octavemixm1_0", "Octave down level", 0.0f, 1.0f, 0.0f)); // 16
addParameter(new AudioParameterFloat("octavemix0_0", "Normal pitch level", 0.0f, 1.0f, 1.0f)); // 17
addParameter(new AudioParameterFloat("octavemix1_0", "1 octave up level", 0.0f, 1.0f, 0.0f)); // 18
addParameter(new AudioParameterFloat("octavemix15_0", "1 octave and fifth up level", 0.0f, 1.0f, 0.0f)); // 19
addParameter(new AudioParameterFloat("octavemix2_0", "2 octaves up level", 0.0f, 1.0f, 0.0f)); // 20
addParameter(new AudioParameterFloat("tonalvsnoisebw_0", "Tonal vs Noise BW", 0.74f, 1.0f, 0.74f)); // 21
addParameter(new AudioParameterFloat("tonalvsnoisepreserve_0", "Tonal vs Noise preserve", -1.0f, 1.0f, 0.5f)); // 22
auto filt_convertFrom0To1Func = [](float rangemin, float rangemax, float value)
{
if (value < 0.5f)
return jmap<float>(value, 0.0f, 0.5f, 20.0f, 1000.0f);
return jmap<float>(value, 0.5f, 1.0f, 1000.0f, 20000.0f);
};
auto filt_convertTo0To1Func = [](float rangemin, float rangemax, float value)
{
if (value < 1000.0f)
return jmap<float>(value, 20.0f, 1000.0f, 0.0f, 0.5f);
return jmap<float>(value, 1000.0f, 20000.0f, 0.5f, 1.0f);
};
addParameter(new AudioParameterFloat("filter_low_0", "Filter low",
NormalisableRange<float>(20.0f, 20000.0f,
filt_convertFrom0To1Func, filt_convertTo0To1Func), 20.0f)); // 23
addParameter(new AudioParameterFloat("filter_high_0", "Filter high",
NormalisableRange<float>(20.0f, 20000.0f,
filt_convertFrom0To1Func,filt_convertTo0To1Func), 20000.0f));; // 24
addParameter(new AudioParameterFloat("onsetdetect_0", "Onset detection", 0.0f, 1.0f, 0.0f)); // 25
addParameter(new AudioParameterBool("capture_enabled0", "Capture", false)); // 26
m_outchansparam = new AudioParameterInt("numoutchans0", "Num output channels", 2, 8, 2); // 27
addParameter(m_outchansparam); // 27
addParameter(new AudioParameterBool("pause_enabled0", "Pause", false)); // 28
addParameter(new AudioParameterFloat("maxcapturelen_0", "Max capture length", 1.0f, 120.0f, 10.0f)); // 29
startTimer(1, 50);
}
PaulstretchpluginAudioProcessor::~PaulstretchpluginAudioProcessor()
{
g_activeprocessors.erase(this);
m_bufferingthread.stopThread(1000);
}
void PaulstretchpluginAudioProcessor::setPreBufferAmount(int x)
{
int temp = jlimit(0, 5, x);
if (temp != m_prebuffer_amount)
{
m_prebuffer_amount = temp;
m_recreate_buffering_source = true;
}
}
//==============================================================================
const String PaulstretchpluginAudioProcessor::getName() const
{
return JucePlugin_Name;
}
bool PaulstretchpluginAudioProcessor::acceptsMidi() const
{
#if JucePlugin_WantsMidiInput
return true;
#else
return false;
#endif
}
bool PaulstretchpluginAudioProcessor::producesMidi() const
{
#if JucePlugin_ProducesMidiOutput
return true;
#else
return false;
#endif
}
bool PaulstretchpluginAudioProcessor::isMidiEffect() const
{
#if JucePlugin_IsMidiEffect
return true;
#else
return false;
#endif
}
double PaulstretchpluginAudioProcessor::getTailLengthSeconds() const
{
return 0.0;
//return (double)m_bufamounts[m_prebuffer_amount]/getSampleRate();
}
int PaulstretchpluginAudioProcessor::getNumPrograms()
{
return 1; // NB: some hosts don't cope very well if you tell them there are 0 programs,
// so this should be at least 1, even if you're not really implementing programs.
}
int PaulstretchpluginAudioProcessor::getCurrentProgram()
{
return 0;
}
void PaulstretchpluginAudioProcessor::setCurrentProgram (int index)
{
}
const String PaulstretchpluginAudioProcessor::getProgramName (int index)
{
return {};
}
void PaulstretchpluginAudioProcessor::changeProgramName (int index, const String& newName)
{
}
void PaulstretchpluginAudioProcessor::setFFTSize(double size)
{
if (m_prebuffer_amount == 5)
m_fft_size_to_use = pow(2, 7.0 + size * 14.5);
else m_fft_size_to_use = pow(2, 7.0 + size * 10.0); // chicken out from allowing huge FFT sizes if not enough prebuffering
int optim = optimizebufsize(m_fft_size_to_use);
m_fft_size_to_use = optim;
m_stretch_source->setFFTSize(optim);
//Logger::writeToLog(String(m_fft_size_to_use));
}
void PaulstretchpluginAudioProcessor::startplay(Range<double> playrange, int numoutchans, int maxBlockSize, String& err)
{
m_stretch_source->setPlayRange(playrange, true);
int bufamt = m_bufamounts[m_prebuffer_amount];
if (m_buffering_source != nullptr && numoutchans != m_buffering_source->getNumberOfChannels())
m_recreate_buffering_source = true;
if (m_recreate_buffering_source == true)
{
m_buffering_source = std::make_unique<MyBufferingAudioSource>(m_stretch_source.get(),
m_bufferingthread, false, bufamt, numoutchans, false);
m_recreate_buffering_source = false;
}
if (m_bufferingthread.isThreadRunning() == false)
m_bufferingthread.startThread();
m_stretch_source->setNumOutChannels(numoutchans);
m_stretch_source->setFFTSize(m_fft_size_to_use);
m_stretch_source->setProcessParameters(&m_ppar);
m_last_outpos_pos = 0.0;
m_last_in_pos = playrange.getStart()*m_stretch_source->getInfileLengthSeconds();
m_buffering_source->prepareToPlay(maxBlockSize, getSampleRateChecked());
}
double PaulstretchpluginAudioProcessor::getSampleRateChecked()
{
if (m_cur_sr < 1.0)
return 44100.0;
return m_cur_sr;
}
void PaulstretchpluginAudioProcessor::prepareToPlay(double sampleRate, int samplesPerBlock)
{
ScopedLock locker(m_cs);
m_cur_sr = sampleRate;
m_curmaxblocksize = samplesPerBlock;
int numoutchans = *m_outchansparam;
if (numoutchans != m_cur_num_out_chans)
m_ready_to_play = false;
if (m_using_memory_buffer == true)
{
int len = jlimit(100,m_recbuffer.getNumSamples(), m_rec_pos);
m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer,
getSampleRateChecked(),
len);
callGUI(this,[this,len](auto ed) { ed->setAudioBuffer(&m_recbuffer, getSampleRateChecked(), len); },false);
}
if (m_ready_to_play == false)
{
setFFTSize(*getFloatParameter(cpi_fftsize));
m_stretch_source->setProcessParameters(&m_ppar);
m_stretch_source->setFFTWindowingType(1);
String err;
startplay({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) },
numoutchans, samplesPerBlock, err);
m_cur_num_out_chans = numoutchans;
m_ready_to_play = true;
}
}
void PaulstretchpluginAudioProcessor::releaseResources()
{
//m_control->stopplay();
//m_ready_to_play = false;
}
#ifndef JucePlugin_PreferredChannelConfigurations
bool PaulstretchpluginAudioProcessor::isBusesLayoutSupported (const BusesLayout& layouts) const
{
#if JucePlugin_IsMidiEffect
ignoreUnused (layouts);
return true;
#else
// This is the place where you check if the layout is supported.
// In this template code we only support mono or stereo.
if (layouts.getMainOutputChannelSet() != AudioChannelSet::mono()
&& layouts.getMainOutputChannelSet() != AudioChannelSet::stereo())
return false;
// This checks if the input layout matches the output layout
#if ! JucePlugin_IsSynth
if (layouts.getMainOutputChannelSet() != layouts.getMainInputChannelSet())
return false;
#endif
return true;
#endif
}
#endif
void copyAudioBufferWrappingPosition(const AudioBuffer<float>& src, AudioBuffer<float>& dest, int destbufpos, int maxdestpos)
{
for (int i = 0; i < dest.getNumChannels(); ++i)
{
int channel_to_copy = i % src.getNumChannels();
if (destbufpos + src.getNumSamples() > maxdestpos)
{
int wrappos = (destbufpos + src.getNumSamples()) % maxdestpos;
int partial_len = src.getNumSamples() - wrappos;
dest.copyFrom(channel_to_copy, destbufpos, src, channel_to_copy, 0, partial_len);
dest.copyFrom(channel_to_copy, partial_len, src, channel_to_copy, 0, wrappos);
}
else
{
dest.copyFrom(channel_to_copy, destbufpos, src, channel_to_copy, 0, src.getNumSamples());
}
}
}
void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
ScopedLock locker(m_cs);
ScopedNoDenormals noDenormals;
double srtemp = getSampleRate();
if (srtemp != m_cur_sr)
m_cur_sr = srtemp;
const int totalNumInputChannels = getTotalNumInputChannels();
const int totalNumOutputChannels = getTotalNumOutputChannels();
for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
buffer.clear (i, 0, buffer.getNumSamples());
if (m_ready_to_play == false)
return;
if (m_is_recording == true)
{
int recbuflenframes = m_max_reclen * getSampleRate();
copyAudioBufferWrappingPosition(buffer, m_recbuffer, m_rec_pos, recbuflenframes);
callGUI(this,[this, &buffer](PaulstretchpluginAudioProcessorEditor*ed)
{
ed->addAudioBlock(buffer, getSampleRate(), m_rec_pos);
}, false);
m_rec_pos = (m_rec_pos + buffer.getNumSamples()) % recbuflenframes;
return;
}
jassert(m_buffering_source != nullptr);
jassert(m_bufferingthread.isThreadRunning());
m_stretch_source->setMainVolume(*getFloatParameter(cpi_main_volume));
m_stretch_source->setRate(*getFloatParameter(cpi_stretchamount));
setFFTSize(*getFloatParameter(cpi_fftsize));
m_ppar.pitch_shift.cents = *getFloatParameter(cpi_pitchshift) * 100.0;
m_ppar.freq_shift.Hz = *getFloatParameter(cpi_frequencyshift);
m_ppar.spread.enabled = *getFloatParameter(cpi_spreadamount) > 0.0f;
m_ppar.spread.bandwidth = *getFloatParameter(cpi_spreadamount);
m_ppar.compressor.enabled = *getFloatParameter(cpi_compress)>0.0f;
m_ppar.compressor.power = *getFloatParameter(cpi_compress);
m_ppar.harmonics.enabled = *getFloatParameter(cpi_numharmonics)>=1.0;
m_ppar.harmonics.nharmonics = *getFloatParameter(cpi_numharmonics);
m_ppar.harmonics.freq = *getFloatParameter(cpi_harmonicsfreq);
m_ppar.harmonics.bandwidth = *getFloatParameter(cpi_harmonicsbw);
m_ppar.harmonics.gauss = getParameter(cpi_harmonicsgauss);
m_ppar.octave.om2 = *getFloatParameter(cpi_octavesm2);
m_ppar.octave.om1 = *getFloatParameter(cpi_octavesm1);
m_ppar.octave.o0 = *getFloatParameter(cpi_octaves0);
m_ppar.octave.o1 = *getFloatParameter(cpi_octaves1);
m_ppar.octave.o15 = *getFloatParameter(cpi_octaves15);
m_ppar.octave.o2 = *getFloatParameter(cpi_octaves2);
m_ppar.octave.enabled = true;
m_ppar.filter.low = *getFloatParameter(cpi_filter_low);
m_ppar.filter.high = *getFloatParameter(cpi_filter_high);
m_ppar.tonal_vs_noise.enabled = (*getFloatParameter(cpi_tonalvsnoisebw)) > 0.75;
m_ppar.tonal_vs_noise.bandwidth = *getFloatParameter(cpi_tonalvsnoisebw);
m_ppar.tonal_vs_noise.preserve = *getFloatParameter(cpi_tonalvsnoisepreserve);
m_stretch_source->setOnsetDetection(*getFloatParameter(cpi_onsetdetection));
m_stretch_source->setLoopXFadeLength(*getFloatParameter(cpi_loopxfadelen));
double t0 = *getFloatParameter(cpi_soundstart);
double t1 = *getFloatParameter(cpi_soundend);
if (t0 > t1)
std::swap(t0, t1);
if (t1 - t0 < 0.001)
t1 = t0 + 0.001;
m_stretch_source->setPlayRange({ t0,t1 }, true);
m_stretch_source->setFreezing(getParameter(cpi_freeze));
m_stretch_source->setPaused(getParameter(cpi_pause_enabled));
m_stretch_source->setProcessParameters(&m_ppar);
AudioSourceChannelInfo aif(buffer);
if (isNonRealtime())
{
m_stretch_source->getNextAudioBlock(aif);
}
else
{
m_buffering_source->getNextAudioBlock(aif);
}
for (int i = 0; i < buffer.getNumChannels(); ++i)
{
for (int j = 0; j < buffer.getNumSamples(); ++j)
{
float sample = buffer.getSample(i,j);
if (std::isnan(sample) || std::isinf(sample))
++m_abnormal_output_samples;
}
}
}
//==============================================================================
bool PaulstretchpluginAudioProcessor::hasEditor() const
{
return true; // (change this to false if you choose to not supply an editor)
}
AudioProcessorEditor* PaulstretchpluginAudioProcessor::createEditor()
{
return new PaulstretchpluginAudioProcessorEditor (*this);
}
//==============================================================================
void PaulstretchpluginAudioProcessor::getStateInformation (MemoryBlock& destData)
{
ValueTree paramtree("paulstretch3pluginstate");
for (int i=0;i<getNumParameters();++i)
{
auto par = getFloatParameter(i);
if (par != nullptr)
{
paramtree.setProperty(par->paramID, (double)*par, nullptr);
}
}
paramtree.setProperty(m_outchansparam->paramID, (int)*m_outchansparam, nullptr);
if (m_current_file != File())
{
paramtree.setProperty("importedfile", m_current_file.getFullPathName(), nullptr);
}
auto specorder = m_stretch_source->getSpectrumProcessOrder();
paramtree.setProperty("numspectralstages", (int)specorder.size(), nullptr);
for (int i = 0; i < specorder.size(); ++i)
{
paramtree.setProperty("specorder" + String(i), specorder[i], nullptr);
}
MemoryOutputStream stream(destData,true);
paramtree.writeToStream(stream);
}
void PaulstretchpluginAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
{
ValueTree tree = ValueTree::readFromData(data, sizeInBytes);
if (tree.isValid())
{
{
ScopedLock locker(m_cs);
if (tree.hasProperty("numspectralstages"))
{
std::vector<int> order;
int ordersize = tree.getProperty("numspectralstages");
for (int i = 0; i < ordersize; ++i)
{
order.push_back((int)tree.getProperty("specorder" + String(i)));
}
m_stretch_source->setSpectrumProcessOrder(order);
}
for (int i = 0; i < getNumParameters(); ++i)
{
auto par = getFloatParameter(i);
if (par != nullptr)
{
double parval = tree.getProperty(par->paramID, (double)*par);
*par = parval;
}
}
if (tree.hasProperty(m_outchansparam->paramID))
*m_outchansparam = tree.getProperty(m_outchansparam->paramID, 2);
}
String fn = tree.getProperty("importedfile");
if (fn.isEmpty() == false)
{
File f(fn);
setAudioFile(f);
}
}
}
void PaulstretchpluginAudioProcessor::setRecordingEnabled(bool b)
{
ScopedLock locker(m_cs);
int lenbufframes = getSampleRateChecked()*m_max_reclen;
if (b == true)
{
m_using_memory_buffer = true;
m_current_file = File();
m_recbuffer.setSize(2, m_max_reclen*getSampleRateChecked()+4096,false,false,true);
m_recbuffer.clear();
m_rec_pos = 0;
callGUI(this,[this,lenbufframes](PaulstretchpluginAudioProcessorEditor* ed)
{
ed->beginAddingAudioBlocks(2, getSampleRateChecked(), lenbufframes);
},false);
m_is_recording = true;
}
else
{
if (m_is_recording == true)
{
finishRecording(lenbufframes);
}
}
}
double PaulstretchpluginAudioProcessor::getRecordingPositionPercent()
{
if (m_is_recording==false)
return 0.0;
return 1.0 / m_recbuffer.getNumSamples()*m_rec_pos;
}
String PaulstretchpluginAudioProcessor::setAudioFile(File f)
{
//if (f==File())
// return String();
//if (f==m_current_file && f.getLastModificationTime()==m_current_file_date)
// return String();
auto ai = unique_from_raw(m_afm->createReaderFor(f));
if (ai != nullptr)
{
if (ai->numChannels > 32)
{
//MessageManager::callAsync([cb, file]() { cb("Too many channels in file " + file.getFullPathName()); });
return "Too many channels in file "+f.getFullPathName();
}
if (ai->bitsPerSample>32)
{
//MessageManager::callAsync([cb, file]() { cb("Too high bit depth in file " + file.getFullPathName()); });
return "Too high bit depth in file " + f.getFullPathName();
}
ScopedLock locker(m_cs);
m_stretch_source->setAudioFile(f);
m_current_file = f;
m_current_file_date = m_current_file.getLastModificationTime();
m_using_memory_buffer = false;
return String();
//MessageManager::callAsync([cb, file]() { cb(String()); });
}
return "Could not open file " + f.getFullPathName();
}
Range<double> PaulstretchpluginAudioProcessor::getTimeSelection()
{
return { *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) };
}
double PaulstretchpluginAudioProcessor::getPreBufferingPercent()
{
if (m_buffering_source==nullptr)
return 0.0;
return m_buffering_source->getPercentReady();
}
void PaulstretchpluginAudioProcessor::timerCallback(int id)
{
if (id == 1)
{
bool capture = getParameter(cpi_capture_enabled);
if (capture == false && m_max_reclen != *getFloatParameter(cpi_max_capture_len))
{
m_max_reclen = *getFloatParameter(cpi_max_capture_len);
//Logger::writeToLog("Changing max capture len to " + String(m_max_reclen));
}
if (capture == true && m_is_recording == false)
{
setRecordingEnabled(true);
return;
}
if (capture == false && m_is_recording == true)
{
setRecordingEnabled(false);
return;
}
if (m_cur_num_out_chans != *m_outchansparam)
{
jassert(m_curmaxblocksize > 0);
ScopedLock locker(m_cs);
m_ready_to_play = false;
m_cur_num_out_chans = *m_outchansparam;
//Logger::writeToLog("Switching to use " + String(m_cur_num_out_chans) + " out channels");
String err;
startplay({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) },
m_cur_num_out_chans, m_curmaxblocksize, err);
m_ready_to_play = true;
}
}
}
void PaulstretchpluginAudioProcessor::finishRecording(int lenrecording)
{
m_is_recording = false;
m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer, getSampleRateChecked(), lenrecording);
m_stretch_source->setPlayRange({ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) }, true);
auto ed = dynamic_cast<PaulstretchpluginAudioProcessorEditor*>(getActiveEditor());
if (ed)
{
//ed->setAudioBuffer(&m_recbuffer, getSampleRate(), lenrecording);
}
}
//==============================================================================
// This creates new instances of the plugin..
AudioProcessor* JUCE_CALLTYPE createPluginFilter()
{
return new PaulstretchpluginAudioProcessor();
}
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