Merge branch 'master' of https://bitbucket.org/xenakios/paulstretchplugin

juuh

Source/PS_Source/Input/AInputS.h

@@ -74,7 +74,9 @@ public:
 			std::lock_guard<std::mutex> locker(m_mutex);
             m_using_memory_buffer = false;
 			m_afreader = std::unique_ptr<AudioFormatReader>(reader);
-            m_currentsample = 0;
+			if (m_activerange.isEmpty())
+				m_activerange = { 0.0,1.0 };
+			m_currentsample = m_activerange.getStart()*info.nsamples;
             info.samplerate = (int)m_afreader->sampleRate;
             info.nchannels = m_afreader->numChannels;
             info.nsamples = m_afreader->lengthInSamples;
@@ -83,6 +85,7 @@ public:
 				m_readbuf.setSize(info.nchannels, m_readbuf.getNumSamples());
 				m_crossfadebuf.setSize(info.nchannels, m_crossfadebuf.getNumSamples());
 			}
+			updateXFadeCache();
 			m_readbuf.clear();
             return true;
         }

Source/PS_Source/ProcessedStretch.cpp

@@ -104,9 +104,9 @@ void ProcessedStretch::process_spectrum(REALTYPE *freq)
 {
 	for (auto& e : m_spectrum_processes)
     {
-		copy(freq, infreq.data());
+		spectrum_copy(nfreq, freq, infreq.data());
 		if (e == 0 && pars.harmonics.enabled)
-			do_harmonics(infreq.data(), freq);
+			spectrum_do_harmonics(pars, tmpfreq1, nfreq, samplerate, infreq.data(), freq);
 		if (e == 1 && pars.tonal_vs_noise.enabled)
 			do_tonal_vs_noise(infreq.data(), freq);
 		if (e == 2 && pars.freq_shift.enabled)
@@ -174,15 +174,6 @@ void ProcessedStretch::process_spectrum(REALTYPE *freq)
 //void ProcessedStretch::process_output(REALTYPE *smps,int nsmps){
 //};
 
-
-REALTYPE profile(REALTYPE fi, REALTYPE bwi){
-	REALTYPE x=fi/bwi;
-	x*=x;
-	if (x>14.71280603) return 0.0;
-	return exp(-x);///bwi;
-
-};
-
 void ProcessedStretch::do_harmonics(REALTYPE *freq1,REALTYPE *freq2){
 	REALTYPE freq=pars.harmonics.freq;
 	REALTYPE bandwidth=pars.harmonics.bandwidth;

Source/PS_Source/ProcessedStretch.h

@@ -180,6 +180,263 @@ struct ProcessParameters
 	}
 };
 
+inline REALTYPE profile(REALTYPE fi, REALTYPE bwi) {
+	REALTYPE x = fi / bwi;
+	x *= x;
+	if (x>14.71280603) return 0.0;
+	return exp(-x);///bwi;
+
+};
+
+inline void spectrum_copy(int nfreq, REALTYPE* freq1, REALTYPE* freq2)
+{
+	for (int i = 0; i<nfreq; i++) freq2[i] = freq1[i];
+};
+
+
+inline void spectrum_spread(int nfreq, double samplerate, 
+	std::vector<REALTYPE>& tmpfreq1,
+	REALTYPE *freq1, REALTYPE *freq2, REALTYPE spread_bandwidth) {
+	//convert to log spectrum
+	REALTYPE minfreq = 20.0f;
+	REALTYPE maxfreq = 0.5f*samplerate;
+
+	REALTYPE log_minfreq = log(minfreq);
+	REALTYPE log_maxfreq = log(maxfreq);
+
+	for (int i = 0; i<nfreq; i++) {
+		REALTYPE freqx = i / (REALTYPE)nfreq;
+		REALTYPE x = exp(log_minfreq + freqx * (log_maxfreq - log_minfreq)) / maxfreq * nfreq;
+		REALTYPE y = 0.0f;
+		int x0 = (int)floor(x); if (x0 >= nfreq) x0 = nfreq - 1;
+		int x1 = x0 + 1; if (x1 >= nfreq) x1 = nfreq - 1;
+		REALTYPE xp = x - x0;
+		if (x<nfreq) {
+			y = freq1[x0] * (1.0f - xp) + freq1[x1] * xp;
+		};
+		tmpfreq1[i] = y;
+	};
+
+	//increase the bandwidth of each harmonic (by smoothing the log spectrum)
+	int n = 2;
+	REALTYPE bandwidth = spread_bandwidth;
+	REALTYPE a = 1.0f - pow(2.0f, -bandwidth * bandwidth*10.0f);
+	a = pow(a, 8192.0f / nfreq * n);
+
+	for (int k = 0; k<n; k++) {
+		tmpfreq1[0] = 0.0f;
+		for (int i = 1; i<nfreq; i++) {
+			tmpfreq1[i] = tmpfreq1[i - 1] * a + tmpfreq1[i] * (1.0f - a);
+		};
+		tmpfreq1[nfreq - 1] = 0.0f;
+		for (int i = nfreq - 2; i>0; i--) {
+			tmpfreq1[i] = tmpfreq1[i + 1] * a + tmpfreq1[i] * (1.0f - a);
+		};
+	};
+
+	freq2[0] = 0;
+	REALTYPE log_maxfreq_d_minfreq = log(maxfreq / minfreq);
+	for (int i = 1; i<nfreq; i++) {
+		REALTYPE freqx = i / (REALTYPE)nfreq;
+		REALTYPE x = log((freqx*maxfreq) / minfreq) / log_maxfreq_d_minfreq * nfreq;
+		REALTYPE y = 0.0;
+		if ((x>0.0) && (x<nfreq)) {
+			int x0 = (int)floor(x); if (x0 >= nfreq) x0 = nfreq - 1;
+			int x1 = x0 + 1; if (x1 >= nfreq) x1 = nfreq - 1;
+			REALTYPE xp = x - x0;
+			y = tmpfreq1[x0] * (1.0f - xp) + tmpfreq1[x1] * xp;
+		};
+		freq2[i] = y;
+	};
+
+
+};
+
+
+inline void spectrum_do_compressor(const ProcessParameters& pars, int nfreq, REALTYPE *freq1, REALTYPE *freq2) {
+	REALTYPE rms = 0.0;
+	for (int i = 0; i<nfreq; i++) rms += freq1[i] * freq1[i];
+	rms = sqrt(rms / nfreq)*0.1f;
+	if (rms<1e-3f) rms = 1e-3f;
+
+	REALTYPE _rap = pow(rms, -pars.compressor.power);
+	for (int i = 0; i<nfreq; i++) freq2[i] = freq1[i] * _rap;
+};
+
+inline void spectrum_do_tonal_vs_noise(const ProcessParameters& pars, int nfreq, double samplerate,
+	std::vector<REALTYPE>& tmpfreq1,
+	REALTYPE *freq1, REALTYPE *freq2) {
+	spectrum_spread(nfreq, samplerate, tmpfreq1, freq1, tmpfreq1.data(), pars.tonal_vs_noise.bandwidth);
+
+	if (pars.tonal_vs_noise.preserve >= 0.0) {
+		REALTYPE mul = (pow(10.0f, pars.tonal_vs_noise.preserve) - 1.0f);
+		for (int i = 0; i<nfreq; i++) {
+			REALTYPE x = freq1[i];
+			REALTYPE smooth_x = tmpfreq1[i] + 1e-6f;
+
+			REALTYPE result = 0.0f;
+			result = x - smooth_x * mul;
+			if (result<0.0f) result = 0.0f;
+			freq2[i] = result;
+		};
+	}
+	else {
+		REALTYPE mul = (pow(5.0f, 1.0f + pars.tonal_vs_noise.preserve) - 1.0f);
+		for (int i = 0; i<nfreq; i++) {
+			REALTYPE x = freq1[i];
+			REALTYPE smooth_x = tmpfreq1[i] + 1e-6f;
+
+			REALTYPE result = 0.0f;
+			result = x - smooth_x * mul + 0.1f*mul;
+			if (result<0.0f) result = x;
+			else result = 0.0f;
+
+			freq2[i] = result;
+		};
+	};
+
+};
+
+inline void spectrum_do_harmonics(const ProcessParameters& pars, std::vector<REALTYPE>& tmpfreq1, int nfreq, double samplerate, REALTYPE *freq1, REALTYPE *freq2) {
+	REALTYPE freq = pars.harmonics.freq;
+	REALTYPE bandwidth = pars.harmonics.bandwidth;
+	int nharmonics = pars.harmonics.nharmonics;
+
+	if (freq<10.0) freq = 10.0;
+
+	REALTYPE *amp = tmpfreq1.data();
+	for (int i = 0; i<nfreq; i++) amp[i] = 0.0;
+
+	for (int nh = 1; nh <= nharmonics; nh++) {//for each harmonic
+		REALTYPE bw_Hz;//bandwidth of the current harmonic measured in Hz
+		REALTYPE bwi;
+		REALTYPE fi;
+		REALTYPE f = nh * freq;
+
+		if (f >= samplerate / 2) break;
+
+		bw_Hz = (pow(2.0f, bandwidth / 1200.0f) - 1.0f)*f;
+		bwi = bw_Hz / (2.0f*samplerate);
+		fi = f / samplerate;
+
+		REALTYPE sum = 0.0f;
+		REALTYPE max = 0.0f;
+		for (int i = 1; i<nfreq; i++) {//todo: optimize here
+			REALTYPE hprofile;
+			hprofile = profile((i / (REALTYPE)nfreq*0.5f) - fi, bwi);
+			amp[i] += hprofile;
+			if (max<hprofile) max = hprofile;
+			sum += hprofile;
+		};
+	};
+
+	REALTYPE max = 0.0;
+	for (int i = 1; i<nfreq; i++) {
+		if (amp[i]>max) max = amp[i];
+	};
+	if (max<1e-8f) max = 1e-8f;
+
+	for (int i = 1; i<nfreq; i++) {
+		//REALTYPE c,s;
+		REALTYPE a = amp[i] / max;
+		if (!pars.harmonics.gauss) a = (a<0.368f ? 0.0f : 1.0f);
+		freq2[i] = freq1[i] * a;
+	};
+
+};
+
+inline void spectrum_add(int nfreq, REALTYPE *freq2, REALTYPE *freq1, REALTYPE a) {
+	for (int i = 0; i<nfreq; i++) freq2[i] += freq1[i] * a;
+};
+
+inline void spectrum_zero(int nfreq,REALTYPE *freq1) {
+	for (int i = 0; i<nfreq; i++) freq1[i] = 0.0;
+};
+
+inline void spectrum_do_freq_shift(const ProcessParameters& pars, int nfreq, double samplerate, REALTYPE *freq1, REALTYPE *freq2) {
+	spectrum_zero(nfreq, freq2);
+	int ifreq = (int)(pars.freq_shift.Hz / (samplerate*0.5)*nfreq);
+	for (int i = 0; i<nfreq; i++) {
+		int i2 = ifreq + i;
+		if ((i2>0) && (i2<nfreq)) freq2[i2] = freq1[i];
+	};
+};
+
+inline void spectrum_do_pitch_shift(const ProcessParameters& pars, int nfreq, REALTYPE *freq1, REALTYPE *freq2, REALTYPE _rap) {
+	spectrum_zero(nfreq,freq2);
+	if (_rap<1.0) {//down
+		for (int i = 0; i<nfreq; i++) {
+			int i2 = (int)(i*_rap);
+			if (i2 >= nfreq) break;
+			freq2[i2] += freq1[i];
+		};
+	};
+	if (_rap >= 1.0) {//up
+		_rap = 1.0f / _rap;
+		for (int i = 0; i<nfreq; i++) {
+			freq2[i] = freq1[(int)(i*_rap)];
+		};
+
+	};
+};
+
+inline void spectrum_do_octave(const ProcessParameters& pars, int nfreq, double samplerate, 
+	std::vector<REALTYPE>& sumfreq, 
+	std::vector<REALTYPE>& tmpfreq1,
+	REALTYPE *freq1, REALTYPE *freq2) {
+	spectrum_zero(nfreq,sumfreq.data());
+	if (pars.octave.om2>1e-3) {
+		spectrum_do_pitch_shift(pars,nfreq, freq1, tmpfreq1.data(), 0.25);
+		spectrum_add(nfreq, sumfreq.data(), tmpfreq1.data(), pars.octave.om2);
+	};
+	if (pars.octave.om1>1e-3) {
+		spectrum_do_pitch_shift(pars,nfreq, freq1, tmpfreq1.data(), 0.5);
+		spectrum_add(nfreq,sumfreq.data(), tmpfreq1.data(), pars.octave.om1);
+	};
+	if (pars.octave.o0>1e-3) {
+		spectrum_add(nfreq,sumfreq.data(), freq1, pars.octave.o0);
+	};
+	if (pars.octave.o1>1e-3) {
+		spectrum_do_pitch_shift(pars,nfreq, freq1, tmpfreq1.data(), 2.0);
+		spectrum_add(nfreq,sumfreq.data(), tmpfreq1.data(), pars.octave.o1);
+	};
+	if (pars.octave.o15>1e-3) {
+		spectrum_do_pitch_shift(pars,nfreq, freq1, tmpfreq1.data(), 3.0);
+		spectrum_add(nfreq,sumfreq.data(), tmpfreq1.data(), pars.octave.o15);
+	};
+	if (pars.octave.o2>1e-3) {
+		spectrum_do_pitch_shift(pars, nfreq, freq1, tmpfreq1.data(), 4.0);
+		spectrum_add(nfreq,sumfreq.data(), tmpfreq1.data(), pars.octave.o2);
+	};
+
+	REALTYPE sum = 0.01f + pars.octave.om2 + pars.octave.om1 + pars.octave.o0 + pars.octave.o1 + pars.octave.o15 + pars.octave.o2;
+	if (sum<0.5f) sum = 0.5f;
+	for (int i = 0; i<nfreq; i++) freq2[i] = sumfreq[i] / sum;
+};
+
+inline void spectrum_do_filter(const ProcessParameters& pars, int nfreq, double samplerate, REALTYPE *freq1, REALTYPE *freq2) {
+	REALTYPE low = 0, high = 0;
+	if (pars.filter.low<pars.filter.high) {//sort the low/high freqs
+		low = pars.filter.low;
+		high = pars.filter.high;
+	}
+	else {
+		high = pars.filter.low;
+		low = pars.filter.high;
+	};
+	int ilow = (int)(low / samplerate * nfreq*2.0f);
+	int ihigh = (int)(high / samplerate * nfreq*2.0f);
+	REALTYPE dmp = 1.0;
+	REALTYPE dmprap = 1.0f - pow(pars.filter.hdamp*0.5f, 4.0f);
+	for (int i = 0; i<nfreq; i++) {
+		REALTYPE a = 0.0f;
+		if ((i >= ilow) && (i<ihigh)) a = 1.0f;
+		if (pars.filter.stop) a = 1.0f - a;
+		freq2[i] = freq1[i] * a*dmp;
+		dmp *= dmprap + 1e-8f;
+	};
+};
+
 class SpectrumProcess
 {
 public:

Source/PS_Source/Stretch.cpp

@@ -20,7 +20,7 @@
 #include <stdlib.h>
 #include <math.h>
 
-FFT::FFT(int nsamples_)
+FFT::FFT(int nsamples_, bool no_inverse)
 {
     nsamples=nsamples_;
 	if (nsamples%2!=0) {
@@ -43,13 +43,15 @@ FFT::FFT(int nsamples_)
     {
         //fftwf_plan_with_nthreads(2);
         planfftw=fftwf_plan_r2r_1d(nsamples,data.data(),data.data(),FFTW_R2HC,FFTW_MEASURE);
-        planifftw=fftwf_plan_r2r_1d(nsamples,data.data(),data.data(),FFTW_HC2R,FFTW_MEASURE);
+        if (no_inverse == false)
+			planifftw=fftwf_plan_r2r_1d(nsamples,data.data(),data.data(),FFTW_HC2R,FFTW_MEASURE);
     } else
     {
         //fftwf_plan_with_nthreads(2);
         planfftw=fftwf_plan_r2r_1d(nsamples,data.data(),data.data(),FFTW_R2HC,FFTW_ESTIMATE);
         //fftwf_plan_with_nthreads(2);
-        planifftw=fftwf_plan_r2r_1d(nsamples,data.data(),data.data(),FFTW_HC2R,FFTW_ESTIMATE);
+        if (no_inverse == false)
+			planifftw=fftwf_plan_r2r_1d(nsamples,data.data(),data.data(),FFTW_HC2R,FFTW_ESTIMATE);
     }
     //double t1 = Time::getMillisecondCounterHiRes();
     //Logger::writeToLog("Creating FFTW3 plans took "+String(t1-t0)+ "ms");
@@ -64,7 +66,8 @@ FFT::FFT(int nsamples_)
 FFT::~FFT()
 {
 	fftwf_destroy_plan(planfftw);
-	fftwf_destroy_plan(planifftw);
+	if (planifftw!=nullptr)
+		fftwf_destroy_plan(planifftw);
 };
 
 void FFT::smp2freq()

Source/PS_Source/Stretch.h

@@ -122,7 +122,7 @@ enum FFTWindow{W_RECTANGULAR,W_HAMMING,W_HANN,W_BLACKMAN,W_BLACKMAN_HARRIS};
 class FFT
 {//FFT class that considers phases as random
 	public:
-		FFT(int nsamples_);//samples must be even
+		FFT(int nsamples_, bool no_inverse=false);//samples must be even
 		~FFT();
 		void smp2freq();//input is smp, output is freq (phases are discarded)
 		void freq2smp();//input is freq,output is smp (phases are random)
@@ -192,7 +192,7 @@ class Stretch
 
 		virtual void process_spectrum(REALTYPE *){};
 		virtual REALTYPE get_stretch_multiplier(REALTYPE pos_percents);
-		REALTYPE samplerate;
+		REALTYPE samplerate=0.0f;
 		
 	private:
 

Source/PS_Source/StretchSource.cpp

@@ -129,6 +129,24 @@ void StretchAudioSource::setAudioBufferAsInputSource(AudioBuffer<float>* buf, in
 		setPlayRange({ 0.0,1.0 }, true);
 }
 
+void StretchAudioSource::setMainVolume(double decibels)
+{
+	if (decibels == m_main_volume)
+		return;
+	std::lock_guard <decltype(m_mutex)> locker(m_mutex);
+	m_main_volume = jlimit(-144.0, 12.0, decibels);
+	++m_param_change_count;
+}
+
+void StretchAudioSource::setLoopXFadeLength(double lenseconds)
+{
+	if (lenseconds == m_loopxfadelen)
+		return;
+	std::lock_guard <decltype(m_mutex)> locker(m_mutex);
+	m_loopxfadelen = jlimit(0.0, 1.0, lenseconds);
+	++m_param_change_count;
+}
+
 void StretchAudioSource::getNextAudioBlock(const AudioSourceChannelInfo & bufferToFill)
 {
 	// for realtime play, this is assumed to be used with BufferingAudioSource, so mutex locking should not be too bad...
@@ -145,7 +163,7 @@ void StretchAudioSource::getNextAudioBlock(const AudioSourceChannelInfo & buffer
 			e->set_freezing(m_freezing);
 	}
     
-	double maingain = Decibels::decibelsToGain((double)val_MainVolume.getValue());
+	double maingain = Decibels::decibelsToGain(m_main_volume);
 	if (m_vol_smoother.getTargetValue() != maingain)
 		m_vol_smoother.setValue(maingain);
 	FloatVectorOperations::disableDenormalisedNumberSupport();
@@ -158,7 +176,7 @@ void StretchAudioSource::getNextAudioBlock(const AudioSourceChannelInfo & buffer
 		return;
 	if (m_inputfile->info.nsamples == 0)
 		return;
-	m_inputfile->setXFadeLenSeconds(val_XFadeLen.getValue());
+	m_inputfile->setXFadeLenSeconds(m_loopxfadelen);
     
 	double silencethreshold = Decibels::decibelsToGain(-70.0);
 	bool tempfirst = true;
@@ -458,7 +476,7 @@ void StretchAudioSource::setProcessParameters(ProcessParameters * pars)
 	++m_param_change_count;
 }
 
-ProcessParameters StretchAudioSource::getProcessParameters()
+const ProcessParameters& StretchAudioSource::getProcessParameters()
 {
 	return m_ppar;
 }
@@ -545,6 +563,7 @@ void StretchAudioSource::setPlayRange(Range<double> playrange, bool isloop)
 		m_inputfile->seek(m_playrange.getStart());
 	m_seekpos = m_playrange.getStart();
 	++m_param_change_count;
+
 }
 
 bool StretchAudioSource::isLoopEnabled()
@@ -633,8 +652,8 @@ void MultiStretchAudioSource::getNextAudioBlock(const AudioSourceChannelInfo & b
 	m_blocksize = bufferToFill.numSamples;
 	if (m_is_in_switch == false)
 	{
-		getActiveStretchSource()->val_MainVolume.setValue(val_MainVolume.getValue());
-        getActiveStretchSource()->val_XFadeLen.setValue(val_XFadeLen.getValue());
+		getActiveStretchSource()->setMainVolume(val_MainVolume.getValue());
+        getActiveStretchSource()->setLoopXFadeLength(val_XFadeLen.getValue());
 		getActiveStretchSource()->setFreezing(m_freezing);
         getActiveStretchSource()->getNextAudioBlock(bufferToFill);
         
@@ -648,10 +667,10 @@ void MultiStretchAudioSource::getNextAudioBlock(const AudioSourceChannelInfo & b
 		}
 		AudioSourceChannelInfo ascinfo1(m_processbuffers[0]);
 		AudioSourceChannelInfo ascinfo2(m_processbuffers[1]);
-		m_stretchsources[0]->val_MainVolume.setValue(val_MainVolume.getValue());
-		m_stretchsources[1]->val_MainVolume.setValue(val_MainVolume.getValue());
-        m_stretchsources[0]->val_XFadeLen.setValue(val_XFadeLen.getValue());
-        m_stretchsources[1]->val_XFadeLen.setValue(val_XFadeLen.getValue());
+		m_stretchsources[0]->setMainVolume(val_MainVolume.getValue());
+		m_stretchsources[1]->setMainVolume(val_MainVolume.getValue());
+        m_stretchsources[0]->setLoopXFadeLength(val_XFadeLen.getValue());
+        m_stretchsources[1]->setLoopXFadeLength(val_XFadeLen.getValue());
 		m_stretchsources[0]->setFreezing(m_freezing);
 		m_stretchsources[1]->setFreezing(m_freezing);
 		m_stretchsources[1]->setFFTWindowingType(m_stretchsources[0]->getFFTWindowingType());

Source/PS_Source/StretchSource.h

@@ -59,7 +59,7 @@ public:
 	void setRate(double rate);
 	double getRate() { return m_playrate; }
 	void setProcessParameters(ProcessParameters* pars);
-	ProcessParameters getProcessParameters();
+	const ProcessParameters& getProcessParameters();
 	void setFFTSize(int size);
 	int getFFTSize() { return m_process_fftsize; }
 	
@@ -82,8 +82,7 @@ public:
 	void setFFTWindowingType(int windowtype);
     int getFFTWindowingType() { return m_fft_window_type; }
     std::pair<Range<double>,Range<double>> getFileCachedRangesNormalized();
-	Value val_MainVolume;
-    Value val_XFadeLen;
+	
 	ValueTree getStateTree();
 	void setStateTree(ValueTree state);
 	void setClippingEnabled(bool b) { m_clip_output = b; }
@@ -91,6 +90,10 @@ public:
 	void setLoopingEnabled(bool b);
 	void setMaxLoops(int64_t numloops) { m_maxloops = numloops; }
 	void setAudioBufferAsInputSource(AudioBuffer<float>* buf, int sr, int len);
+	void setMainVolume(double decibels);
+	double getMainVolume() const { return m_main_volume; }
+	void setLoopXFadeLength(double lenseconds);
+	double getLoopXFadeLengtj() const { return m_loopxfadelen; }
 	int m_param_change_count = 0;
 private:
 	CircularBuffer<float> m_stretchoutringbuf{ 1024 * 1024 };
@@ -107,6 +110,8 @@ private:
 	double m_outsr = 44100.0;
 	int m_process_fftsize = 0;
     int m_fft_window_type = -1;
+	double m_main_volume = 0.0;
+	double m_loopxfadelen = 0.0;
 	ProcessParameters m_ppar;
 	BinauralBeatsParameters	m_bbpar;
 	double m_playrate = 1.0;

Source/PluginEditor.cpp

@@ -42,7 +42,7 @@ PaulstretchpluginAudioProcessorEditor::PaulstretchpluginAudioProcessorEditor (Pa
 	addAndMakeVisible(&m_rec_enable);
 	m_rec_enable.setButtonText("Capture");
 	attachCallback(m_rec_enable, [this]() { processor.setRecordingEnabled(m_rec_enable.getToggleState()); });
-
+	addAndMakeVisible(&m_specvis);
 	setSize (700, 30+pars.size()*25+200);
 	m_wavecomponent.TimeSelectionChangedCallback = [this](Range<double> range, int which)
 	{
@@ -56,6 +56,7 @@ PaulstretchpluginAudioProcessorEditor::PaulstretchpluginAudioProcessorEditor (Pa
 	m_wavecomponent.ShowFileCacheRange = true;
 	startTimer(1, 100);
 	startTimer(2, 1000);
+	startTimer(3, 200);
 	m_wavecomponent.startTimer(100);
 	
 }
@@ -79,10 +80,11 @@ void PaulstretchpluginAudioProcessorEditor::resized()
 	
 	for (int i = 0; i < m_parcomps.size(); ++i)
 	{
-		m_parcomps[i]->setBounds(1, 30 + i * 25, 598, 24);
+		m_parcomps[i]->setBounds(1, 30 + i * 25, getWidth()-2, 24);
 	}
 	int yoffs = m_parcomps.back()->getBottom() + 1;
-	m_wavecomponent.setBounds(1, yoffs, getWidth()-2, getHeight()-1-yoffs);
+	//m_wavecomponent.setBounds(1, yoffs, getWidth()-2, getHeight()-1-yoffs);
+	m_specvis.setBounds(1, yoffs, getWidth() - 2, getHeight() - 1 - yoffs);
 }
 
 void PaulstretchpluginAudioProcessorEditor::timerCallback(int id)
@@ -108,6 +110,12 @@ void PaulstretchpluginAudioProcessorEditor::timerCallback(int id)
 			m_wavecomponent.setAudioFile(processor.getAudioFile());
 		}
 		m_wavecomponent.setTimeSelection(processor.getTimeSelection());
+		
+	}
+	if (id == 3)
+	{
+		m_specvis.setState(processor.getStretchSource()->getProcessParameters(), processor.getStretchSource()->getFFTSize() / 2,
+			processor.getSampleRate());
 	}
 }
 
@@ -431,3 +439,77 @@ int WaveformComponent::getTimeSelectionEdge(int x, int y)
 	return 0;
 }
 
+SpectralVisualizer::SpectralVisualizer()
+{
+	m_img = Image(Image::RGB, 500, 200, true);
+}
+
+void SpectralVisualizer::setState(const ProcessParameters & pars, int nfreqs, double samplerate)
+{
+	double t0 = Time::getMillisecondCounterHiRes();
+	double hz = 440.0;
+	int numharmonics = 40;
+	double scaler = 1.0 / numharmonics;
+	if (m_img.getWidth()!=getWidth() || m_img.getHeight()!=getHeight())
+		m_img = Image(Image::RGB, getWidth(), getHeight(), true);
+	if (m_nfreqs == 0 || nfreqs != m_nfreqs)
+	{
+		m_nfreqs = nfreqs;
+		m_insamples = std::vector<REALTYPE>(nfreqs * 2);
+		m_freqs1 = std::vector<REALTYPE>(nfreqs);
+		m_freqs2 = std::vector<REALTYPE>(nfreqs);
+		m_freqs3 = std::vector<REALTYPE>(nfreqs);
+		m_fft = std::make_unique<FFT>(nfreqs*2);
+		std::fill(m_insamples.begin(), m_insamples.end(), 0.0f);
+		for (int i = 0; i < nfreqs; ++i)
+		{
+			for (int j = 0; j < numharmonics; ++j)
+			{
+				double oscgain = 1.0 - (1.0 / numharmonics)*j;
+				m_insamples[i] += scaler * oscgain * sin(2 * 3.141592653 / samplerate * i* (hz + hz * j));
+			}
+		}
+	}
+	
+	//std::fill(m_freqs1.begin(), m_freqs1.end(), 0.0f);
+	//std::fill(m_freqs2.begin(), m_freqs2.end(), 0.0f);
+	//std::fill(m_freqs3.begin(), m_freqs3.end(), 0.0f);
+	//std::fill(m_fft->freq.begin(), m_fft->freq.end(), 0.0f);
+	
+	
+	for (int i = 0; i < nfreqs; ++i)
+	{
+		m_fft->smp[i] = m_insamples[i];
+	}
+	m_fft->applywindow(W_HAMMING);
+	m_fft->smp2freq();
+	double ratio = pow(2.0f, pars.pitch_shift.cents / 1200.0f);
+	spectrum_do_pitch_shift(pars, nfreqs, m_fft->freq.data(), m_freqs2.data(), ratio);
+	spectrum_do_freq_shift(pars, nfreqs, samplerate, m_freqs2.data(), m_freqs1.data());
+	spectrum_do_compressor(pars, nfreqs, m_freqs1.data(), m_freqs2.data());
+	spectrum_spread(nfreqs, samplerate, m_freqs3, m_freqs2.data(), m_freqs1.data(), pars.spread.bandwidth);
+	if (pars.harmonics.enabled)
+		spectrum_do_harmonics(pars, m_freqs3, nfreqs, samplerate, m_freqs1.data(), m_freqs2.data());
+	else spectrum_copy(nfreqs, m_freqs1.data(), m_freqs2.data());
+	Graphics g(m_img);
+	g.fillAll(Colours::black);
+	g.setColour(Colours::white);
+	for (int i = 0; i < nfreqs; ++i)
+	{
+		double binfreq = (samplerate / 2 / nfreqs)*i;
+		double xcor = jmap<double>(binfreq, 0.0, samplerate / 2.0, 0.0, getWidth());
+		double ycor = getHeight()- jmap<double>(m_freqs2[i], 0.0, nfreqs/128, 0.0, getHeight());
+		ycor = jlimit<double>(0.0, getHeight(), ycor);
+		g.drawLine(xcor, getHeight(), xcor, ycor, 1.0);
+	}
+	double t1 = Time::getMillisecondCounterHiRes();
+	m_elapsed = t1 - t0;
+	repaint();
+}
+
+void SpectralVisualizer::paint(Graphics & g)
+{
+	g.drawImage(m_img, 0, 0, getWidth(), getHeight(), 0, 0, m_img.getWidth(), m_img.getHeight());
+	g.setColour(Colours::yellow);
+	g.drawText(String(m_elapsed, 1)+" ms", 1, 1, getWidth(), 30, Justification::topLeft);
+}

Source/PluginEditor.h

@@ -15,6 +15,21 @@
 #include <memory>
 #include <vector>
 
+class SpectralVisualizer : public Component
+{
+public:
+	SpectralVisualizer();
+	void setState(const ProcessParameters& pars, int nfreqs, double samplerate);
+	void paint(Graphics& g) override;
+	
+private:
+	Image m_img;
+	std::vector<REALTYPE> m_insamples,m_freqs1, m_freqs2, m_freqs3;
+	std::unique_ptr<FFT> m_fft;
+	int m_nfreqs = 0;
+	double m_elapsed = 0.0;
+};
+
 inline void attachCallback(Button& button, std::function<void()> callback)
 {
 	struct ButtonCallback : public Button::Listener,
@@ -240,6 +255,7 @@ public:
 private:
     PaulstretchpluginAudioProcessor& processor;
 	std::vector<std::shared_ptr<ParameterComponent>> m_parcomps;
+	SpectralVisualizer m_specvis;
 	ToggleButton m_rec_enable;
 	TextButton m_import_button;
 	Label m_info_label;

Source/PluginProcessor.cpp

@@ -12,8 +12,10 @@
 #include "PluginEditor.h"
 #include <set>
 
+#ifdef WIN32
 #undef min
 #undef max
+#endif
 
 std::set<PaulstretchpluginAudioProcessor*> g_activeprocessors;
 
@@ -82,6 +84,7 @@ PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
 	m_ppar.freq_shift.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, 128.0f, 0.01f, 0.5),1.0f)); // 1
@@ -92,6 +95,10 @@ PaulstretchpluginAudioProcessor::PaulstretchpluginAudioProcessor()
 	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", 1.0f, 5000.0f, 100.0f)); // 12
 }
 
 PaulstretchpluginAudioProcessor::~PaulstretchpluginAudioProcessor()
@@ -145,7 +152,7 @@ bool PaulstretchpluginAudioProcessor::isMidiEffect() const
 
 double PaulstretchpluginAudioProcessor::getTailLengthSeconds() const
 {
-    return 0.0;
+    return (double)m_bufamounts[m_prebuffer_amount]/getSampleRate();
 }
 
 int PaulstretchpluginAudioProcessor::getNumPrograms()
@@ -306,14 +313,19 @@ void PaulstretchpluginAudioProcessor::processBlock (AudioSampleBuffer& buffer, M
 	}
 	jassert(m_buffering_source != nullptr);
 	jassert(m_bufferingthread.isThreadRunning());
-	m_stretch_source->val_MainVolume = (float)*getFloatParameter(0);
+	m_stretch_source->setMainVolume(*getFloatParameter(0));
 	m_stretch_source->setRate(*getFloatParameter(1));
-	m_stretch_source->val_XFadeLen = 0.1;
+
 	setFFTSize(*getFloatParameter(2));
 	m_ppar.pitch_shift.cents = *getFloatParameter(3) * 100.0;
 	m_ppar.freq_shift.Hz = *getFloatParameter(4);
 	m_ppar.spread.enabled = *getFloatParameter(8) > 0.0f;
 	m_ppar.spread.bandwidth = *getFloatParameter(8);
+	m_ppar.compressor.power = *getFloatParameter(9);
+	m_ppar.harmonics.enabled = *getFloatParameter(11)>=1.0;
+	m_ppar.harmonics.nharmonics = *getFloatParameter(11);
+	m_ppar.harmonics.freq = *getFloatParameter(12);
+	m_stretch_source->setLoopXFadeLength(*getFloatParameter(10));
 	double t0 = *getFloatParameter(5);
 	double t1 = *getFloatParameter(6);
 	if (t0 > t1)

Source/PluginProcessor.h

@@ -73,7 +73,7 @@ private:
 	
 	bool m_ready_to_play = false;
 	AudioBuffer<float> m_recbuffer;
-	double m_max_reclen = 5;
+	double m_max_reclen = 10.0;
 	bool m_is_recording = false;
 	int m_rec_pos = 0;
 	void finishRecording(int lenrecorded);