539 lines
18 KiB
C++
539 lines
18 KiB
C++
/*
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==============================================================================
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This file was auto-generated!
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It contains the basic framework code for a JUCE plugin processor.
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==============================================================================
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*/
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#include "PluginProcessor.h"
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#include "PluginEditor.h"
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#include <set>
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#ifdef WIN32
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#undef min
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#undef max
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#endif
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std::set<PaulstretchpluginAudioProcessor*> g_activeprocessors;
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template<typename F>
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void callGUI(AudioProcessor* ap, F&& f, bool async)
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{
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auto ed = dynamic_cast<PaulstretchpluginAudioProcessorEditor*>(ap->getActiveEditor());
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if (ed)
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{
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if (async == false)
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f(ed);
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else
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MessageManager::callAsync([ed,f]() { f(ed); });
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}
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}
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int get_optimized_updown(int n, bool up) {
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int orig_n = n;
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while (true) {
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n = orig_n;
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while (!(n % 11)) n /= 11;
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while (!(n % 7)) n /= 7;
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while (!(n % 5)) n /= 5;
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while (!(n % 3)) n /= 3;
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while (!(n % 2)) n /= 2;
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if (n<2) break;
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if (up) orig_n++;
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else orig_n--;
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if (orig_n<4) return 4;
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};
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return orig_n;
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};
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int optimizebufsize(int n) {
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int n1 = get_optimized_updown(n, false);
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int n2 = get_optimized_updown(n, true);
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if ((n - n1)<(n2 - n)) return n1;
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else return n2;
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};
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//==============================================================================
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PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
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: m_bufferingthread("pspluginprebufferthread")
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#ifndef JucePlugin_PreferredChannelConfigurations
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: AudioProcessor (BusesProperties()
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#if ! JucePlugin_IsMidiEffect
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#if ! JucePlugin_IsSynth
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.withInput ("Input", AudioChannelSet::stereo(), true)
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#endif
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.withOutput ("Output", AudioChannelSet::stereo(), true)
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#endif
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)
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#endif
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{
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g_activeprocessors.insert(this);
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m_recbuffer.setSize(2, 44100);
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m_recbuffer.clear();
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if (m_afm->getNumKnownFormats()==0)
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m_afm->registerBasicFormats();
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m_stretch_source = std::make_unique<StretchAudioSource>(2, m_afm);
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setPreBufferAmount(2);
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m_ppar.pitch_shift.enabled = true;
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m_ppar.freq_shift.enabled = true;
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m_ppar.filter.enabled = true;
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m_stretch_source->setOnsetDetection(0.0);
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m_stretch_source->setLoopingEnabled(true);
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m_stretch_source->setFFTWindowingType(1);
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addParameter(new AudioParameterFloat("mainvolume0", "Main volume", -24.0f, 12.0f, -3.0f)); // 0
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addParameter(new AudioParameterFloat("stretchamount0", "Stretch amount",
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NormalisableRange<float>(0.1f, 128.0f, 0.01f, 0.5),1.0f)); // 1
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addParameter(new AudioParameterFloat("fftsize0", "FFT size", 0.0f, 1.0f, 0.7f)); // 2
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addParameter(new AudioParameterFloat("pitchshift0", "Pitch shift", -24.0f, 24.0f, 0.0f)); // 3
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addParameter(new AudioParameterFloat("freqshift0", "Frequency shift", -1000.0f, 1000.0f, 0.0f)); // 4
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addParameter(new AudioParameterFloat("playrange_start0", "Sound start", 0.0f, 1.0f, 0.0f)); // 5
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addParameter(new AudioParameterFloat("playrange_end0", "Sound end", 0.0f, 1.0f, 1.0f)); // 6
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addParameter(new AudioParameterBool("freeze0", "Freeze", false)); // 7
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addParameter(new AudioParameterFloat("spread0", "Frequency spread", 0.0f, 1.0f, 0.0f)); // 8
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addParameter(new AudioParameterFloat("compress0", "Compress", 0.0f, 1.0f, 0.0f)); // 9
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addParameter(new AudioParameterFloat("loopxfadelen0", "Loop xfade length", 0.0f, 1.0f, 0.0f)); // 10
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addParameter(new AudioParameterFloat("numharmonics0", "Num harmonics", 0.0f, 100.0f, 0.0f)); // 11
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addParameter(new AudioParameterFloat("harmonicsfreq0", "Harmonics base freq",
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NormalisableRange<float>(1.0f, 5000.0f, 1.00f, 0.5), 128.0f)); // 12
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addParameter(new AudioParameterFloat("harmonicsbw0", "Harmonics bandwidth", 0.1f, 200.0f, 25.0f)); // 13
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addParameter(new AudioParameterBool("harmonicsgauss0", "Gaussian harmonics", false)); // 14
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addParameter(new AudioParameterFloat("octavemixm2_0", "2 octaves down level", 0.0f, 1.0f, 0.0f)); // 15
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addParameter(new AudioParameterFloat("octavemixm1_0", "Octave down level", 0.0f, 1.0f, 0.0f)); // 16
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addParameter(new AudioParameterFloat("octavemix0_0", "Normal pitch level", 0.0f, 1.0f, 1.0f)); // 17
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addParameter(new AudioParameterFloat("octavemix1_0", "1 octave up level", 0.0f, 1.0f, 0.0f)); // 18
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addParameter(new AudioParameterFloat("octavemix15_0", "1 octave and fifth up level", 0.0f, 1.0f, 0.0f)); // 19
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addParameter(new AudioParameterFloat("octavemix2_0", "2 octaves up level", 0.0f, 1.0f, 0.0f)); // 20
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addParameter(new AudioParameterFloat("tonalvsnoisebw_0", "Tonal vs Noise BW", 0.74f, 1.0f, 0.74f)); // 21
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addParameter(new AudioParameterFloat("tonalvsnoisepreserve_0", "Tonal vs Noise preserve", -1.0f, 1.0f, 0.5f)); // 22
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addParameter(new AudioParameterFloat("filter_low_0", "Filter low", 20.0f, 10000.0f, 20.0f)); // 23
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addParameter(new AudioParameterFloat("filter_high_0", "Filter high", 20.0f, 20000.0f, 20000.0f)); // 24
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addParameter(new AudioParameterFloat("onsetdetect_0", "Onset detection", 0.0f, 1.0f, 0.0f)); // 25
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addParameter(new AudioParameterBool("capture_enabled0", "Capture", false)); // 26
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startTimer(1, 50);
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}
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PaulstretchpluginAudioProcessor::~PaulstretchpluginAudioProcessor()
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{
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g_activeprocessors.erase(this);
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m_bufferingthread.stopThread(1000);
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}
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void PaulstretchpluginAudioProcessor::setPreBufferAmount(int x)
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{
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int temp = jlimit(0, 5, x);
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if (temp != m_prebuffer_amount)
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{
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m_prebuffer_amount = temp;
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m_recreate_buffering_source = true;
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}
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}
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//==============================================================================
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const String PaulstretchpluginAudioProcessor::getName() const
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{
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return JucePlugin_Name;
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}
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bool PaulstretchpluginAudioProcessor::acceptsMidi() const
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{
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#if JucePlugin_WantsMidiInput
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return true;
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#else
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return false;
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#endif
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}
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bool PaulstretchpluginAudioProcessor::producesMidi() const
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{
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#if JucePlugin_ProducesMidiOutput
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return true;
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#else
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return false;
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#endif
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}
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bool PaulstretchpluginAudioProcessor::isMidiEffect() const
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{
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#if JucePlugin_IsMidiEffect
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return true;
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#else
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return false;
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#endif
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}
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double PaulstretchpluginAudioProcessor::getTailLengthSeconds() const
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{
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return (double)m_bufamounts[m_prebuffer_amount]/getSampleRate();
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}
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int PaulstretchpluginAudioProcessor::getNumPrograms()
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{
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return 1; // NB: some hosts don't cope very well if you tell them there are 0 programs,
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// so this should be at least 1, even if you're not really implementing programs.
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}
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int PaulstretchpluginAudioProcessor::getCurrentProgram()
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{
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return 0;
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}
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void PaulstretchpluginAudioProcessor::setCurrentProgram (int index)
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{
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}
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const String PaulstretchpluginAudioProcessor::getProgramName (int index)
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{
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return {};
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}
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void PaulstretchpluginAudioProcessor::changeProgramName (int index, const String& newName)
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{
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}
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void PaulstretchpluginAudioProcessor::setFFTSize(double size)
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{
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if (m_prebuffer_amount == 5)
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m_fft_size_to_use = pow(2, 7.0 + size * 14.5);
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else m_fft_size_to_use = pow(2, 7.0 + size * 10.0); // chicken out from allowing huge FFT sizes if not enough prebuffering
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int optim = optimizebufsize(m_fft_size_to_use);
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m_fft_size_to_use = optim;
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m_stretch_source->setFFTSize(optim);
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//Logger::writeToLog(String(m_fft_size_to_use));
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}
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void PaulstretchpluginAudioProcessor::startplay(Range<double> playrange, int numoutchans, String& err)
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{
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m_stretch_source->setPlayRange(playrange, true);
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int bufamt = m_bufamounts[m_prebuffer_amount];
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if (m_buffering_source != nullptr && numoutchans != m_buffering_source->getNumberOfChannels())
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m_recreate_buffering_source = true;
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if (m_recreate_buffering_source == true)
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{
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m_buffering_source = std::make_unique<MyBufferingAudioSource>(m_stretch_source.get(),
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m_bufferingthread, false, bufamt, numoutchans, false);
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m_recreate_buffering_source = false;
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}
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if (m_bufferingthread.isThreadRunning() == false)
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m_bufferingthread.startThread();
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m_stretch_source->setNumOutChannels(numoutchans);
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m_stretch_source->setFFTSize(m_fft_size_to_use);
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m_stretch_source->setProcessParameters(&m_ppar);
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m_last_outpos_pos = 0.0;
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m_last_in_pos = playrange.getStart()*m_stretch_source->getInfileLengthSeconds();
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m_buffering_source->prepareToPlay(1024, 44100.0);
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};
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void PaulstretchpluginAudioProcessor::prepareToPlay(double sampleRate, int samplesPerBlock)
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{
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ScopedLock locker(m_cs);
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if (getNumOutputChannels() != m_cur_num_out_chans)
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m_ready_to_play = false;
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if (m_using_memory_buffer == true)
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{
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int len = jlimit(100,m_recbuffer.getNumSamples(), m_rec_pos);
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m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer,
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getSampleRate(),
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len);
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callGUI(this,[this,len](auto ed) { ed->setAudioBuffer(&m_recbuffer, getSampleRate(), len); },false);
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}
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if (m_ready_to_play == false)
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{
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setFFTSize(*getFloatParameter(2));
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m_stretch_source->setProcessParameters(&m_ppar);
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String err;
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startplay({ *getFloatParameter(5),*getFloatParameter(6) },
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2, err);
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m_cur_num_out_chans = getNumOutputChannels();
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m_ready_to_play = true;
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}
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}
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void PaulstretchpluginAudioProcessor::releaseResources()
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{
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//m_control->stopplay();
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//m_ready_to_play = false;
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}
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#ifndef JucePlugin_PreferredChannelConfigurations
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bool PaulstretchpluginAudioProcessor::isBusesLayoutSupported (const BusesLayout& layouts) const
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{
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#if JucePlugin_IsMidiEffect
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ignoreUnused (layouts);
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return true;
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#else
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// This is the place where you check if the layout is supported.
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// In this template code we only support mono or stereo.
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if (layouts.getMainOutputChannelSet() != AudioChannelSet::mono()
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&& layouts.getMainOutputChannelSet() != AudioChannelSet::stereo())
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return false;
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// This checks if the input layout matches the output layout
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#if ! JucePlugin_IsSynth
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if (layouts.getMainOutputChannelSet() != layouts.getMainInputChannelSet())
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return false;
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#endif
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return true;
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#endif
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}
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#endif
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void copyAudioBufferWrappingPosition(const AudioBuffer<float>& src, AudioBuffer<float>& dest, int destbufpos, int maxdestpos)
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{
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for (int i = 0; i < dest.getNumChannels(); ++i)
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{
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int channel_to_copy = i % src.getNumChannels();
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if (destbufpos + src.getNumSamples() > maxdestpos)
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{
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int wrappos = (destbufpos + src.getNumSamples()) % maxdestpos;
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int partial_len = src.getNumSamples() - wrappos;
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dest.copyFrom(channel_to_copy, destbufpos, src, channel_to_copy, 0, partial_len);
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dest.copyFrom(channel_to_copy, partial_len, src, channel_to_copy, 0, wrappos);
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}
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else
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{
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dest.copyFrom(channel_to_copy, destbufpos, src, channel_to_copy, 0, src.getNumSamples());
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}
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}
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}
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void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
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{
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ScopedLock locker(m_cs);
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ScopedNoDenormals noDenormals;
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const int totalNumInputChannels = getTotalNumInputChannels();
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const int totalNumOutputChannels = getTotalNumOutputChannels();
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for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
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buffer.clear (i, 0, buffer.getNumSamples());
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if (m_ready_to_play == false)
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return;
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if (m_is_recording == true)
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{
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int recbuflenframes = m_max_reclen * getSampleRate();
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copyAudioBufferWrappingPosition(buffer, m_recbuffer, m_rec_pos, recbuflenframes);
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callGUI(this,[this, &buffer](PaulstretchpluginAudioProcessorEditor*ed)
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{
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ed->addAudioBlock(buffer, getSampleRate(), m_rec_pos);
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}, false);
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m_rec_pos = (m_rec_pos + buffer.getNumSamples()) % recbuflenframes;
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return;
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}
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jassert(m_buffering_source != nullptr);
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jassert(m_bufferingthread.isThreadRunning());
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m_stretch_source->setMainVolume(*getFloatParameter(0));
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m_stretch_source->setRate(*getFloatParameter(1));
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setFFTSize(*getFloatParameter(cpi_fftsize));
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m_ppar.pitch_shift.cents = *getFloatParameter(cpi_pitchshift) * 100.0;
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m_ppar.freq_shift.Hz = *getFloatParameter(cpi_frequencyshift);
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m_ppar.spread.enabled = *getFloatParameter(cpi_spreadamount) > 0.0f;
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m_ppar.spread.bandwidth = *getFloatParameter(cpi_spreadamount);
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m_ppar.compressor.power = *getFloatParameter(cpi_compress);
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m_ppar.harmonics.enabled = *getFloatParameter(cpi_numharmonics)>=1.0;
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m_ppar.harmonics.nharmonics = *getFloatParameter(cpi_numharmonics);
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m_ppar.harmonics.freq = *getFloatParameter(cpi_harmonicsfreq);
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m_ppar.octave.om2 = *getFloatParameter(cpi_octavesm2);
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m_ppar.octave.om1 = *getFloatParameter(cpi_octavesm1);
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m_ppar.octave.o0 = *getFloatParameter(cpi_octaves0);
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m_ppar.octave.o1 = *getFloatParameter(cpi_octaves1);
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m_ppar.octave.o15 = *getFloatParameter(cpi_octaves15);
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m_ppar.octave.o2 = *getFloatParameter(cpi_octaves2);
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m_ppar.octave.enabled = true;
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m_ppar.filter.low = *getFloatParameter(cpi_filter_low);
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m_ppar.filter.high = *getFloatParameter(cpi_filter_high);
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m_ppar.tonal_vs_noise.enabled = (*getFloatParameter(cpi_tonalvsnoisebw)) > 0.75;
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m_ppar.tonal_vs_noise.bandwidth = *getFloatParameter(cpi_tonalvsnoisebw);
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m_ppar.tonal_vs_noise.preserve = *getFloatParameter(cpi_tonalvsnoisepreserve);
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m_stretch_source->setOnsetDetection(*getFloatParameter(cpi_onsetdetection));
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m_stretch_source->setLoopXFadeLength(*getFloatParameter(cpi_loopxfadelen));
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double t0 = *getFloatParameter(cpi_soundstart);
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double t1 = *getFloatParameter(cpi_soundend);
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if (t0 > t1)
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std::swap(t0, t1);
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if (t1 - t0 < 0.001)
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t1 = t0 + 0.001;
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m_stretch_source->setPlayRange({ t0,t1 }, true);
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m_stretch_source->setFreezing(getParameter(cpi_freeze));
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m_stretch_source->setProcessParameters(&m_ppar);
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AudioSourceChannelInfo aif(buffer);
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m_buffering_source->getNextAudioBlock(aif);
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}
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//==============================================================================
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bool PaulstretchpluginAudioProcessor::hasEditor() const
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{
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return true; // (change this to false if you choose to not supply an editor)
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}
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AudioProcessorEditor* PaulstretchpluginAudioProcessor::createEditor()
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{
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return new PaulstretchpluginAudioProcessorEditor (*this);
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}
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//==============================================================================
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void PaulstretchpluginAudioProcessor::getStateInformation (MemoryBlock& destData)
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{
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ValueTree paramtree("paulstretch3pluginstate");
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for (int i=0;i<getNumParameters();++i)
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{
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auto par = getFloatParameter(i);
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if (par != nullptr)
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{
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paramtree.setProperty(par->paramID, (double)*par, nullptr);
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}
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}
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if (m_current_file != File())
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{
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paramtree.setProperty("importedfile", m_current_file.getFullPathName(), nullptr);
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}
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MemoryOutputStream stream(destData,true);
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paramtree.writeToStream(stream);
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}
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void PaulstretchpluginAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
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{
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ValueTree tree = ValueTree::readFromData(data, sizeInBytes);
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if (tree.isValid())
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{
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{
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ScopedLock locker(m_cs);
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for (int i = 0; i < getNumParameters(); ++i)
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{
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auto par = getFloatParameter(i);
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if (par != nullptr)
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{
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double parval = tree.getProperty(par->paramID, (double)*par);
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*par = parval;
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}
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}
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}
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String fn = tree.getProperty("importedfile");
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if (fn.isEmpty() == false)
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{
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File f(fn);
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setAudioFile(f);
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}
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}
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}
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void PaulstretchpluginAudioProcessor::setRecordingEnabled(bool b)
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{
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ScopedLock locker(m_cs);
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int lenbufframes = getSampleRate()*m_max_reclen;
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if (b == true)
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{
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m_using_memory_buffer = true;
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m_current_file = File();
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m_recbuffer.setSize(2, m_max_reclen*getSampleRate()+4096,false,false,true);
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m_recbuffer.clear();
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m_rec_pos = 0;
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callGUI(this,[this,lenbufframes](PaulstretchpluginAudioProcessorEditor* ed)
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{
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ed->beginAddingAudioBlocks(2, getSampleRate(), 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)
|
|
{
|
|
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_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(5),*getFloatParameter(6) };
|
|
}
|
|
|
|
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 == true && m_is_recording == false)
|
|
{
|
|
setRecordingEnabled(true);
|
|
return;
|
|
}
|
|
if (capture == false && m_is_recording == true)
|
|
{
|
|
setRecordingEnabled(false);
|
|
return;
|
|
}
|
|
|
|
}
|
|
}
|
|
|
|
void PaulstretchpluginAudioProcessor::finishRecording(int lenrecording)
|
|
{
|
|
m_is_recording = false;
|
|
m_stretch_source->setAudioBufferAsInputSource(&m_recbuffer, getSampleRate(), lenrecording);
|
|
m_stretch_source->setPlayRange({ *getFloatParameter(5),*getFloatParameter(6) }, 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();
|
|
}
|