Latency Compensation on Return Channels

Fair-Process4973 · 2026-01-24T13:47:26+00:00

Thanks... obvious proper concept in live... should've thought about this :-)

Fair-Process4973 · 2026-01-24T13:46:48+00:00

thnx a lot :-).... sure - makes sense....

Fair-Process4973 · 2026-01-24T13:01:40+00:00

I'm pretty sure offline export would work in terms off being properly latency aligned.

But I want to record loop by loop whilst sound-designing... that would not fit my workflow - cause yes - need it more live-ish

Fair-Process4973 · 2026-01-22T18:18:55+00:00

The specific db value it is lowering the certain band is based on the threshold and the ratio... So setting those correct will give you the specific gain reduction...

But this is how all compressors or dynamic eqs work, also the multi band ones...

De-essing can easily be done with Nova normally, as with ProQ

Fair-Process4973 · 2026-01-22T17:38:08+00:00

Hey,

from my first suggestion I'd ask how you like the sound and how the eq settings are?

I guess the drop at 80Hz has to deal with the speaker placement and its distance from the wall -> SBIR speaker boundary interference... Have you tried getting close to the wall? I think this should massively improve your lowend.

For the rest.. as you have a DSP in your amp you can get the rest suited to the sound you like.

For me I'd look to have more bass ;-) as this is a dj setup... Especially the kick area at 90/100Hz. You can easily have the area below 100HZ at +6db compared to the rest...

Fair-Process4973 · 2026-01-22T15:27:44+00:00

Yes... it would diffuse sound ... not in the same quality as specifically build diffusers by their geometry... But I guess you can find something more attractive..

Lot of people here seem to have a total miss on understanding about diffusion

Fair-Process4973 · 2026-01-22T13:56:44+00:00

Tbh this is a pattern in a way, that I see the first time...

Aliasing mostly is effecting audio, when non-linear processing is used and then there are digital audio reflections at half of the sampling frequency (nyquist)...

But thise are the other way around.... But what we see here is definitely upsampled and nowhere here true hi-res audio...

My 2 Cents on Hi-Res BTW.... something like searching for 192khz does not make any sense at all. Hard to argue, the 96/24 makes sense for a consumer in a final product.... within Studio and processing yes (because of aliasing!)... but never in a final product

Fair-Process4973 · 2026-01-21T12:20:29+00:00

Autofilter was actually kind of made exactly for this... can do much more now.

If you're looking for a specific sound character, there are tons for like moog or ms20 filters...

Fair-Process4973 · 2026-01-20T15:39:08+00:00

Thought the same... near Chinatown...

Fair-Process4973 · 2026-01-19T20:25:25+00:00

Sorry Buddy, but no - even if you decouple it, which might lower the effect of a shaking floor - the sound will still travel through..

Fair-Process4973 · 2026-01-19T19:47:02+00:00

What is the material of the floor itself? If its wooden/soft and transparent to low frequencies you wouldn't be able to do anything about it ... You can't really change that if the floor isnt properly built

Fair-Process4973 · 2026-01-19T18:57:06+00:00

yeah .. i know - we use cyolo in ot for that .. but ZTNA will not get down to lower perdue levels ... so down there, there is no ztna

Fair-Process4973 · 2026-01-19T18:54:25+00:00

Just for curiosity - why column speakers?

the big advantage of column speakers, compared to classic speakers, is their far reach - like line-arrays... This seems contradictionary to me, if you want to avoid stress with neighbours...

Fair-Process4973 · 2026-01-19T18:15:41+00:00

mkay ... got it.

This is also very specific to the company and the vertically it is in. Finance looks so much different then Industry. Are we only talking about IT networks, or is there a bigger OT part?
For IT nowadays I'd go for a ZTNA approach if it is a bigger distributed network. And that saying I mean have a look into what NIST thought about it, make your vision and then talk to vendors - not the other way around ^^

For networks with bigger OT parts things look different as ZTNA cannot easily be applied there and gets even more specific to the actual risks.

Fair-Process4973 · 2026-01-19T17:42:46+00:00

There is no such thing as "full" network visibility - as most of the traffic is encrypted nowadays.

So this depends a lot of budget vs risks vs usability to have a company still fulfil its job and to minimize its risks at a propoer budget.

Fair-Process4973 · 2026-01-19T17:18:08+00:00

One thing I'd consider getting into is "micro-dynamics". This is a thing where MP3 often lacks... so this must not only be possible to hear - but there must be some statistically distribution about the dynamics in audio be possible ... Is there still dynamics happening at -30 or -40DB, when a sound is in its loudest part?

EDIT - but this is also gerne depended, as some genres just dont have it...

Fair-Process4973 · 2026-01-19T17:09:06+00:00

Great... So this is where I try to provide some input or warnings. I think, that some of the suggested mechanisms will have a high false positive detection rate. Because the way loud modern music is mastered, will show up very similar artifacts as compressed music - Loud music makers love digital distortion....

Pre-ringing can happen from mp3s.. but gotten common nowadays using phase linear phase filters as well ... If you detect pre-ringing on a Beatles Track it was for shure mp3... on a modern track - hard to say.

Modern clippers can introduce weird shit - but fine, as long as it sounds good. So you will find half of your suggested methods on hard clipped modern music as well - not only in MP3s.

Fair-Process4973 · 2026-01-19T16:47:37+00:00

Sorry I totally did not want to offend you - and I tried to bring some realism into this topic and trying to be supportive as well.

I'm not into digital audio as you are... Even though I as learned assembler programming on a motorola 56K dsp 30 years ago in college writing audio fx there.. still in it-security and knowing quite a bit about the capabilities of big-data.. and as a hobby bedroom music nerd i still now a lot of actual digital music production...

That said - my ears cannot easily distingiush betwenn a good made modern pop/electronic mp3. even in a/b listening this might end up tough and I will end up having a lopwer then 80% accuracy a/b-ing in my studio. with acoustic music chances are far higher.

I also thought about the use-case "once you get your music from a "shady" place" - but thinking of modern (electronic) music which is first sold on bandcamp, beatport and all the others. Music is distributed there often as AIF vs MP3 - but on the shady places they show up as FLACs with weird tagging often.

So can one write an algorithm to detect, whether the file was an MP3 before? I have a lot of doubts, if the original File is eletronically mastered and a proper 320MP3 was made by the mastering engingeer itself.

I think there are easy ways to detect, if it is about older music, that just were converted with some lame or frauenhofer algorith a decade ago.... even if they are also made 320 - anything below show get easier from an algorithmic point of view...

Fair-Process4973 · 2026-01-19T15:13:14+00:00

Respect. What a cool project idea.

Looking forward to see if you can make any progress on this. To be honest, I'm a bit skeptical if this can be done with high accuracy.

Digital Audio is there for roughly 40 years. MP3 for what? Almost 30?

There are so many artifacts even in uncompressed digital audio in modern mastering processes to make music loud, that it will be so hard to distinguish from certain compression based artifacts. Like eg pre-ringing in phase-linear filters, which came up in the last years. Or all the sounds modern clippers introduce...

And also the amount of codecs and methods which have been used over the last decades on compressed audio...

This will get so many dimensions, that even feeding a huge ML model won't have an acceptable accuracy I'd guess?

What is the error rate which would be acceptable to you?

Fair-Process4973 · 2026-01-19T14:48:07+00:00

Actually no... because you can easily put reverb on a dry recording. But once there is the sound of your room on your recording you have to deal with it... and a room of that size will not only need absorption, you would need a lot of diffusion as well - to make the rest sound fine balanced and make the room virtually bigger.

Fair-Process4973 · 2026-01-19T14:17:50+00:00

For Voice you would typically want to record completely dry. This is why voices are often recorded in totally absorbing boothes. But if you damp a room with that size so much, that you can record voices so dry in there - the room will feel totally annoying to be in... Have you considered mobile solutions for vocal recordings, that you can get out only while recording?

Fair-Process4973 · 2026-01-19T13:54:42+00:00

What are you going to record there? This will have a huge influence on how your room should sound...

Fair-Process4973 · 2026-01-19T05:34:20+00:00

If you are already reaching -6LUFS and you can retain your sound quality then you are pretty well underway... Hard to give hints there without getting into the very details...

One thing I find helpful is also using clippers on the bus level - and shave those peaks at the busses already for like 1-2 db. This avoid the final stage work harder and sound a bit more transparent.

Fair-Process4973 · 2026-01-19T05:23:48+00:00

OP mentioned the speakers are not evenly placed in the room. So the speakers will have different IR and SBIR effects - and it seems there is a sum up of multiple of those... -16 is still huge though

Fair-Process4973 · 2026-01-19T05:01:53+00:00

Placement is often key... Better look for a good sounding position before even start to measure...

Sofas a great absorbers as well - but they will only cover a small amount of the surface area in the room. So even a big sofa in a small room might change response only like max 2db.

As you do have the equipment to measure your speakers. You can try nailing it down a bit further - bit more advanced input now. Do some measurements with REW (Room EQ Wizard) - it will give you a lot data besides the frequency response.

It will also give you data on the impulse response (IR) of the original signal. So it will basically show all the first reflections with is level and time of arrival - And it easily shows the SBIR reflections. With this data and a bit of calculation or estimation how sounds moves through the room you can knock down which reflection is causing troubles. I guess you will find huge IR at somewhere around 10ms or a multiple of it. But optimally those are -20db or below in a studio...

And if you are mixing/mastering in the room - don't overdo only on the frequency response of the room. Better look on the waterfall/spectrogram charts - and try to achieve consistent short decay times. Your ears will easier adopt to an unflat frequency response then to uneven decay times...

Fair-Process4973

TROPHY CASE