@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.

i think it's on the high end of things to consider, i mean, how can you record, mix, and master what you cannot even hear if your room is poorly treated or not at all? putting egg cartons on the wall results on a dead high and midrange and an acoustically lifeless room, the exact opposite of what you want if you're doing any sort of recording

I think you're confusing how you'd treat a live room and how you'd treat the control room :/ (or the matter of taste in treating a live room)

In the control room you want the least refelections possible and diffusion helps by breaking up the high end reflections and reducing flutter echoes. Something you definitely want if you're mixing, and also if you want to record in a small room with loud things like drums.

But a lot of live rooms are intentionally dead for a reason. So you can add your own 'life' later. Some rooms are the opposite. Some of the best drums recordings are a good kit in a big wooden room.

So this:

[QUOTE]i think it's on the high end of things to consider, i mean, how can you record, mix, and master what you cannot even hear if your room is poorly treated or not at all? putting egg cartons on the wall results on a dead high and midrange and an acoustically lifeless room, the exact opposite of what you want if you're doing any sort of recording[/QUOTE]
is kinda wrong. It depends on what you want.

Egg cartons are just diffusers, splaying reflections to help parallell walls not have endless echoes back and forth. Bookshelfs with books do the same.

A control/mix room needs to be as dead as possible, it's really definition of treating the room, without getting into intense mathematics.

A dead control room or mix room is the wrong way to do things, you want the room to sound as natural as possible as in this is is an accurate representation of the sound wave and its frequencies. A dead room is an unnatural sound that would only be acceptable in an incredibly small studio or a vocal booth, where you would add reverb and ambiance later.

[quote=Convectuoso;18170492]In the control room you want the least refelections possible and diffusion helps by breaking up the high end reflections and reducing flutter echoes. Something you definitely want if you're mixing, and also if you want to record in a small room with loud things like drums.

...

A control/mix room needs to be as dead as possible, it's really definition of treating the room, without getting into intense mathematics.[/quote]


[quote=JoshIsNumber3;18170565]A dead control room or mix room is the wrong way to do things, you want the room to sound as natural as possible as in this is is an accurate representation of the sound wave and its frequencies. A dead room is an unnatural sound that would only be acceptable in an incredibly small studio or a vocal booth, where you would add reverb and ambiance later.[/quote]


These are both equally valid perspectives, and in general the contemporary idea of what is best tends to switch back and forth with the trends of the times. I believe the current trend is for a more natural sounding control room. I think there's something to be said about considering both in a home environment. Again, in general, it's easier to make a room dead-sounding than it is to make it live-sounding in a good way (especially if the room is small).


[quote=Convectuoso;18170492]Egg cartons are just diffusers, splaying reflections to help parallell walls not have endless echoes back and forth. Bookshelfs with books do the same.[/quote]


One other thing to keep in mind is how much better a bookshelf will perform. Egg Cartons tend not to have much effect because they have very little mass, which is one of the only things that can really alter how sound waves propogate (vacuums, resonators and wave guides are the only others I can think of, and they're all way harder to work with than pure mass).

Yeah on reflection and research, egg cartons are on the verge of pointless.


But lol idk how they even came up.


I dunno. With the amount of guesswork that goes with a live sounding room, I'd rather have a roomI only really have to worry about the response of the speakers.

IMO.

Room + monitor speakers (and all variables of these; position etc.) is the most important part of mixing. You could have Pro Tools HD and all the best gear, but if you can't monitor properly, it's all wasted.

I have learnt how to compensate when mixing with my AKG K701s, BUT i always check my mixes in the University studio (which is well-treated) and all other playback systems I can think of. I know of people who can mix just as well on headphones as they can on monitors, as they know they're headphones well enough to compensate for the generally poor low frequency response and generally poor accuracy of spatial qualities. I'm not at that point yet, but I'm getting there. Like I say, I always check back on good monitors in a good room before completing any projects.

[quote=Convectuoso;18171005]Yeah on reflection and research, egg cartons are on the verge of pointless.


But lol idk how they even came up.


I dunno. With the amount of guesswork that goes with a live sounding room, I'd rather have a roomI only really have to worry about the response of the speakers.

IMO.[/quote]

One thing I've always been curious about, but haven't ever tried (too lazy, lack of opportunity, etc) would be taking egg cartons and filling them with some kind of cheap putty or concrete or something similar. Because the shape does suggest a decent amount of diffusion/dispersion if there was any mass to fill it.

It'd still result in an uneven room. Egg cartons cannot 'control' low frequencies or even the low-mids, where most fundamentals lie

[QUOTE=Moseph;18168733]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.




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.[/QUOTE]


which is why people still have latency issues with non TDM based computers? whatever reasearch you did in college you get an F. for both not citing your sources for such retarded claims and also for sucking dick at engineering. You're only further proving my point that you sound like you know what you're talking about, but don't know what the shit you're doing

[QUOTE=EmbraceRandom;18171187]Room + monitor speakers (and all variables of these; position etc.) is the most important part of mixing. You could have Pro Tools HD and all the best gear, but if you can't monitor properly, it's all wasted.

I have learnt how to compensate when mixing with my AKG K701s, BUT i always check my mixes in the University studio (which is well-treated) and all other playback systems I can think of. I know of people who can mix just as well on headphones as they can on monitors, as they know they're headphones well enough to compensate for the generally poor low frequency response and generally poor accuracy of spatial qualities. I'm not at that point yet, but I'm getting there. Like I say, I always check back on good monitors in a good room before completing any projects.[/QUOTE]


for low end you need a big room with some foam with a good NRC rating in the lower frequencies, baffles, ideally if you can get walls within walls with a little bit of space between them (both sound treated) would be awesome

[QUOTE=EmbraceRandom;18171187]Room + monitor speakers (and all variables of these; position etc.) is the most important part of mixing. You could have Pro Tools HD and all the best gear, but if you can't monitor properly, it's all wasted.

I have learnt how to compensate when mixing with my AKG K701s, BUT i always check my mixes in the University studio (which is well-treated) and all other playback systems I can think of. I know of people who can mix just as well on headphones as they can on monitors, as they know they're headphones well enough to compensate for the generally poor low frequency response and generally poor accuracy of spatial qualities. I'm not at that point yet, but I'm getting there. Like I say, I always check back on good monitors in a good room before completing any projects.[/QUOTE]

this is so right i don't even know what to say!

Xomblies

sup

wat is tdm

Time division multiplexing.

I guess I'll google it

Lol it's just kinda the architecture and the bus for audio in Pro Tools HD.

[quote=Xomblies;18174647]which is why people still have latency issues with non TDM based computers? whatever reasearch you did in college you get an F. for both not citing your sources for such retarded claims and also for sucking dick at engineering. You're only further proving my point that you sound like you know what you're talking about, but don't know what the shit you're doing.[/quote]

Latency exists in TDM systems as well. It's just generally very small. A/D/A conversion is a time-consuming thing (it's the inherent nature of needing to pace out your samples on the order of kHz, you can't just keep throwing processing at it because you still have to wait around for a temporal signal in the analog domain).

As for citing the exact sources, I mentioned the ASIO spec. That's a pretty well-known reference to consider. Admittedly the second-half was about conjecture, but it was the conjecture of somebody who's spent a few decades getting paid to educate for the EE department of an accredited university. So forgive me if that's not a good enough foundation to justify listening to his opinion.

In case you're not following, go here:

[url]http://www.google.com/url?sa=t&source=web&cd=1&ved=0CBIQFjAA&url=http%3A%2F%2Fstatic.helge.net%2F2010%2F06%2FASIO%2520SDK%25202.pdf&rct=j&q=ASIO%20steinberg%20spec&ei=fZuLTOS1M4SglAeJwplh&usg=AFQjCNH7C0L9kHFxA4kO58SDnjTgYp85Pg&sig2=te__hpv09-z24Rgl5-elNQ&cad=rja[/url]

Start reading at Page 42. Turns out that maybe I was wrong for Windows based systems (looks like it takes advantage of threading, which might actually make my assertion correct for the wrong reasons, but I don't know enough about Windows threading to speculate comfortably), but correct for OSX. Specifically, the buffers for I/O appear to accessed during the interrupt cycles on OSX.

Also to the point, there are people who can get negligible latency (on the order of 2 msec or less) out of native processing systems (I have been one of them). You just need to have sufficiently quick algorithms. Most "pure-effect" algorithms fall into this category (EQ, dynamics, your basic temporal effects).

Now, we're admittedly getting outside the scope of my focused studies (I was concerning myself with the operations of a TI CM55X chipset, and the inner-workings of a particular OS are more Computer Science than Signal Processing). But I'm still willing to wager that what's happening to move along the OS-sensitive operations in interrupt cycles is pretty minimal. That's kind of the point of the interrupt cycles: only the things that will break if they [I]don't[/I] happen get used there. But again, I'm still willing to wager that bussing a few bits here and there and clocking your hardware (the bare-bones stuff an OS is supposed to do) doesn't exactly tax a modern system into a catatonic state.

It's entirely possible I'm wrong, but you saying so means precisely dick to prove it (that's no better what you said I wasn't allowed to do myself). So either bring something to the table to show me otherwise, or let it go.

I'm going to repeat it again: it's entirely possible that you think a TDM system is a necessity. Saying it's necessary because the OS functions will always take precedent over the audio of a native processing system is not a valid reason because it's not particularly true.

i'll admit you can get low latencies with rtas vst etc, but unless you've got some retardedly fast processor, you'll overload your shit, tdm allows dedicated resources to JUST plugins/ ADA (basically most of pro tools' functions) at the lowest latency possible. YES, it's old tech, but for streaming audio it's still a lot better. That AISO is also used in pro tools bud, except with HD you've got an extra resource to handle the multi threading: TDM CARDS. As of late they've got 8 and 12 core macs that you can assign cores to PT and let the remaining handle system tasks, which are also more important than you think since you're going to be constantly accessing large amounts data off hard drives, building waveforms constantly and if you're editing; copying and pasting. Not to mention if you're using sample based instruments that's even MORE information to read off of other hard disks.

so yeah, if your computer is just chillin or you're using LFO's or something not sample based... sure, AISO and system ram are fine... but if you've got 10-16 drum tracks, 4-6 guitar tracks, 2 bass tracks and 8 vocal tracks, even at 44.1 and a 2 min song, are still going to be 30 megs a piece. i'll go with the standard 24 tracks that's 720 megs of audio to access just off of your session drive. if you're doing edits, or not working with consolidated tracks your computer NOW has to play a lot more files randomly. so moseph you can do the research if you want and waste your time or you can just TRUST someone who knows how to do audio like kfc does chicken... RIGHT. I've obviously got the chops to back what i'm sayin and i can help you out a lot if you would let me.

anyways you CAN get a decent sounding mix, recording off an LE rig or something like it, in fact i have heard tons of people do it... sorry about sounding like an asshole yesterday, lack of sleep and a passion for music don't mix when you're having one of these kinds of discussions

Let me back up for just a second here:

[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).[/quote]

What I think is getting lost in translation here is that about 85-95% of the stuff you say (from a technical standpoint) I agree with as being completely correct and proper. I generally only chime in or tack on when I think there's something to add (see comment about book cases above), or if see something that I believe is said in error. Having looked back on things I've chimed in regarding your particular advice posts, it looks like it's more frequently the latter case. That may be why you could get the idea that I'm beating up on you or something (you already know the difference in how I post when I'm actually beating up on you). That's definitely not the case. I know you're on the higher-end of the curve with regards to this stuff (your posts show that much), but I think we can both agree that neither one of us is on par with guys like Massenburg, Blumlein, Matthews, or even Frankle. At least not in their respective fields.

One of the goals of this forum (and most I would join), so far as I'm aware, is mutual [I]learning[/I]. This requires that not only questions get asked, but things get said in response. This isn't a forum that caters to experts, so a lot of stuff is bound to get misrepresented (case in point: my comment about ASIO as it applies to Windows). I'm more than willing to admit when I'm wrong, but I'm at least gonna want some evidence/rationale showing me why: giving me the finger and taking a dump on my lack of talent isn't gonna get anywhere.

Now, getting to the matter at hand...

Oddly enough, the most pertinent part is at the end:

[quote=Xomblies;18176160]so moseph you can do the research if you want and waste your time or you can just TRUST someone who knows how to do audio like kfc does chicken... RIGHT.[/quote]

I could, but that's the opposite of learning. Don't give us a fish, teach us how to fish, is what I'm getting at. You generally do alright in this respect, but there is that last 10% or so of stuff that needs some validation. That's not weird, and it's actually a good thing. If you're not stretching, you're not learning either.

Also, the KFC analogy might not be the best. It's kind of the Domino's of chicken...


[quote=Xomblies;18176160]I've obviously got the chops to back what i'm sayin and i can help you out a lot if you would let me.[/quote]


Here's a big pet peeve of mine, actually. Talent and good ears do not necessarily have any bearing on technical understanding. I'm not going to get into the specifics of how good I think I am, or how good you think I am, or any of that sort (clearly the consensus is known about all that).

Let me make an example. Once upon a time in the Jam Session you recommended the Heil PR-20 for snare drum over a Shure SM57. If I had made that same recommendation, but I said not to do it because the SM57 needs phantom power, whereas the PR-20 does not, what would your reaction be?

That's how I saw the OS-specific priorities thing. You contrasted audio streaming as a whole (rather than effects processing specifically) with OS-specific operations. That not only may not be pertinent to the issue at hand but it didn't agree with my particular understanding of reality. Hence the comment.

My old studio tech job was predominantly training people to use our facilities and acting as tech support when they hit problems (read into that what you want). When you decide you're gonna share knowledge like that, you sort of need to be able to have a basis to work off of, otherwise people aren't understanding what they did wrong in the first place.


[quote=Xomblies;18176160]i'll admit you can get low latencies with rtas vst etc, but unless you've got some retardedly fast processor, you'll overload your shit, tdm allows dedicated resources to JUST plugins/ ADA (basically most of pro tools' functions) at the lowest latency possible. YES, it's old tech, but for streaming audio it's still a lot better.[/quote]


You'll note that I never disputed it wasn't better. I said it wasn't better [I]because of OS priorities[/I] (the dispute was about prioritization). Yeah, the system is more efficient, it's throwing a lot more hardware at the problem. That's why they had PT systems up and running audio in like 1994, when the next best competitors were struggling to get decent MIDI synthesis.

Side-note: even though the definition of "retardedly fast" tends to change over time, I don't think you'd need one to do what you're talking about if your system is well put together. Modern systems are [I]fast[/I]. I've successfully done 18 channels (about 60% of what you're talking about) using a 1.8 GHz Pentium M, with only 1 GB of RAM. My current rig does 26 and is about 90% stable (I chalk that up to Presonus Drivers and Sonar not cooperating more than the system itself).


[quote=Xomblies;18176160]so yeah, if your computer is just chillin or you're using LFO's or something not sample based... [/quote]


You should also note that I conceded that point in a past post as well. But I'm also of the mentality that you shouldn't assume a user is gonna be using huge wave table-/sample-based synths unless that's mentioned.


[quote=Xomblies;18176160]you can assign cores to PT and let the remaining handle system tasks, which are also more important than you think since you're going to be constantly accessing large amounts data off hard drives, building waveforms constantly and if you're editing; copying and pasting. Not to mention if you're using sample based instruments that's even MORE information to read off of other hard disks.
...
i'll go with the standard 24 tracks that's 720 megs of audio to access just off of your session drive. if you're doing edits, or not working with consolidated tracks your computer NOW has to play a lot more files randomly.[/quote]


Except that's not exactly true is it? You're not streaming straight from the hard disk drives, you're streaming from buffers (the driver buffers) that are filled from data on the drive. I realize it seems like I'm nitpicking, but that's only because I am. Again, this is a learning thing here. So far as I know, the RAM doesn't need to be out the wazoo unless your edits and samples are off the charts, because the computer can generally jump to another page/cache faster than the buffers can unload at 441000 samples/sec.

[quote=Moseph;18176213]So far as I know, the RAM doesn't need to be out the wazoo unless your edits and samples are off the charts, because the computer can generally jump to another page/cache faster than the buffers can unload at 441000 samples/sec.[/quote]
Lol full drum kit Beat Detective at 96khz with plug ins on everything, man I need to try that.

Convection beats