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."Sorry for ghosting this thread, got busy. Of course! My question, however, has been does 1/48000 (Max) of latency effect this process significantly? Both filters will have a rise time, the digital systems will be longer but would it be perceptually noticeable? It'd have to be in the tail of each kick.projectMalamute wrote: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.No genius myself, for the record, my C++ is pretty basic and the majority of my knowledge comes from a guy called Will Pirkle's book on designing audio plugins. Irrespective of that, all IIR type filters rely on feedback to get any sort of usable Q value, analogue or digital. Only passive RC circuits lack feedback, but any EQ on a desk should be an active filter with at least a passive feedback stage (shunt or series). Parametric ones having active feedback stages to keep Q independent of gain or center frequency.The real issue is whether or not the digital version is functionally identical. I actually think it would be, and I think the grievance with the digital system here is more philosophical than something that would interrupt this particular method. Thankfully people are avoiding the A vs. D debate. I have a 4 track cassette machine from the 80's that I love, I'm more of a whatever-does-what-I-want kind of guy.To whomever said the PT's DSP version kept stable when bussing and the Native didn't: WOW. Fuck PT, those guys are a bunch of jokers and the main reason digital audio has a bad name. Really glad the industry has drifted away from PT, I always hated them and their proprietary hacked together systems. Cubase has had sample accurate latency compensation since the 00's, there's no excuse.

bishopdante wrote:This depends upon all sorts of parameters, particularly the frequency of EQ (eg: for 115Hz, one sample delay at 192kHz sample rate is of minimal phase angle here or there, whereas 16kHz at one sample delay at 44.1kHz sample rate there will be far more influence to consider)...[Loads of true things, just shortening for brevity]...For the purposes of emulating FM steve's process with minimum hassle, simply route an aux send to a channel of DAC and a mixing desk channel, and set up the feedback and EQ from there, either using an outboard EQ or a desk channel. Ie: mix "out of the box", or use analogue processing as an aux send from the digital system, and have the feedback loop functioning in an all-analogue domain fed by a digital stem.There is a lot to be said for having even a basic analogue mixer and working "out of the box" when working with a multichannel audio interface and computer. It has a number of advantages.I agree 100%, I think at the low frequencies we're talking most sample rates would not cause issues with the technique. I think you'd find it'd work in a way that is, outside of the technical, perceptually not noticeably different in effect.But yeah I agree, seems ways more fun to do it in analogue anyway. As I said, not a digital advocate per-se, I just know some bits. Might run a kick drum out to the Moog some day and see if it does it.

Redline wrote:As I said, not a digital advocate per-se, I just know some bits.That was funny stuff...Didn't even C:/ that..

What's interesting to me about this "effect" is that the instant it was explained, I could hear it in my memory of certain recordings. That boomy, vaguely menacing quality. Though it's difficult to simulate in the digital domain, I came up with something similar. On my own recent kick drum recording, I used phase cancellation to extract a very narrow eq band, added extremely fast delay with a substantial feedback, put a hard limiter on it, and mixed with the original recording to taste. It's not too far off, just a bit artificial sounding compared to a pure feedback.Still can't be done in real time, but that's only important if you're printing the effect to tape, which I'm sure is important in a professional studio environment where workflow needs to be as efficient as possible.

i doubt it would be used to tape.

I thought I completed my kick-drum only solo album project, but now I have a new path to follow.

On my way to mix on the console (hybrid mixing is my first step), I could compare how sounds the resonant eq in my DAW and how it sounds in the console. Forget the DAW to make your resonant eq. Even with very short buffer size, you will never get to this result we're looking for. But I still have a question: do you use pre or postfader ? steve wrote:I'm going on about this because there's a semantic tic that keeps getting used as though it were a fact, and it's fun to beat dead horses. Live ones no, that's cruel.A process cannot actually have zero latency because an operation takes time. The speed of operations is determined by the clock of the CPU or whatever coprocessor governs that operation. The term of art "zero latency" means the processing time can be compensated for, usually within a buffer, not that it happened instantly by some magic that belies the existence of the fucking clock. A better term would be zero "effective latency."If it were possible to do processing with "zero latency," then reiterative feedback would happen instantaneously. Instead, the signal feeding back is delayed by the greater of the buffer or the latency of the process. If you use a bus to send to a processor, then route the output back to the same bus, the latency will reveal itself as a period (frequency) in the feedback generated.Why is there a buffer? Because doing math takes time, and how much time defines the latency of the processed audio. If the buffer is big enough then there will be enough time to do all the math and offset the audio so that it stays in sync, giving you zero "effective latency," which is not the same thing as the calculation taking "zero" time.I mean, please don't believe me. All of this is spelled out in the first post I made on this topic. I don't use computers for audio, but I know they can't do math by magic instantly. Just try it. If a thing truly has zero latency, you should be able to feed its output back into its input without any delay because zero plus zero is still zero.But what the fuck do I know. Careful, that's a rake.

bishopdante wrote:tmoneygetpaid wrote:Aside from the issue of wires and the speed of lightPerhaps Quantum Entanglement will be the solution to some of these time based issues in the future?