Removed the phase freeze processing for now.

Source/PS_Source/Stretch.cpp

@@ -38,8 +38,7 @@ FFT::FFT(int nsamples_, bool no_inverse)
 		window.data[i]=0.707f;
 	window.type=W_RECTANGULAR;
 
-	phases.resize(nsamples);
+	data.resize(nsamples,true);
-    data.resize(nsamples,true);
 	bool allow_long_planning = false; // g_propsfile->getBoolValue("fftw_allow_long_planning", false);
     //double t0 = Time::getMillisecondCounterHiRes();
     if (allow_long_planning)
@@ -92,15 +91,9 @@ void FFT::smp2freq()
 void FFT::freq2smp()
 {
 	REALTYPE inv_2p15_2pi=1.0f/16384.0f*(float)c_PI;
-	if (m_phaserefreshcounter % m_phaserefreshrate == 0)
-	{
-		updatePhases();
-		//Logger::writeToLog("phases updated "+String(m_phaserefreshrate));
-	}
-	++m_phaserefreshcounter;
 	for (int i=1;i<nsamples/2;i++)
     {
-		unsigned int rand = phases[i]; // m_randdist(m_randgen);
+		unsigned int rand = m_randdist(m_randgen);
         REALTYPE phase=rand*inv_2p15_2pi;
         data[i]=freq[i]*cos(phase);
         data[nsamples-i]=freq[i]*sin(phase);
@@ -139,23 +132,6 @@ void FFT::applywindow(FFTWindow type)
 	for (int i=0;i<nsamples;i++) smp[i]*=window.data[i];
 }
 
-void FFT::setPhaseRefreshRate(int rate)
-{
-	m_phaserefreshrate = rate;
-}
-
-void FFT::updatePhases()
-{
-	for (int i = 1; i < nsamples / 2; i++)
-	{
-		phases[i] = m_randdist(m_randgen);
-	}
-}
-
-
-/*******************************************/
-
-
 Stretch::Stretch(REALTYPE rap_,int /*bufsize_*/,FFTWindow w,bool bypass_,REALTYPE samplerate_,int /*stereo_mode_*/)
 {
 	freezing=false;
@@ -381,12 +357,6 @@ void Stretch::here_is_onset(REALTYPE onset){
 	};
 };
 
-void Stretch::setPhaseRefreshRate(int rate)
-{
-	jassert(fft != nullptr);
-	fft->setPhaseRefreshRate(rate);
-}
-
 int Stretch::get_nsamples(REALTYPE current_pos_percents){
 	if (bypass) return bufsize;
 	if (freezing) return 0;

Source/PS_Source/Stretch.h

@@ -124,19 +124,17 @@ class FFT
 		void applywindow(FFTWindow type);
 		std::vector<REALTYPE> smp;//size of samples/2
 		std::vector<REALTYPE> freq;//size of samples
-		std::vector<unsigned int> phases;
+		
 		
 		int nsamples=0;
 
-		void setPhaseRefreshRate(int rate);
+		
 
 	private:
 
 		fftwf_plan planfftw,planifftw;
         FFTWBuffer<REALTYPE> data;
-		int m_phaserefreshcounter = 0;
+		
-		int m_phaserefreshrate = 1;
-		void updatePhases();
 		struct{
 			std::vector<REALTYPE> data;
 			FFTWindow type;
@@ -171,7 +169,7 @@ class Stretch
 			freezing=new_freezing;
 		};
 		bool isFreezing() { return freezing; }
-		void setPhaseRefreshRate(int rate);
+		
 		std::vector<REALTYPE> out_buf;//pot sa pun o variabila "max_out_bufsize" si asta sa fie marimea lui out_buf si pe out_bufsize sa il folosesc ca marime adaptiva
 
 		int get_nsamples(REALTYPE current_pos_percents);//how many samples are required 

Source/PS_Source/StretchSource.cpp

@@ -277,23 +277,6 @@ void StretchAudioSource::setSpectralOrderPreset(int id)
 	}
 }
 
-void StretchAudioSource::setPhaseRefreshRate(int rate)
-{
-	for (auto& e : m_stretchers)
-		e->setPhaseRefreshRate(rate);
-	return;
-	if (rate == m_phase_refresh_rate)
-		return;
-	if (m_cs.tryEnter())
-	{
-		for (auto& e : m_stretchers)
-			e->setPhaseRefreshRate(rate);
-		m_phase_refresh_rate = rate;
-		++m_param_change_count;
-		m_cs.exit();
-	}
-}
-
 void StretchAudioSource::getNextAudioBlock(const AudioSourceChannelInfo & bufferToFill)
 {
 	ScopedLock locker(m_cs);

Source/PS_Source/StretchSource.h

@@ -114,8 +114,7 @@ public:
 	int64_t getLastSourcePosition() const { return m_last_filepos; }
 	int m_prebuffersize = 0;
 	void setSpectralOrderPreset(int id);
-	int getPhaseRefreshRate() const { return m_phase_refresh_rate; }
+	
-	void setPhaseRefreshRate(int rate);
 private:
 	CircularBuffer<float> m_stretchoutringbuf{ 1024 * 1024 };
 	AudioBuffer<float> m_file_inbuf;
@@ -144,7 +143,7 @@ private:
 	int m_pause_state = 0;
 	Range<double> m_playrange{ 0.0,1.0 };
 	int64_t m_rand_count = 0;
-	int m_phase_refresh_rate = 1;
+	
 	bool m_stream_end_reached = false;
 	int64_t m_output_silence_counter = 0;
 	File m_curfile;

Source/PluginProcessor.cpp

@@ -184,7 +184,7 @@ PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
 	addParameter(new AudioParameterFloat("dryplayrate0", "Dry playrate",
 		NormalisableRange<float>(0.1f, 8.0f,
 			dprate_convertFrom0To1Func, dprate_convertTo0To1Func), 1.0f)); // 62
-	addParameter(new AudioParameterInt("phaserefreshrate0", "Phase randomization rate", 1, 32, 32)); // 63
+	
 	auto& pars = getParameters();
 	for (const auto& p : pars)
 		m_reset_pars.push_back(p->getValue());
@@ -745,7 +745,7 @@ void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, M
 	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->setPhaseRefreshRate(*getIntParameter(cpi_phase_refresh_rate));
+	
 
 	//m_stretch_source->setSpectralModulesEnabled(m_sm_enab_pars);
 

Source/PluginProcessor.h

@@ -88,7 +88,6 @@ const int cpi_octaves_ratio7 = 59;
 const int cpi_looping_enabled = 60;
 const int cpi_rewind = 61;
 const int cpi_dryplayrate = 62;
-const int cpi_phase_refresh_rate = 63;
 
 class MyThreadPool : public ThreadPool
 {

readme.txt

@@ -7,13 +7,12 @@ Copyright (C) 2017-2018 Xenakios
 Released under GNU General Public License v.3 license.
 
 History :
-08-01-2019 1.3.0
+01-17-2019 1.3.0
 	-Captured audio buffers can optionally be saved as files on disk for later recall.
 	-Added varispeed (resampling speed change) feature when spectral stretch engine is bypassed
 	-Added shortcut key (may not work properly in all plugin formats and hosts) :
 		"I" to open file import dialog
 	-Attempt to prevent capture enabled state from being recalled when undoing in the host
-	-Added VST3 version
 07-09-2018 1.2.2
 	-Add option to mute audio when capturing audio
 	-Automatically adjust play range after capturing to captured length