Any suggestions? :---))) besides just renting studio time. Maybe I just need to change the settings when it gets the mask??? To?

Had thought the compressor I have been working on would solve the problem.

But, alas, alack, (DOH!!! mark it up as just another example on the learning curve) all the noise is still there when the compressor kicks in.

So that concept is a wash. i.e.: sure it works when there is just noise. But that is not what I want/need.

Eagerly and much excitedly awaiting your response,
-svanya

Desire: that when I play there is no noise on the recording even when I am playing.

You can do this after the fact in Audacity. But it's sort of flaky and only about 30-40% truly effective.

Does that mean I want: noise cancelling?

This sort of does it but like I said it sounds like I'm talking thru water: from the Help>Browser I04.noisegate.pd

Does that help clarify?

Will look for noise reduction/suppression patches.

Do wish I could "hear it clean" tho. Will keep looking. Thanks, for the pointer.

If you don't get carried away with it, the side effects stay pretty much out of the way.

It isn't level dependent. If you set it up so that you get say 3dB of noise reduction, then it will remove 3dB of noise and doesn't care if the noise is at -10dB or at -50dB. It removes 3dB in any case, so down to -13 or -53 in the examples.

There is a time factor that may need to be modified way down inside - this thing was originally intended for voice (speech) rather than music. If you find it muting your (long, lingering) notes then it can be easily modified.

Another effect it has is that even at the lowest setting, it will kill feedback squeal dead.

Thanks, Joseph: and yes I have come upon your work. Very Cool! (besides me having no Real clue how it works :---)). which brings me to a point: which patch would I use to supress incoming noise? And of the various options which could/are the most reasonable to parameterize so I/others could control the effect?

Also: @All: as a footnote/preliminary condition: the incoming noise is static: that is background noise coming in (between 3-5 am) in and around my apartment. So I Can grab a mask/sample of it. (which brings up another point: on the fft type of reduction: are shorter or longer (time) samples/masks better?

?: am thinking now: there might be a way to grab (like the fft and stamp album plugins do) a sample then just subtract it from the incoming signal???

Is that a valid approach to take?

I see there are quite a few libraries which grab samples (and that is a condition I can do).

Info greeted with gratitude and thank you for your hard/smart work and persistence in this matter.

peace
-svanya

Attach a slider to the right inlet with a range from 0 to 2. That ought to be a wide enough range that you can adjust it to clean things up without also destroying the voice signal.

How it works is pretty simple, but difficult for me to put into words. If you are interested in how it works, take a look inside NoiseFilter and NoiseDetector.

NoiseFilter pretty much just attenuates the audio based on the noise level that it finds. How it finds the noise level is the trick.

It uses the subpatch dBr to find the level of the signal, then passes that through a highpass filter and a low pass filter. Those two determine what is seen as "static" noise. Anything that changes slower than the high pass is static noise. Anything that changes slower than the lowpass is signal. Anything that changes faster than the lowpass is noise.

Changing the value of the highpass determines how fast things get filtered out - if it "eats" your long notes, then you need to make the cutoff of the high pass lower.

Once it has the filtered level, it takes the absolute value of the variations in the level and compares them to the selected noise reduction level. Anything below the desired level closes the gate. Since this causes a pulsing signal, theres's a lowpass to smooth it out.

Now, you have a signal proportional to how much noise is in your signal. The more the noise, the lower this proportional signal is.
Multiply that with your original audio, and it attenuates the audio according to how much noise there is.

The rest is splitting it into bands so that you don't attenuate everything at once, but just the frequencies where there's noise.

With that in mind (and until I use the following to see how your patch can help and I can tune it to my ends) I include below the gui I built to better understand and implement your fine and gracious work: a gui/front-end for your MultiBand NoiseFilter.

Feel free to use, shared, adjust, etc. it as it was meant as a learning tool for me and just toooo easy NOT to do.

That all been said, let me get back to some other more pressing work and once I have used this to comprehend what yours does (you were right it is "hard to put into words) I will come back to share what I have discovered.

Peace and good health to you
-svanya

Your welcome to share this on/in/around yours if you would like. I hope you find it a worthy extension of your work. And also, thank you, so much to your efforts in supporting my search. It Does mean a great deal to me.

[These are the Only files in your original work I amended to suit this end...]

MultibandNoiseFilter-help_gui_sv.pd
MultibandNoiseFilter_gui_sv.pd
NoiseFilter_sv.pd

The values I had used for the various bands were chosen to approximate a flat frequency response. So, you adjusted one value, and the noise was suppresed (more or less) evenly accross all bands.

You've made it so that you can suppress the noise by band - sort of a graphic noise equalizer.

The value you've labeled hip_frq is what differentiates static noise from actual content. The lower this value, the longer a sound must be present for it to be removed as noise. Say, you sing a really long note that goes on for 10 seconds. If the hip_frq is set to 1Hz, then that note will be recognized as noise after a second or so and will be filtered out. So, you would set the hip_frq to a lower value (say, 0.1Hz) which will make the filter allow the long note. It also means that the filter will respond slowly to all noise. With hip_frq set to 0.1Hz, it will take around 10 seconds before a sound with a constant volume is filtered out.

The lop_frq is sort of the opposite. Anything whose volume changes to quickly is assumed to be noise. If lop_frq is set too low, then you will lose things like snaps or pops (say, the click of beating drumsticks together or the onset of a cymbal crash.)

The values for hip_frq and lop_frq I used were what I found useful for speech.

hip_frq probably needs to be lower for music, and lop_frq probably needs to be higher for music.

How did it work out for you? Was it any use, or are the side effects too much?