updated FDNMF test code, helpfile-in-progress and class description (and filename)

corrected FDSTFTPass class description and helpfile
corrected FDGain helpfile

fdNMF/tests.scd

@@ -17,7 +17,7 @@ y = Buffer.new(s);
 // without sources
 r = Routine{
 	t = Main.elapsedTime;
-	FDNMF.nmf(s,b,0,-1,0,-1,nil,x,0,y,0,5,100,~frame_size,~hop_size,~fft_size);
+	FDNMF.nmf(s,b,0,-1,0,-1,nil,x,0,y,0,5,100,0,~frame_size,~hop_size,~fft_size);
 	s.sync;
 	(Main.elapsedTime - t).postln;
 }.play
@@ -27,7 +27,7 @@ r = Routine{
 (
 r = Routine{
 	t = Main.elapsedTime;
-	FDNMF.nmf(s,b, 0,-1,0,-1,c,nil,0,nil,0,5,100,~frame_size,~hop_size,~fft_size);
+	FDNMF.nmf(s,b, 0,-1,0,-1,c,nil,0,nil,0,5,100,0,~frame_size,~hop_size,~fft_size);
 	s.sync;
 	(Main.elapsedTime - t).postln;
 }.play
@@ -78,6 +78,7 @@ c.plot;x.plot; y.plot;
 
 // test with stereo input
 Buffer.freeAll(s)
+b.play
 
 (
 b = Buffer.read(s,"/Users/pa/Desktop/verystereo.wav");
@@ -92,12 +93,12 @@ y = Buffer.new(s);
 (
 r = Routine{
 	t = Main.elapsedTime;
-	FDNMF.nmf(s,b,0,-1,0,-1,c,x,0,y,0,5,100,~frame_size,~hop_size,~fft_size);
+	FDNMF.nmf(s,b,0,-1,0,-1,c,x,0,y,0,5,100,0,~frame_size,~hop_size,~fft_size);
 	s.sync;
 	(Main.elapsedTime - t).postln;
 }.play
 )
 
-{PlayBuf.ar(10,c.bufnum,doneAction:2)[2].dup}.play
+{PlayBuf.ar(10,c.bufnum,doneAction:2)[9].dup}.play
 
 c.getn(0, 10, {|x|x.postln})

release-packaging/fdGain/HelpSource/Classes/FDGain.schelp

@@ -1,5 +1,5 @@
 class:: FDGain
-summary:: Real-Time Gain Changer
+summary:: Real-Time Buffered Gain Changer
 categories:: UGens>Algebraic
 related::Classes/UnaryOpFunction
 
@@ -15,21 +15,13 @@ ARGUMENT:: in
 	The input to be processed
 
 ARGUMENT:: frameSize
-	The size of the real-time i/o buffer. It will add that much latency to the signal.
+	The size of the real-time i/o buffer, in samples. It will add that much latency to the signal. This is not modulatable.
 
 ARGUMENT:: gain
 	Audio or control rate change of the gain.
 
-returns:: nothing
-
-INSTANCEMETHODS::
-
-METHOD:: checkInputs
-	This makes sure the value and phase inputs are both at audio rate.
-
 returns::
-	A slap on the wrist, should it not be the case.
-
+	Same as input, delayed by the windowSize, multiplied by the gain factor.
 
 EXAMPLES::
 	Summing with the inverse (gain of -1) with a delay of the latency gives us CPU-expensive silence.
@@ -38,9 +30,9 @@ EXAMPLES::
 	::
 	Varying the gain at audio rate.
 	code::
-	{ FDGain.ar(PinkNoise.ar(0.1), gain:LFSaw.ar(1)) }.play
+	{ FDGain.ar(PinkNoise.ar(0.1), gain:LFTri.ar(1)) }.play
 	::
-	Varying the gain at comtrol rate.
+	Varying the gain at comtrol rate, in beautiful stereo.
 	code::
-	{ FDGain.ar(PinkNoise.ar(0.1), gain:LFSaw.kr(0.5)) }.play
+	{ FDGain.ar(SinOsc.ar([222,333],mul:0.1), gain:LFTri.kr([0.5,0.7])) }.play
 	::

release-packaging/fdNMF/HelpSource/Classes/FDNMF.schelp

@@ -79,7 +79,6 @@ INSTANCEMETHODS::
 EXAMPLES::
 
 code::
-	b = Buffer.read(s,"../../AudioFiles/01-mix.wav".resolveRelative);
-thisProcess.nowExecutingPath
+	b = Buffer.read(s,"../../../AudioFiles/01-mix.wav".resolveRelative);
 	b.play
 ::

release-packaging/fdNMF/classes/fdNMF.sc

@@ -6,8 +6,6 @@ FDNMF {
 
 		if(srcBuf.bufnum.isNil) { Error("Invalid buffer").format(thisMethod.name, this.class.name).throw};
 
-		dstBuf.bufnum.postln;
-
 		server.sendMsg(\cmd, \BufNMF,
 			srcBuf.bufnum, startAt, nFrames, startChan, nChans,
 			if(dstBuf.isNil, -1, {dstBuf.bufnum}),

release-packaging/fdSTFTPass/HelpSource/Classes/FDSTFTPass.schelp

@@ -1,2 +1,49 @@
-//Ar modulator
-{FDSTFTPass.ar(SinOsc.ar(440),1024,256,2048)}.play
+TITLE:: FDSTFTPass
+summary:: Real-Time FFT/IFFT return trip.
+categories:: UGens>Algebraic
+related:: Classes/UnaryOpFunction
+
+DESCRIPTION::
+A sanity test for the FluCoMa Real-Time Client FFT/IFFT Wrapper footnote::This  was made possible thanks to the FluCoMa project (http://www.flucoma.org/) funded by the European Research Council (https://erc.europa.eu/) under the European Union’s Horizon 2020 research and innovation programme (grant agreement No 725899).::
+
+
+CLASSMETHODS::
+
+METHOD:: ar
+	The audio rate version of the object.
+
+ARGUMENT:: in
+	The input to be passed-through
+
+ARGUMENT:: windowSize
+	The size of the buffered window to be analysed, in samples. It will add that much latency to the signal. This is not modulatable.
+
+ARGUMENT:: hopSize
+	How much the buffered window moves forward, in samples. This is not modulatable.
+
+ARGUMENT:: fftSize
+	How large will the FFT be, zero-padding the buffer to the right size, which should be bigger than the windowSize argument, bigger than 4 samples, and should be a power of 2. This is a way to oversample the FFT for extra precision. The -1 default value will default to windowSize. This is not modulatable.
+
+returns::
+	Same as input, delayed by the windowSize.
+
+
+EXAMPLES::
+
+code::
+	Summing with the inverse (gain of -1) with a delay of the latency gives us CPU-expensive silence.
+	code::
+	{ var source = PinkNoise.ar(0.1); DelayN.ar(source, delaytime:1024/s.sampleRate, mul: -1) + FDSTFTPass.ar(source, 1024, 256, 1024); }.play
+	::
+	Larger, oversampled, FFT
+	code::
+	{ FDSTFTPass.ar(PinkNoise.ar(0.1), 2048, 128, 8192) }.play
+	::
+	Stereo Input Tests.
+	code::
+	{ FDSTFTPass.ar([SinOsc.ar(222,mul: 0.1), PinkNoise.ar(Decay.ar(Impulse.ar(0.666,mul: 0.1), 0.5))], fftSize:1024)}.play
+	::
+	Stereo Parameter Tests.
+	code::
+	{ FDSTFTPass.ar(SinOsc.ar(222,mul: 0.1), [1024,8192],256,8192)}.play
+	::

release-packaging/fdSTFTPass/classes/fdSTFTPass.sc

@@ -1,5 +1,5 @@
 FDSTFTPass : UGen {
-	*ar { arg in = 0, windowsize=1024, hopsize=256, fftsize=windowsize;
-		^this.multiNew('audio', in.asAudioRateInput(this),windowsize, hopsize, fftsize)
+	*ar { arg in = 0, windowSize= 1024, hopSize= 256, fftSize= -1;
+		^this.multiNew('audio', in.asAudioRateInput(this),windowSize, hopSize, fftSize)
 	}
 }