MRoyce wrote:Exhales in a weary groan before we get into yet another a v d debate...For reals though: Yes, the two differ and not every technique can be transferred from one to the other. I really like that this thread has so far gone into the technical part of things and would be very glad if it kept going that way.agreed. you nerds keeps crunching 1s and 0s and i'll keep making rock.

Stinky Pete wrote: The data is stored in a circular buffer and you just read and write around it. Why, on a conceptual level at least, would it be different here?Because the feedback has zero delay. That's the difference you aren't getting. You need to scale the sample and add it back in instantly. Not to the next sample, not to the next block. Instantly. This sample, not the next sample 1/96000th of a second later. And now that you have done that you follow the processing graph back through the feedback loop and need to scale the result and add it back in, instantly. and on and on. That's a zero delay feedback loop.If you add it to the next sample you've made yourself a one pole filter. Which is exactly the kind of delay based artifact that started this whole conversation. Look at the code for the biquad you offered to post. You will find no zero delay feedback, what you'll find is that the previous output sample (z^-1) and the one before that (z^-2) are what are being fed back in to the filter.

tmoneygetpaid wrote:Thanks for your replies. It's good to hear from someone who understands this stuff, even in broad strokes, at the level of the underlying math and code. Same to you, StinkyPete.How did you get into this stuff? Are there resources you'd recommend?I cannot recommend this (free) book highly enough: http://www.dspguide.com/pdfbook.htm

The point is that you can't put an arbitrary processing unit in a feedback path without having artifacts resulting from the intrinsic latency in a digital system. as an aside: This conversation has inspired me to move some stuff from my old Soundcloud account to the Youtube channel where I post things these days. Check out the Luciermatic, an exercise in processing stuff in the feedback loop of a delay line:

tmoneygetpaid wrote: At question is whether this 1-sample will actually impact the effectiveness of this trick,The one sample delay is the difference between each sample passing through the filter once and each sample passing through the filter hundreds or thousands of times, until any result is below the noise floor.They are fundamentally different things.The way to do this on a digital system is with an EQ with some sort of controllable resonance built in that is simulating the same process.On Linux I recommend the MVCLPF-3, a free and open source resonant low pass filter LADSPA plugin that has been available stock on every distribution since at least 2003 or so.I just tried it out, took all of 2 minutes to set up. Sounds cool.If you are using proprietary software on a closed source OS I can't help you.

tmoneygetpaid wrote:Do you mean to say that the one-sample delay actually disallows the feedback? Or that the one-sample delay is compounded?In the analog world the delay is instantaneous. The signal is regenerated over and over again until it falls below the noise floor instantly.If you put a delay in there the first pass through happens instantly. The second a sample later, the third two samples later, the fourth three samples later and so on. This is even assuming you can figure out a way to get the delay in the feedback loop down to one sample. If you just connect output to input you will more likely get a delay of whatever block size you are running at. tmoneygetpaid wrote:Also, I'm unclear on how a resonance control would simulate this. Doesn't a resonance just narrow the Q of a filter/ EQ band? Does it actually commonly use some feedback, as well?I am a (hack) programmer and a (amateur) DSP guy, so we are straying outside of my area of expertise here, but feedback is a fundamental part of how active analog filters work. When you are adjusting the resonance on something like a moog low pass filter you are, at least in part, tuning the feedback.

People didn't seem to be picking up on it.

Stinky Pete wrote:Okay, so the issue is the feedback's latency?Yes. The feedback being the only significant feature of this technique, which would otherwise be "boosting the bass a little."

tmoneygetpaid wrote:steve wrote:Stinky Pete wrote:Okay, so the issue is the feedback's latency?Yes. The feedback being the only significant feature of this technique, which would otherwise be "boosting the bass a little."Right, but they are discussing whether each feedback needs to have a 1-sample delay, and it seems the consensus is there will need to be. I said N samples, but theoretically there might be a way to do the math fast enough that N=1.Stinky Pete says the time constants of the electronic components and slew rates and what not of the analog components in the EQ and console mean that there is a non-zero delay in the analog version of this trick, as well.Right. I mean, I said it first but... as I mentioned, those time constants are on the order of a small phase shift, not an audible delay.

projectMalamute wrote:The way to do this on a digital system is with an EQ with some sort of controllable resonance built in that is simulating the same process.Can I point out that I said this at the outset? ctrl-F for "moog filter" if you like.