Software to Apply FIR Filter - No Math

No I don't -- I'm repeating things I've heard, but haven't experienced.

A freeware program called Ardour might be the ticket. It's kinda like Audacity on steroids.

It's a full-fledged audio mixing "deck", with recording and monitoring capability. Its "features" list claims "Route anything to anywhere. Matrix-style patching/routing. Connect Ardour tracks or busses to hardware, each other, other applications, the network... input, output, sends, inserts, returns all managed the same way."

It supports sound-effect plugins in "LV2, native VST and LADSPA formats on Linux and OS X... AudioUnit (AU) plugin support on OS X... high quality proprietary plugins on Linux... excellent open source plugins from Distrho and others."

"As a general rule of thumb, most modern computers will be able to handle 24 tracks with moderate plugin processing (some EQ and other less intensive processing per track, plus compression and reverb on busses) without much of a problem as long as your latency settings are not too demanding."

Sounds to me as if handling a single input, with one DSP equalizer/filter, and a couple of monitor taps (before and after EQ) would be well within its capabilities.

There's probably a significant learning curve to getting it set up, of course... just as there would be with a real, physical mixing deck.

Did you bother to investigate the audacity plugins? Or do you expect me to solve your problem completely?

I gave you three ways to do your task except for real time. I never needed that. I just take a sample and determine what kind of filter I need, then apply the filter. Problem solved.

BTW, there are reasons not to filter in real time. You can reduce artifacts of filtering by using a filter that is half as good as what you need for the final product. Time reverse the signal, filter it, reverse it again, then filter. This reduces the ringing. Yes, this was also done in the analog tape days.

Tim, it took a man to admit that. You made points in my book (for what that's worth...)

doubles the order, square magnitude response and zero phase

but why would it reduce ringing?

-Lasse

My OP above specifies processing of the signal _ in real time_ as acquired. It also hints that I do not want to spend weeks/months learning to do something from scratch if someone else already has a handle on it.

Judging from most responses so far, it is either technically impossible (extraneous "solutions"), or insulting to people who don't want to "do my work for me".

Or ... if you can't design it you wouldn't know how to use it anyway, etc. Really?

All I am asking is if there is an existing product or methodology to do the job. Something midway between two paper cups and a string, and a graduate degree in high level programming.

Maybe not.

Didn't we love the days when the HAM radio operator nextdoor knew all the answers.

Ken Rockwell

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Ther is another critter called a parametric equalizer which does different though somewhat similar things. All filters introduce some delay though. The delay may interfere with the listening experience.

?-)

And have you looked at the audacity plugins?

Personally, I have zero need for real time fiddling with the filter parameters. Zero. Zip. Nada.

This doesn't mean it hasn't be done. For instance, there are noise limiting schemes where multiple bandpass filters are run on a signal in real time. The noise limiting is done by reducing the amplitude of bands with little energy. Still, that will not meet your needs.

You have to realize the GUI is often more difficult than the signal processing. I could give a sharp high school age student the task of programming a digital filter. It is that easy if the filter is static.

Actually the opposite is true. I explained this in my other post. Some people reverse the audio, filter, reverse it again, then filter it. This was done in the analog days with mag tape recording.

With Audacity, this is pretty easy to test for yourself. You need two filters. One is the aggressive filter. The other filter would ideally have the square root response of the aggressive filter. Your comparison would be the filtered output of the aggressive filter versus the sequence of reverse audio, filter, reverse again, then filter again. The idea is to reduce the ringing. The human brain knows listening to audio in a hollow pipe quite well. The pre-echo doesn't register as badly with the brain.

I'm sure this, like any human perception issue, is argued every which way, often leading to bar fights between the pocket protector crowd.

On a sunny day (Wed, 05 Feb 2014 12:20:36 +1100) it happened snipped-for-privacy@imagenet.com wrote in :

Come of it man, you cannot even properly specify _WHAT_ you want to filter. This I wrote and has a nice equalizer that will filter YOUR noise in real time:

Now, wow, it is free, wow, it has a GUI, and wow probability is you still cannot use it. The whole GUI requirement shows you are clueless. As somebody suggested, use 'sox', in Linux.

And even my xpequ has latency, so if you want to play along with some other part of the band via this you will have to start a tick earlier :-)

BTW the equalizer code I took from xine IIR. If you bothered to learn C, what really is not difficult and only takes a few decennia, you could do things yerselves.

Since you are secretive about what you are doing, here is another direction that may help, though no one can tell until you open your Kimono a bit. You can make active filters from op amps - ti.com and analog.com have free calculators to specify the components. Some of these filters are one op amp and two other components. jb jb

It reduces the amplitude of the ringing by spreading half of the energy forwards in time and the other half backwards

All we need now is Mr. Fusion!

Jamie

On a sunny day (Wed, 5 Feb 2014 18:00:37 -0500) it happened "Maynard A. Philbrook Jr." wrote in :

Somebody once mentioned to me that the was sort of surprized that bad music played backwards was still bad, and good music played backwards still sounded good.

This can be perhaps be understood from a spectral POV, it also goes for playing at double and half speed.

Hi Randy, I think Tim's making a valid point here.

I recall, several years ago, passing a few words of a speech signal (in '.wav' format) through a linear phase FIR filter. The input signal was a second or two of silence and then speech. I seem to recall that the output of the filter had a low-level, but audible, little "pop" just before the beginning of the first spoken word. I'll try to find that '.wav' file and send it to you.

I would be interested to hear that, although I'm not sure my ears could at this stage. I'll give it a shot though.