In Synthland there's a similar technique called pinging where a filter just below self-oscillation is pinged by a short sound. It momentarily oscillates and the decay is controlled by just how much resonance is dialed in. It's really nice for boomy kick-drums such as 808s. In the digital world I'd use a resonant filter plugin on a separate channel and turn up the resonance, that will get a very similar result.

Stinky Pete wrote:projectMalamute wrote:What Steve is describing involves the signal going from the output and back to the input a theoretically infinite number of times instantly. You can't do this in a DAW. What you can do is calculate what's happening and implement it another way, but you can't just patch the output to the input.Of course you can, take this psuedo-code for example:Output X = (whatever the calculation to perform the EQ).Input = N + (X*0.8 [or whatever the feedback gain value would be to get the desired result].Your code is just running it through the EQ one time and then scaling it. There is no feedback going on.What Steve is describing would look like this in C:Code: Select all// X is our input sample, eq(float x) calculates the processing done by the EQwhile(true){X = X + eq(X);}ie. x + eq(x) + eq(eq(x)) + eq(eq(eq(x))) + ... out to infinity.

tmoneygetpaid wrote:So are you acknowledging now that a plugin that doesn't create latency is possible?Bussing does not cause latency/ delay either, so where would this delay be coming from?A plugin that doesn't create any latency beyond the buffer in the DAW is possible.There is no way to get rid of the latency in the DAW itself. In theory, with specialized hardware, you could get down to 1 sample. I think a Protools HD rig can do 32 or maybe 16 samples. What Steve is describing involves the signal going from the output and back to the input a theoretically infinite number of times instantly. You can't do this in a DAW. What you can do is calculate what's happening and implement it another way, but you can't just patch the output to the input.

eliya wrote:I believe you're right here.Well la-de-fuckin' da.

tmoneygetpaid wrote:Worth noting that HD Pro Tools has plugins that have zero samples delay, so you could use a TDM/ AAX DSP plugin for this purpose.This is of course impossible, so the language "zero samples delay" means they are compensating for the algorithmic processing time on the effected signals or the session as a whole.Which is what I was talking about.

Stinky Pete wrote:The Q of most parametric EQ is set largely through an internal feedback loop.Exactly. If you regenerate the audio through the EQ you increase this effect beyond the design of the hardware unit, tightening the Q to be precisely tuneable and verging on oscillation, and that's the auxiliary signal you mix with the "normal" bass drum. The reason you do it in parallel is to leave the original bass drum track intact, while adding a small amount of the ringing resonance. If the audio is run through the equalizer with high feedback near the point of oscillation, the effect will be much more drastic than the same signal being mixed with the unmolested bass drum to taste.

tmoneygetpaid wrote:All audio is delayed by the buffer amount.Plugins access the audio stored in the buffer and manipulate it before it is "queued" in the digital audio stream.And then the audio, after being queued into the digital audio stream, is partially fed back through the bus and into the buffer.

andteater wrote:steve wrote:I'm happy to demonstrate the effect sometime, just pop in during any mix session.is this an invite for all of us, because i've been specifically told i'm not allowed near the studio for a while.I believe the court order says 150 feet. I'll turn it up so you can hear it from the corner.

tmoneygetpaid wrote:So are you acknowledging now that a plugin that doesn't create latency is possible?Bussing does not cause latency/ delay either, so where would this delay be coming from?When you put the audio into a buffer (delay) in order to process a signal through the plug-in, then take the processed audio (delayed) and feed it back through the buffer (delay) in order to re-process it (the feedback path), you will reiterate the delay.That is why I suggested several times that you might be able to do it with an analog filter model that incorporates resonance (like a moog filter), but not using any old eq plug-in.I mean, please don't believe me. What do I know. Try it yourself.

projectMalamute wrote:nvolves the signal going from the output and back to the input a theoretically infinite number of times instantly. Exactly. In an analog circuit, the voltage appears instantly at the input of the circuit when fed back from the output because the voltage appears across the conductive path all at once (okay there are skin effects etc but practically it's instantaneous). The filter network has a time constant, but that is on the order of a phase change rather than an audible delay.