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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>
#include <thread>
#ifdef WIN32
#undef min
#undef max
#endif
String g_plugintitle{ "PaulXStretch 1.1.3" };
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;
};
inline AudioParameterFloat* make_floatpar(String id, String name, float minv, float maxv, float defv, float step, float skew)
{
return new AudioParameterFloat(id, name, NormalisableRange<float>(minv, maxv, step, skew), defv);
}
//==============================================================================
PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
: m_bufferingthread("pspluginprebufferthread")
{
m_playposinfo.timeInSeconds = 0.0;
m_free_filter_envelope = std::make_shared<breakpoint_envelope>();
m_free_filter_envelope->SetName("Free filter");
m_free_filter_envelope->AddNode({ 0.0,0.75 });
m_free_filter_envelope->AddNode({ 1.0,0.75 });
m_free_filter_envelope->set_reset_nodes(m_free_filter_envelope->get_all_nodes());
m_recbuffer.setSize(2, 44100);
m_recbuffer.clear();
if (m_afm->getNumKnownFormats()==0)
m_afm->registerBasicFormats();
m_thumb = std::make_unique<AudioThumbnail>(512, *m_afm, *m_thumbcache);
// The default priority of 2 is a bit too low in some cases, it seems...
m_thumbcache->getTimeSliceThread().setPriority(3);
m_sm_enab_pars[0] = new AudioParameterBool("enab_specmodule0", "Enable spectral module 1", false);
m_sm_enab_pars[1] = new AudioParameterBool("enab_specmodule1", "Enable spectral module 2", false);
m_sm_enab_pars[2] = new AudioParameterBool("enab_specmodule2", "Enable spectral module 3", true);
m_sm_enab_pars[3] = new AudioParameterBool("enab_specmodule3", "Enable spectral module 4", true);
m_sm_enab_pars[4] = new AudioParameterBool("enab_specmodule4", "Enable spectral module 5", false);
m_sm_enab_pars[5] = new AudioParameterBool("enab_specmodule5", "Enable spectral module 6", false);
m_sm_enab_pars[6] = new AudioParameterBool("enab_specmodule6", "Enable spectral module 7", true);
m_sm_enab_pars[7] = new AudioParameterBool("enab_specmodule7", "Enable spectral module 8", true);
m_sm_enab_pars[8] = new AudioParameterBool("enab_specmodule8", "Enable spectral module 9", false);
m_stretch_source = std::make_unique<StretchAudioSource>(2, m_afm,m_sm_enab_pars);
m_stretch_source->setOnsetDetection(0.0);
m_stretch_source->setLoopingEnabled(true);
m_stretch_source->setFFTWindowingType(1);
addParameter(make_floatpar("mainvolume0", "Main volume", -24.0, 12.0, -3.0, 0.1, 1.0));
addParameter(make_floatpar("stretchamount0", "Stretch amount", 0.1, 1024.0, 2.0, 0.1, 0.25));
addParameter(make_floatpar("fftsize0", "FFT size", 0.0, 1.0, 0.7, 0.01, 1.0));
addParameter(make_floatpar("pitchshift0", "Pitch shift", -24.0f, 24.0f, 0.0f, 0.1,1.0)); // 3
addParameter(make_floatpar("freqshift0", "Frequency shift", -1000.0f, 1000.0f, 0.0f, 1.0, 1.0)); // 4
addParameter(make_floatpar("playrange_start0", "Sound start", 0.0f, 1.0f, 0.0f, 0.0001,1.0)); // 5
addParameter(make_floatpar("playrange_end0", "Sound end", 0.0f, 1.0f, 1.0f, 0.0001,1.0)); // 6
addParameter(new AudioParameterBool("freeze0", "Freeze", false)); // 7
addParameter(make_floatpar("spread0", "Frequency spread", 0.0f, 1.0f, 0.0f, 0.001,1.0)); // 8
addParameter(make_floatpar("compress0", "Compress", 0.0f, 1.0f, 0.0f, 0.001,1.0)); // 9
addParameter(make_floatpar("loopxfadelen0", "Loop xfade length", 0.0f, 1.0f, 0.01f, 0.001, 1.0)); // 10
addParameter(new AudioParameterInt("numharmonics0", "Num harmonics", 1, 100, 10)); // 11
addParameter(make_floatpar("harmonicsfreq0", "Harmonics base freq", 1.0, 5000.0, 128.0, 0.1, 0.5));
addParameter(make_floatpar("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f, 0.01, 1.0)); // 13
addParameter(new AudioParameterBool("harmonicsgauss0", "Gaussian harmonics", false)); // 14
addParameter(make_floatpar("octavemixm2_0", "2 octaves down level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 15
addParameter(make_floatpar("octavemixm1_0", "Octave down level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 16
addParameter(make_floatpar("octavemix0_0", "Normal pitch level", 0.0f, 1.0f, 1.0f, 0.001, 1.0)); // 17
addParameter(make_floatpar("octavemix1_0", "1 octave up level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 18
addParameter(make_floatpar("octavemix15_0", "1 octave and fifth up level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 19
addParameter(make_floatpar("octavemix2_0", "2 octaves up level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 20
addParameter(make_floatpar("tonalvsnoisebw_0", "Tonal vs Noise BW", 0.74f, 1.0f, 0.74f, 0.001, 1.0)); // 21
addParameter(make_floatpar("tonalvsnoisepreserve_0", "Tonal vs Noise preserve", -1.0f, 1.0f, 0.5f, 0.001, 1.0)); // 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(make_floatpar("onsetdetect_0", "Onset detection", 0.0f, 1.0f, 0.0f, 0.01, 1.0)); // 25
addParameter(new AudioParameterBool("capture_enabled0", "Capture", false)); // 26
m_outchansparam = new AudioParameterInt("numoutchans0", "Num outs", 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
addParameter(new AudioParameterBool("passthrough0", "Pass input through", false)); // 30
addParameter(new AudioParameterBool("markdirty0", "Internal (don't use)", false)); // 31
m_inchansparam = new AudioParameterInt("numinchans0", "Num ins", 2, 8, 2); // 32
addParameter(m_inchansparam); // 32
addParameter(new AudioParameterBool("bypass_stretch0", "Bypass stretch", false)); // 33
addParameter(new AudioParameterFloat("freefilter_shiftx_0", "Free filter shift X", -1.0f, 1.0f, 0.0f)); // 34
addParameter(new AudioParameterFloat("freefilter_shifty_0", "Free filter shift Y", -1.0f, 1.0f, 0.0f)); // 35
addParameter(new AudioParameterFloat("freefilter_scaley_0", "Free filter scale Y", -1.0f, 1.0f, 1.0f)); // 36
addParameter(new AudioParameterFloat("freefilter_tilty_0", "Free filter tilt Y", -1.0f, 1.0f, 0.0f)); // 37
addParameter(new AudioParameterInt("freefilter_randomybands0", "Random bands", 2, 128, 16)); // 38
addParameter(new AudioParameterInt("freefilter_randomyrate0", "Random rate", 1, 32, 2)); // 39
addParameter(new AudioParameterFloat("freefilter_randomyamount0", "Random amount", 0.0, 1.0, 0.0)); // 40
for (int i = 0; i < 9; ++i) // 41-49
{
addParameter(m_sm_enab_pars[i]);
}
addParameter(make_floatpar("octavemix_extra0_0", "Ratio mix 7 level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 50
addParameter(make_floatpar("octavemix_extra1_0", "Ratio mix 8 level", 0.0f, 1.0f, 0.0f, 0.001, 1.0)); // 51
std::array<double,8> initialratios{ 0.25,0.5,1.0,2.0,3.0,4.0,1.5,1.0 / 1.5 };
// 52-59
for (int i = 0; i < 8; ++i)
{
addParameter(make_floatpar("ratiomix_ratio_"+String(i)+"_0", "Ratio mix ratio "+String(i+1), 0.125f, 8.0f,
initialratios[i],
0.001,
1.0));
}
auto& pars = getParameters();
for (const auto& p : pars)
m_reset_pars.push_back(p->getValue());
setPreBufferAmount(2);
startTimer(1, 50);
m_show_technical_info = m_propsfile->m_props_file->getBoolValue("showtechnicalinfo", false);
}
PaulstretchpluginAudioProcessor::~PaulstretchpluginAudioProcessor()
{
m_thumb->removeAllChangeListeners();
m_thumb = nullptr;
m_bufferingthread.stopThread(1000);
}
void PaulstretchpluginAudioProcessor::resetParameters()
{
ScopedLock locker(m_cs);
for (int i = 0; i < m_reset_pars.size(); ++i)
{
if (i!=cpi_main_volume && i!=cpi_passthrough)
setParameter(i, m_reset_pars[i]);
}
}
void PaulstretchpluginAudioProcessor::setPreBufferAmount(int x)
{
int temp = jlimit(0, 5, x);
if (temp != m_prebuffer_amount || m_use_backgroundbuffering == false)
{
m_use_backgroundbuffering = true;
m_prebuffer_amount = temp;
m_recreate_buffering_source = true;
ScopedLock locker(m_cs);
m_prebuffering_inited = 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_prebuffering_inited = true;
}
}
int PaulstretchpluginAudioProcessor::getPreBufferAmount()
{
if (m_use_backgroundbuffering == false)
return -1;
return m_prebuffer_amount;
}
ValueTree PaulstretchpluginAudioProcessor::getStateTree(bool ignoreoptions, bool ignorefile)
{
ValueTree paramtree("paulstretch3pluginstate");
storeToTreeProperties(paramtree, nullptr, getParameters());
if (m_current_file != File() && ignorefile == false)
{
paramtree.setProperty("importedfile", m_current_file.getFullPathName(), nullptr);
}
auto specorder = m_stretch_source->getSpectrumProcessOrder();
paramtree.setProperty("numspectralstagesb", (int)specorder.size(), nullptr);
for (int i = 0; i < specorder.size(); ++i)
{
paramtree.setProperty("specorderb" + String(i), specorder[i].m_index, nullptr);
}
if (ignoreoptions == false)
{
if (m_use_backgroundbuffering)
paramtree.setProperty("prebufamount", m_prebuffer_amount, nullptr);
else
paramtree.setProperty("prebufamount", -1, nullptr);
paramtree.setProperty("loadfilewithstate", m_load_file_with_state, nullptr);
storeToTreeProperties(paramtree, nullptr, "playwhenhostrunning", m_play_when_host_plays, "capturewhenhostrunning", m_capture_when_host_plays);
}
storeToTreeProperties(paramtree, nullptr, "waveviewrange", m_wave_view_range);
ValueTree freefilterstate = m_free_filter_envelope->saveState(Identifier("freefilter_envelope"));
paramtree.addChild(freefilterstate, -1, nullptr);
return paramtree;
}
void PaulstretchpluginAudioProcessor::setStateFromTree(ValueTree tree)
{
if (tree.isValid())
{
{
ScopedLock locker(m_cs);
ValueTree freefilterstate = tree.getChildWithName("freefilter_envelope");
m_free_filter_envelope->restoreState(freefilterstate);
m_load_file_with_state = tree.getProperty("loadfilewithstate", true);
getFromTreeProperties(tree, "playwhenhostrunning", m_play_when_host_plays, "capturewhenhostrunning", m_capture_when_host_plays);
if (tree.hasProperty("numspectralstagesb"))
{
std::vector<SpectrumProcess> old_order = m_stretch_source->getSpectrumProcessOrder();
std::vector<SpectrumProcess> new_order;
int ordersize = tree.getProperty("numspectralstagesb");
if (ordersize == old_order.size())
{
for (int i = 0; i < ordersize; ++i)
{
new_order.push_back({ (int)tree.getProperty("specorderb" + String(i)), old_order[i].m_enabled });
}
m_stretch_source->setSpectrumProcessOrder(new_order);
}
}
getFromTreeProperties(tree, "waveviewrange", m_wave_view_range);
getFromTreeProperties(tree, getParameters());
}
int prebufamt = tree.getProperty("prebufamount", 2);
if (prebufamt == -1)
m_use_backgroundbuffering = false;
else
setPreBufferAmount(prebufamt);
if (m_load_file_with_state == true)
{
String fn = tree.getProperty("importedfile");
if (fn.isEmpty() == false)
{
File f(fn);
setAudioFile(f);
}
}
m_state_dirty = 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;
}
int PaulstretchpluginAudioProcessor::getCurrentProgram()
{
return 0;
}
void PaulstretchpluginAudioProcessor::setCurrentProgram (int index)
{
}
const String PaulstretchpluginAudioProcessor::getProgramName (int index)
{
return String();
}
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);
m_stretch_source->setFreeFilterEnvelope(m_free_filter_envelope);
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());
}
void PaulstretchpluginAudioProcessor::setParameters(const std::vector<double>& pars)
{
ScopedLock locker(m_cs);
for (int i = 0; i < getNumParameters(); ++i)
{
if (i<pars.size())
setParameter(i, pars[i]);
}
}
void PaulstretchpluginAudioProcessor::updateStretchParametersFromPluginParameters(ProcessParameters & pars)
{
pars.pitch_shift.cents = *getFloatParameter(cpi_pitchshift) * 100.0;
pars.freq_shift.Hz = *getFloatParameter(cpi_frequencyshift);
pars.spread.bandwidth = *getFloatParameter(cpi_spreadamount);
pars.compressor.power = *getFloatParameter(cpi_compress);
pars.harmonics.nharmonics = *getIntParameter(cpi_numharmonics);
pars.harmonics.freq = *getFloatParameter(cpi_harmonicsfreq);
pars.harmonics.bandwidth = *getFloatParameter(cpi_harmonicsbw);
pars.harmonics.gauss = getParameter(cpi_harmonicsgauss);
pars.octave.om2 = *getFloatParameter(cpi_octavesm2);
pars.octave.om1 = *getFloatParameter(cpi_octavesm1);
pars.octave.o0 = *getFloatParameter(cpi_octaves0);
pars.octave.o1 = *getFloatParameter(cpi_octaves1);
pars.octave.o15 = *getFloatParameter(cpi_octaves15);
pars.octave.o2 = *getFloatParameter(cpi_octaves2);
pars.ratiomix.ratiolevels[0]= *getFloatParameter(cpi_octavesm2);
pars.ratiomix.ratiolevels[1] = *getFloatParameter(cpi_octavesm1);
pars.ratiomix.ratiolevels[2] = *getFloatParameter(cpi_octaves0);
pars.ratiomix.ratiolevels[3] = *getFloatParameter(cpi_octaves1);
pars.ratiomix.ratiolevels[4] = *getFloatParameter(cpi_octaves15);
pars.ratiomix.ratiolevels[5] = *getFloatParameter(cpi_octaves2);
pars.ratiomix.ratiolevels[6] = *getFloatParameter(cpi_octaves_extra1);
pars.ratiomix.ratiolevels[7] = *getFloatParameter(cpi_octaves_extra2);
for (int i = 0; i < 8; ++i)
pars.ratiomix.ratios[i] = *getFloatParameter((int)cpi_octaves_ratio0 + i);
pars.filter.low = *getFloatParameter(cpi_filter_low);
pars.filter.high = *getFloatParameter(cpi_filter_high);
pars.tonal_vs_noise.bandwidth = *getFloatParameter(cpi_tonalvsnoisebw);
pars.tonal_vs_noise.preserve = *getFloatParameter(cpi_tonalvsnoisepreserve);
}
String PaulstretchpluginAudioProcessor::offlineRender(File outputfile)
{
File outputfiletouse = outputfile.getNonexistentSibling();
int numoutchans = *getIntParameter(cpi_num_outchans);
auto ss = std::make_shared<StretchAudioSource>(numoutchans,m_afm,m_sm_enab_pars);
int blocksize = 2048;
ss->setAudioFile(m_current_file);
ProcessParameters renderpars;
updateStretchParametersFromPluginParameters(renderpars);
ss->setProcessParameters(&renderpars);
double t0 = *getFloatParameter(cpi_soundstart);
double t1 = *getFloatParameter(cpi_soundend);
sanitizeTimeRange(t0, t1);
ss->setRate(*getFloatParameter(cpi_stretchamount));
ss->setPlayRange({ t0,t1 }, true);
ss->setLoopingEnabled(true);
ss->setNumOutChannels(numoutchans);
ss->setFFTWindowingType(1);
//ss->setPreviewDry(true);
ss->setOnsetDetection(*getFloatParameter(cpi_onsetdetection));
ss->setLoopXFadeLength(*getFloatParameter(cpi_loopxfadelen));
ss->setFreezing(getParameter(cpi_freeze));
ss->setPaused(getParameter(cpi_pause_enabled));
ss->setSpectrumProcessOrder(m_stretch_source->getSpectrumProcessOrder());
ss->setFFTSize(m_fft_size_to_use);
ss->setMainVolume(*getFloatParameter(cpi_main_volume));
double outsr = getSampleRateChecked();
ss->prepareToPlay(blocksize, outsr);
WavAudioFormat wavformat;
FileOutputStream* outstream = outputfiletouse.createOutputStream();
if (outstream == nullptr)
return "Could not create output file";
auto writer = wavformat.createWriterFor(outstream, getSampleRateChecked(), numoutchans, 32, StringPairArray(), 0);
if (writer == nullptr)
{
delete outstream;
return "Could not create WAV writer";
}
auto rendertask = [ss,writer,blocksize,numoutchans, outsr, this]()
{
AudioBuffer<float> renderbuffer(numoutchans, blocksize);
int64_t outlen = 250 * outsr;
int64_t outcounter = 0;
AudioSourceChannelInfo asci(renderbuffer);
m_offline_render_state = 0;
m_offline_render_cancel_requested = false;
while (outcounter < outlen)
{
if (m_offline_render_cancel_requested == true)
break;
ss->getNextAudioBlock(asci);
writer->writeFromAudioSampleBuffer(renderbuffer, 0, blocksize);
outcounter += blocksize;
m_offline_render_state = 100.0 / outlen * outcounter;
}
m_offline_render_state = 200;
delete writer;
};
std::thread th(rendertask);
th.detach();
return "Rendered OK";
}
double PaulstretchpluginAudioProcessor::getSampleRateChecked()
{
if (m_cur_sr < 1.0 || m_cur_sr>1000000.0)
return 44100.0;
return m_cur_sr;
}
void PaulstretchpluginAudioProcessor::prepareToPlay(double sampleRate, int samplesPerBlock)
{
++m_prepare_count;
ScopedLock locker(m_cs);
m_cur_sr = sampleRate;
m_curmaxblocksize = samplesPerBlock;
m_input_buffer.setSize(getMainBusNumInputChannels(), samplesPerBlock);
int numoutchans = *m_outchansparam;
if (numoutchans != m_cur_num_out_chans)
m_prebuffering_inited = false;
if (m_using_memory_buffer == true)
{
int len = jlimit(100,m_recbuffer.getNumSamples(),
int(getSampleRateChecked()*(*getFloatParameter(cpi_max_capture_len))));
m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer,
getSampleRateChecked(),
len);
//m_thumb->reset(m_recbuffer.getNumChannels(), sampleRate, len);
}
if (m_prebuffering_inited == 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_prebuffering_inited = true;
}
else
{
m_buffering_source->prepareToPlay(samplesPerBlock, getSampleRateChecked());
}
}
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 (AudioBuffer<double>& buffer, MidiBuffer&)
{
jassert(false);
}
*/
void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
ScopedLock locker(m_cs);
AudioPlayHead* phead = getPlayHead();
if (phead != nullptr)
{
phead->getCurrentPosition(m_playposinfo);
}
else
m_playposinfo.isPlaying = false;
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 = 0; i < totalNumInputChannels; ++i)
m_input_buffer.copyFrom(i, 0, buffer, i, 0, buffer.getNumSamples());
for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
buffer.clear (i, 0, buffer.getNumSamples());
if (m_prebuffering_inited == false)
return;
if (m_is_recording == true)
{
if (m_playposinfo.isPlaying == false && m_capture_when_host_plays == true)
return;
int recbuflenframes = m_max_reclen * getSampleRate();
copyAudioBufferWrappingPosition(buffer, m_recbuffer, m_rec_pos, recbuflenframes);
m_thumb->addBlock(m_rec_pos, buffer, 0, buffer.getNumSamples());
m_rec_pos = (m_rec_pos + buffer.getNumSamples()) % recbuflenframes;
return;
}
jassert(m_buffering_source != nullptr);
jassert(m_bufferingthread.isThreadRunning());
if (m_last_host_playing == false && m_playposinfo.isPlaying)
{
m_stretch_source->seekPercent(*getFloatParameter(cpi_soundstart));
m_last_host_playing = true;
}
else if (m_last_host_playing == true && m_playposinfo.isPlaying == false)
{
m_last_host_playing = false;
}
if (m_play_when_host_plays == true && m_playposinfo.isPlaying == false)
return;
m_free_filter_envelope->m_transform_x_shift = *getFloatParameter(cpi_freefilter_shiftx);
m_free_filter_envelope->m_transform_y_shift = *getFloatParameter(cpi_freefilter_shifty);
m_free_filter_envelope->m_transform_y_scale = *getFloatParameter(cpi_freefilter_scaley);
m_free_filter_envelope->m_transform_y_tilt = *getFloatParameter(cpi_freefilter_tilty);
m_free_filter_envelope->m_transform_y_random_bands = *getIntParameter(cpi_freefilter_randomy_numbands);
m_free_filter_envelope->m_transform_y_random_rate = *getIntParameter(cpi_freefilter_randomy_rate);
m_free_filter_envelope->m_transform_y_random_amount = *getFloatParameter(cpi_freefilter_randomy_amount);
m_stretch_source->setSpectralModulesEnabled(m_sm_enab_pars);
m_stretch_source->setMainVolume(*getFloatParameter(cpi_main_volume));
m_stretch_source->setRate(*getFloatParameter(cpi_stretchamount));
m_stretch_source->setPreviewDry(*getBoolParameter(cpi_bypass_stretch));
setFFTSize(*getFloatParameter(cpi_fftsize));
updateStretchParametersFromPluginParameters(m_ppar);
m_stretch_source->setOnsetDetection(*getFloatParameter(cpi_onsetdetection));
m_stretch_source->setLoopXFadeLength(*getFloatParameter(cpi_loopxfadelen));
double t0 = *getFloatParameter(cpi_soundstart);
double t1 = *getFloatParameter(cpi_soundend);
sanitizeTimeRange(t0, t1);
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_use_backgroundbuffering == false)
{
m_stretch_source->getNextAudioBlock(aif);
}
else
{
m_buffering_source->getNextAudioBlock(aif);
}
if (getParameter(cpi_passthrough) > 0.5f)
{
for (int i = 0; i < totalNumInputChannels; ++i)
{
buffer.addFrom(i, 0, m_input_buffer, i, 0, buffer.getNumSamples());
}
}
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 = getStateTree(false,false);
MemoryOutputStream stream(destData,true);
paramtree.writeToStream(stream);
}
void PaulstretchpluginAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
{
ValueTree tree = ValueTree::readFromData(data, sizeInBytes);
setStateFromTree(tree);
}
void PaulstretchpluginAudioProcessor::setDirty()
{
toggleBool(getBoolParameter(cpi_markdirty));
}
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();
int numchans = *m_inchansparam;
m_recbuffer.setSize(numchans, m_max_reclen*getSampleRateChecked()+4096,false,false,true);
m_recbuffer.clear();
m_rec_pos = 0;
m_thumb->reset(m_recbuffer.getNumChannels(), getSampleRateChecked(), lenbufframes);
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)
{
auto ai = unique_from_raw(m_afm->createReaderFor(f));
if (ai != nullptr)
{
if (ai->numChannels > 8)
{
return "Too many channels in file "+f.getFullPathName();
}
if (ai->bitsPerSample>32)
{
return "Too high bit depth in file " + f.getFullPathName();
}
m_thumb->setSource(new FileInputSource(f));
ScopedLock locker(m_cs);
m_stretch_source->setAudioFile(f);
//Range<double> currange{ *getFloatParameter(cpi_soundstart),*getFloatParameter(cpi_soundend) };
//if (currange.contains(m_stretch_source->getInfilePositionPercent())==false)
m_stretch_source->seekPercent(*getFloatParameter(cpi_soundstart));
m_current_file = f;
m_current_file_date = m_current_file.getLastModificationTime();
m_using_memory_buffer = false;
setDirty();
return 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_prebuffering_inited = 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_prebuffering_inited = 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);
}
AudioProcessor* JUCE_CALLTYPE createPluginFilter()
{
return new PaulstretchpluginAudioProcessor();
}
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