I guess you can help educate me a bit then.
So latency track-dependent? I.e. too many plugs on one track will cause that track to lag compared to others with less 'intensive' plugs on?

Also, with regards to 'intensive', it's not necessarily CPU-intensive is it? Isn't it something to do with the plug buffers?

yes, latency is definitely track dependent, and that is subject to how many/ what kind of plugins you're using. The more the plugin does to the signal, (or how efficient the pluginarchitecture is) the more it's going to lag.

funny thing is, how CPU intensive it is, is subject to what you set the buffer to, the lower the buffer, the more work the cpu does. If you've got a baller ass cpu you could set the buffer lower and get away with less latent plugs.

you won't completely avoid latency though, even with pro tools HD you have delay compensation which (obviously) compensates for mismatched delay during playback. but it's still there. the reason why everyone's all about HD is for one the tdm architecture handles higher quality resolutions and B since it's not sharing resources with the internal system ram its WAY less latent and B is HD comes with delay compensation (something digidesign could easily include with LE) i think programs like reaper have delay compensation integrated though, i could be wrong...

when you use plugins on an LE system the computer's priority is system intensive (os operation) THEN AUDIO make sense?

Yeha that's perfect, cheers man!
Surely there would be utility plugs out there that compensate for the plug delay? if you get me

i'm not aware of any, but if you find something please do post it

Mellowmuse Auto time adjuster. It's more complicated that it sounds though.

Yeah I could imagine it being complicated, definitely not impossible though.
Have you used that Mellowmuse plug?
Would have been hilarious/lucrative if I'd just stumbled onto a new plug!



Back to Izotope stuff, if you care to read all this:
It's gonna cost me £333 for Ozone, Alloy, Trash, Spectron. Whilst that is a great price (50% off), it's a lot of money to outlay before I go back to uni for an MA that I'm paying for. Whilst I really like all of them, the only one I'd sacrifice would be Alloy.

Alloy's EQ is great but only for it's visual features. It performs similarly to my native EQ and obviously it's best to just use your ears, not your eyes, when handling sound.

The dynamics module, again, is great for it's visual features (and, to be fair, the fact it's multi-band) but it's sonic results aren't all that different from the native stuff I have, let alone some other dynamics plugs I've got; I've got the IK Multimedia Opto Comp which is pretty fit, and also the Sonalksis CQ1 is quite a good multi-band compander.

The limiter isn't the best, at least it's no where near as good as my Sonalksis MaxLimit (single-band limiter but pretty mint).

I've got Superior Drummer with a built-in Sonalksis Transient shaper and I use SD to do all my drums now. However, the alloy transient shaper could work well on overly-bright/'attack-heavy' acoustic guitars or the like. In the past, I've manually automated the volume to tame the heavy attacks. I don't have a dedicated transient shaper.

Also, the exciter is quite interesting for adding a bit of 'life' here and there, and again it's multi-band. I don't have a decent exciter so this, along with the transient shaper, is one of the hooks of alloy for me. But then Trash has some subtle distortions/overdrives that could work just as well I guess (without enabling all the other pre/post-filter and cabinet sim modules).


What do you reckon? I've got by mixing without it, so should be able to after I've demo'd it's features!

considering the superior drums sound really nice without any plugs on it, i'd try to avoid adding anything more than you need to, less is more j00 know?

Yeah I don't tend to mess much with them, sometimes I just add a bit of sparkle on the OH or do my own parallel compression if need be. But I mean, if I did want to shape the transients, I can do that within SD, so don't need the Alloy transient shaper for that purpose (although I've never used any of the SD plugs because,as you say,they sound nice as they are)

[quote=Xomblies;18165493]when you use plugins on an LE system the computer's priority is system intensive (os operation) THEN AUDIO make sense?[/quote]

Actually, this is generally not true: well-designed drivers will work in the interrupt cycles. They'll get precedent over everything but the most primitive and vital functions of computer.

[QUOTE=Moseph;18166697]Actually, this is generally not true: well-designed drivers will work in the interrupt cycles. They'll get precedent over everything but the most primitive and vital functions of computer.[/QUOTE]

This generally IS true... the "most primitive and vital functions" are what i'm talking about coming before audio handling. Do you program drivers for OSX or microsoft? how do you know priority the AI for the OS vs DAW?

BTW, What are you trying to do here man? I'm trying to help people keep from making mistakes that add up to poor quality and you're trying to counter what i'm saying with some extremely subjective situation (more instances than this). No "native" system is going to compare to a pci-e soundcard with it's own ram (TDM) no matter how "well-designed" the drivers are. I know there's always an extreme situation where "it could work". Your contributions would be more enriching if you avoided those gray area situations.

I've said this before, you may know what you're talking about mosef, but your recordings say you hardly know what you're doing.

to reiterate what i said before, keep your plugins low, the amount of latency from each one eventually adds up, (RTAS, VST or AU) creating offset sine waves, given enough delay the poles of each sine wave peak could become (kind of) inverted, and when that happens you get phase cancellation a huge reason why mixes come out muddy, or why you can blast a track and no matter how loud you make it, it still sounds thin and inaudible.

if you REALLY want to get into detail with it, the positive peak or the upper peak of the sine wave pushes the speaker out, while the negative or lower peak will pull the speaker in, so when i say phase cancellation, i'm talking about the speaker being pulled and pushed in opposite directions.

try printing your superior drummer drums and then after you record make sure whenever the drums HIT that all your tracks (at least most of them) have a positive sine wave (especially with bass, you could even highlight the area where the kick has the most attack and reduce the gain on the bass track just for that small moment, almost like manual side chaining without the ooonse sound some electro music has

@Xomblies:

Yeah I understand phase cancellation, but just to be sure, wouldn't the bottom snare track be negative seeing as its waveform is naturally inverted due to the manner in which it was mic'ed?

Thanks for explaining the latency issues though. Buffers are another thing that confuses me. Like PT has the H/W Buffer size set to 1024 by default. I've read about all this in the past but I've forgotten it all now, I didn't really take it all in. Seeing as you seem to be in the know, what exactly is the hardware buffer size? And I take it plug-ins have their own individual buffers? As you've said, the lower they are the more CPU-intensive the plug becomes; is that because it has to 'buffer' fewer samples at more instances?

Also, if I print/track the drums out of SD, there are natural delay issues with bouncing aren't there? So just printing can knock drums out of time, albeit probably not with themselves, if you get me.

:confused:

quite intuative, the bottom snare IS out of phase when you record it, but they may have reversed the phase of all the bottom snare samples. The reason i say print the tracks is so you can manually line up all your sine waves because of the latency mismatch :) put as many plugs on each track as you want in that case, then line them up, just close attention to where the transient point actually starts and PEAKS, different mics also respond differently, so the peaks may not be in the same place.

You've got a few different buffers in PT, one is the hardware buffer pre renders audio generated by RTAS plugins before you hear playback, turn the hw buffer up and the amount of time between when you press play and when sound comes out increases, this goes the same for tracking, you want to have a low hw buffer otherwise you'll literally be playing to a different beat. where as if you print your tracks, and keep your buffer low, you'll be playing to a less latent track (which you should nudge to the grid) but you'll still be playing to what you're hearing.

i'm not entirely sure HOW it buffers, but i know RTAS or native relies heavily on the system ram which is first dedicated to basic functions like OS handling (you wouldn't be able to run pro tools if your ram was handling audio first). Anyways if you have a low buffer that means your processor has to do more realtime (or close to realtime) work. No matter what, the plugin will have latency because it's not like analog gear where everything happens instantaneously (even then i've heard of some analog gear being latent as well), the plugin has to calculate how to process the audio based on its code architecture, parameters set by you let alone your interface has to convert the analog signal to digital first, then back to analog so you can monitor it.

So back to printing. you cut out processing calculation by doing so, making it more or less one big A/D and D/A conversion which is faster and far less system intensive that plugins likewise, you'll be able to turn down the HW buffer a lot lower. with that you can look at the sine waves and manually delay compensate if needed. Even with pro tools HD the plugins have delay, it's just less with TDM plugs for a lot of reasons but i think you get it by now.

ideally, print your drums from superior drummer line them up (no plugins) record to that, then mix after everything's tracked. If you can track your music raw like that and get all your edits done in LE, i'd say spend a couple hundred bucks and rent a room with HD and mix it/ print stems. Mixing in LE is REALLY irksome if you want it to sound good and stay in phase

[QUOTE=Xomblies;18167413]quite intuative, the bottom snare IS out of phase when you record it, but they may have reversed the phase of all the bottom snare samples.[/quote]
I'll check, never had any out-of-phase sounds on the snare though :)

[quote]The reason i say print the tracks is so you can manually line up all your sine waves because of the latency mismatch :) put as many plugs on each track as you want in that case, then line them up, just close attention to where the transient point actually starts and PEAKS, different mics also respond differently, so the peaks may not be in the same place. [/quote]
But then wouldn't you still get heavy latency on the final bounce if you went mad with plugs, without the luxury of being able to see the mismatched waveforms?

[quote]You've got a few different buffers in PT, one is the hardware buffer pre renders audio generated by RTAS plugins before you hear playback, turn the hw buffer up and the amount of time between when you press play and when sound comes out increases, this goes the same for tracking, you want to have a low hw buffer otherwise you'll literally be playing to a different beat. where as if you print your tracks, and keep your buffer low, you'll be playing to a less latent track (which you should nudge to the grid) but you'll still be playing to what you're hearing.[/quote]
Yeah that makes sense, I'd got into the habit of knocking the HW buffer down when I recorded but I didn't know why I did it, I'd just read it somewhere and forgot why lol.

Regarding the rest, thanks a lot for the explanation, I appreciate it (p.s., make sure you know to differentiate between sine waves and waveforms; sine waves being energy generally at a single frequency - not to be a know it all, clearly haha).

Yeah I always track everything before mixing, just out of habit really. I've got access to HD at uni (graduated with a first in July but going back to do a masters) so I tend to work in there, but I prefer my own plugs :-\ will just have to be careful

[QUOTE=EmbraceRandom;18167467]I'll check, never had any out-of-phase sounds on the snare though :)


But then wouldn't you still get heavy latency on the final bounce if you went mad with plugs, without the luxury of being able to see the mismatched waveforms?


Yeah that makes sense, I'd got into the habit of knocking the HW buffer down when I recorded but I didn't know why I did it, I'd just read it somewhere and forgot why lol.

Regarding the rest, thanks a lot for the explanation, I appreciate it (p.s., make sure you know to differentiate between sine waves and waveforms; sine waves being energy generally at a single frequency - not to be a know it all, clearly haha).

Yeah I always track everything before mixing, just out of habit really. I've got access to HD at uni (graduated with a first in July but going back to do a masters) so I tend to work in there, but I prefer my own plugs :-\ will just have to be careful[/QUOTE]

i didn't think the snare sounded out of phase either

try bouncing out into two separate tracks or playlists, one with all the plugs enabled and then without them enabled (be careful don't do bypass as it still creates latency hold ctrl and windows key or ctrl and apple key and left click a plugin to disable) you can then line up the wave forms, they'll look different but as long as you're not SMASHING the audio with compressors they should look very similar

yeah yeah sine wave, wave form as long as you understand what i'm talking about you can excuse my lack of sleep/ working a lot with electro lately and just know that i'm only talking about waveforms, sine waves are an entirely different fuck in the butt.

I'm sure whichever plugs you're using have a TDM version as well, even if it's a demo it's better than using native. I'm not going to tell you what i think about the issue, however if i were to i'd say: in the case that you already purchased the native back and are now broke, i wouldn't judge you if you say torrented the TDM version... just make sure you wipe em off the comp you're using when you're done ;P

i'm curious to hear what your recordings turn out like, i've yet to make any sort of production with superior drummer as i just got it a month ago and i quit my last band, the new one is still in the writing phase :/

Yeah, I'd just tried that! The waveforms are only subtly different, which I guess means there isn't that much latency unless you apply a process to live audio, i.e. trying to process SD on the fly rather than tracking the separate parts first.

The main problem I'm gonna have is applying one of these Izotope plugs, if they are known to be latent and/or CPU-intensive. Say if I applied Alloy (if I get it, starting to think I'd regret it if I didn't) to a guitar track and radically changed the sound through dynamics processing, EQ and possibly even simulated tape saturation, the resultant would be very different to the original, so even if I internally bounced it onto a new audio track, I wouldn't, hypothetically, be able to match the waveforms.
But I guess that's where just using your ears comes in more than anything; if it sounds right, it is right. Eyes can be deceiving.

Haha yeah I know man, didn't wanna sound like a twat but was just making sure.

Lol yeah I could do that, they have got all the Waves shit to be fair, I'm just not a big fan of Waves, mainly for their update plan and also that there stuff is way too overpriced, they're not that much better than some of the stuff I use. I dare say some of my cheaper plugs are better than waves stuff.
Although one of their reverbs is fit, can't remember which one though..

How come you quit your band? I remember you linked me to them, was pretty good.
I'll upload some of my stuff once I've done it, the only thing I've done recently was To Bid You Farewell - Opeth, but it's not my best by a mile: had to rush the distorted guitars and hated the amp I had at the time, and I minimalised the drum kit a bit too much, as in I made it sound much thinner than I probably should have.

What I love about some of the new Waves stuff (specifically the SSL stuff, CLA stuff etc) is there's either no latency or about 1 sample (SSL Channel has 1).

So often I don't even have to worry about it if I just use the SSL on everything (mainly EQ), and anything else just use a plug with zero latency.

SSL channel is one of my favorite plugins, i use it on almost every track... almost as if i was on an ssl board lol

it record with my webcam its pretty sic

that's better than presonus preamps

[quote=Xomblies;18167744]SSL channel is one of my favorite plugins, i use it on almost every track... almost as if i was on an ssl board lol[/quote]
Lmao that's what I do.

It's such a crock of shit but it does feel like you're the closest you're ever gonna get to an SSL.

whats the best program to use? i use audacity

Did you guys get the full bundle or just the SSL-E?

[quote=Xomblies;18167289]This generally IS true... the "most primitive and vital functions" are what i'm talking about coming before audio handling. Do you program drivers for OSX or microsoft? how do you know priority the AI for the OS vs DAW?[/quote]

I was in a group that designed a dynamically-controlled EQ on a fixed-point DSP processor in school. In the research for that, we had to look into determining which features should be handled by the interrupt cycles on the chip. In researching for that, we came across the ASIO spec. I don't remember much about the details, but at a high-level of abstraction I did take away that this was one of the key considerations. We did a little bit of digging to see about CoreAudio (we were curious) but found nothing that gave hard technical data. However, based on speculation (not just ours, but we talked about it in Office Hours briefly as well) that most low-latency audio drivers would [I]need[/I] to do this to behave without glitchy behavior.


Anyway, that "most primitive" stuff, for the most part, is negligible considering the calculating speeds of modern processors. It's things like making sure the screen is refreshed and checking overflow status of buffers. Things that would generally result in a computer turning into an expensive paper-weight if they didn't happen. The vast majority of OS-based "behind the scenes" operations don't actively work in the interrupt cycles.


[quote=Xomblies;18167289]BTW, What are you trying to do here man? I'm trying to help people keep from making mistakes that add up to poor quality and you're trying to counter what i'm saying with some extremely subjective situation (more instances than this). No "native" system is going to compare to a pci-e soundcard with it's own ram (TDM) no matter how "well-designed" the drivers are. I know there's always an extreme situation where "it could work". Your contributions would be more enriching if you avoided those gray area situations.[/quote]

Raw pedagogy, basically. More pertinently, you're giving decent advice based on bad information, which in itself is bad. There's no gray area here to me: native processing does not suffer from quality issues (in fact, you might be able to argue the converse point based on fixed/floating-point accuracy).

The RAM issue is also, for the most part, pretty negligible for most pure effects, since they generally aren't holding onto a lot of past data. Obviously there will be exceptions to this (e.g., anything that uses convolution). This also can't be stated for synthesizers that use large wave-tables or lots of samples, since the RAM needs to hold onto a lot of raw data for quick access. The rudimentary algorithms themselves (even the big-name "emulations"), however, are generally on the scale of kilobytes.

I'll concede that there is a calculable difference in latency, but since modern processors run through cycles on the order of nanoseconds, you tend to have a lot of cycles of processing before you need to pass the next audio sample.

It's entirely possible that you think an Pro Tools|HD system is a requisite for good performance. But the stated reason (that OS processes happen before Audio processes) doesn't happen to be a good reason for that by virtue of the fact that for the most part it's incorrect.

whats the best program does any1 know?

try reaper

posted on wrong thread derp


The worst offense I see for every person with their new home studio is the crap they put on walls for 'acoustic treatment" whether it be egg cartons or that equally crappy thin auralex stuff for the walls. i'm just putting money away right now for when i get to my new place for grad school to actually treat the walls and then invest in a higher in recording studio. right now i'm pretty much in a square room and I can't hear worth of shit from my speakers, it's awful, but when i get a better place, i may try and get some better speakers and such.

as for buying software, the only thing i see the use for is customer support if you ever need it, and that's only for a few items. I'll never by east west soundsonline stuff because i hate the company and how restrictive it is, if I could pirate it easier, I would.

Yea egg cartons don't do shit

Also alot of that insulation stuff can be very overpriced

also completely inadequate because it's usually too thin for anything that isn't in the higher part of the frequency spectrum. a lot of those acoustic items can be made at home for a significantly lower price if you look online for DIY bass traps and absorption stuff. unless you're in a room of larger size, you don't even need sound diffusion either, you can solve most problems via absorption

also dicks

josh get on aim douche

Egg cartons are ok for diffusion, but that's usually not the biggest problem for home recordings dudes.