This thing called Latency

[StrykeBack]

I'm using an Aardvark q10 with Cubase SX2 on a 2.1 AMD Asus
I was tracking a set of drums with 5 tracks going and on play back I kept gettin pops and crackles and was looking everywhere to find any peaks and couldn't find them. My last bet was changing my audio buffer from 256 to 1048 now my latency has jumped from 5 ms to over 25 in the aardvark interface and in cubase it says 51ms input and 23 ms output latency. Now are these huge deals when recording live audio such as bands or how is the best way to counteract this? Keep the audio buffer down when recording and just boost it up for mixing when it gets worse as you had more vsts and effects?
I see there is a latency compensation in cubase sx2 but i'm not exactly sure how to use it.

Whats the best way to keep latency as low as possible?

I'll try and post this in a couple forums to get as many responses as possible
thanks
Daniel

[StrykeBack]

anybody?

[djui5]

You shouldn't be getting that much latency with a pci card interface. You shouldn't be getting much at all...maybe a couple of ms or so at the most. I'd call Ardvark and see what they say. I've never had a problem like that except with the M-box which is a USB device..so I know the source. Sorry I can't help you further

[xstatic]

With a higher buffer setting your latency will definatley increase. This should not be a problem during mixdown phase, but if you are tracking and monitoring your channels through your program, than the latency will definately be there and depending on the musician, will definately be a problem. If you have to monitor your live tracks through the program, avoid all inserts until after the track has been laid. Latency compensation should not help you during tracking. The way I understand it is that every different plugin has its own different latency. These latency's run from extremely small (completely unnoticable) to pretty hefty. When you apply a very intense plugin to a track, its output is delayed slightly due to the amount of necessary real time calculations (that is if you are using that effect in real time, if you process it in place than their should never be a latency). This means that your different tracks will all have slightly different output times when using real time plug ins and will all be slightly offset from each other. Latency compensation means that the outputs of all tracks are delayed just slightly so that when their audio signal is finally outputted they all play back with the same location in time that we expect. Typically even with very hungry plugins this should result in a latency compensation of less than 1/4 of a second and most likely only 5 to 30 ms which really isn't that noticable from the time you push play till the time you hear it.

Some sound cards have what they call a 0 latency monitoring system. Once again if I understand correctly, this means that they have a basic submixer built into them. When you monitor in real time through your soundcards facilities rather than your Sequencer's (in your case Cubase SX) they output your audio signal BEFORE it has processed through your sequencer and I believe in some cases stright of the PCI card itself without ever having passed through Windows or whatever OS you are using. This is a cool feature, but will not allow you to take advantage of monitoring with plugins on the tracks while you are tracking. Basically there is no way around the latency (however small or large it may be) of your sequencer and/or your OS. While you are recording your tracks you create your monitor (headphone) mix from your DSP mixer or your outboard analog mixer then there should be no problem. After you have laid a track you just play it back to listen to it through your Sequencer. This way you will be able to hear all playback with some processing.

I could be wrong about any or all of this, but it is my understanding of how DAW recording works. Digital is a beautiful thing, but no where near as simple as analog:D (wait till you get into master and slave settings, clock issues etc...)

[djui5]

nice post xstatic. I learned something today...thanks

[StrykeBack]

Thanks guys for your replies. Those are very good points. I am aware of the direct monitoring through my interface and I can't be sure whether it is setup because some of the options in Cubase are opposite...says check to make it work and it only works if unchecked...go figure.

Someone mentioned dling the new drivers which just came out a week ago so I got those installed and with a couple tracks going and a few plugin inserts I've got my audio buffer set at 2 miliseconds with 256 buffer size and the turbo mode on....
However in Cubasesx2' setup window it says input is 3.3 ms
and output is 5 ms
I know that Cubase offers latency compensation but I haven't caught on to how to use it....
Would I tell cubase to delay recording by 3.3 ms to compensate for the input latency or what exactly?

I'm hoping the crackling and my cpu usage going up so high was do to a bad driver, I'll check with aardvark.

Thanks for your guys comments
Daniel

[xstatic]

I always leave my latency compensation on. Once again if I understand it correctly, you should not need to worry about changing any delay times. I believe that when you record a track that it lines up just fine and that the only time that latency becomes the issue is for metering, and playback. Also, cubase is calculating the latency not for Cubase itself, but for your actual soundcard so personally, I would not make any changes. I have noticed that latencies tend to climb really fast if you are using plugins on your master section, if you are running maximizers (Waves L1, L2 etc...), multiband compressors, heavy Reverbs (waves reverbs will drain your CPU fast, as well as any convolution stuff), or if you are running group channels that have processing on them. If at all possible, avoid running any types of plugins until the bulk of your tracks are laid down. At least if you are inputting more than two tracks at a time. If you are recording at higher bit and sample rates I believe you will run out of room a lot quicker as well.

I do however think that in your case (unless you are tracking at 24/96) that something else is going on as well. I record in Nuendo 2 at 24/44 and have projects open with 32+tracks, groups, and maybe 100 plugin and EQ filters open at once. My buffers are at 512 for something like a total of 6ms delay I believe. I have not had any problems with clicks or pops. It is also possible that you have a clock mismatch between Cubase and your soundcard. Make sure only one is set to master, and that you are using the same bit and sample rates at both your soundcard and Cubase. I had that problem once when adding an external HD recorder to my rig. I was normally using Nuendo as my master, but when I hooked up the HD recorder to do a transfer I forgot to change the recorder to slave to Nuendo's clock (or I could have changed Nuendo to slave to the recorders clock). Everything seemed fine until you listened really closely to tracks that were transfered. About every 6.5 seconds there was a small fast "chirping" sound. It was not evident in the drum or vocal tracks, but was definately noticable on the bass tracks, and a little on the guitars. We had heard it before, but it had really sounded like a string squeak. When I tried to isolate it so we could try and solve the problem, that's when we noticed that there was a pattern. I am assuming the two clocks were just a tad different and we were hearing the result of a reclock or dropped samples or something like that. Hope this helps, and good luck:)

[StrykeBack]

Thanks xstatic I'm still not sure on my cpu usage with 14 tracks and about 3 vsts on each...some I applied to the track to save some cpu. I wondered why I couldn't get the aardvark to change resample rates inside of cubase but the "refuse sample rate clock from outside asio source" was checked so i unchecked it and cubase now controls that...cubase' clock was set to internal but I didn't see one for the aardvark, all should be setup right there.

As far as the latency compensation, on the preferences tab it says what the latency compensation should be and it is set to 0.0ms ?? When clicked on and glowing orage or not clicked on...still confused about that

But you have been very helpful. I'll be doing some tracking with a vocalist and some guitar and keys work...I'll have to see what I come across as far as pops and clicks.
I wasn't aware that the higher sampling rate would make that much of a difference on cpu.
I figured I'd go for highest quality and setup my latest recordings for 96k at 32 floating point..or is that not worth it and should I just stay at 24 bit?

[Clueless]

Originally posted by xstatic
I always leave my latency compensation on. [...] I record in Nuendo 2 [...]

There you have it, folks. The solution to the latency problem is to use DAW software that actually compensates correctly for it. I F@#%ng cannot believe how lame DP and other software is for not supporting this feature.

Yes, I am a frustrated DP user, and am planning to upgrade to Nuendo when the other problems holding me back (UAD-1/Mac G5 OSX) are resonably sorted--perhaps by May. Please tell me that there are not other demons lurking in Nuendo that will dash my digital hopes yet again!

[StrykeBack]

well my guitarist friend and i laid down a few trackings in Cubase sx2 today {with this so called delay compensation turned on} and it looks as if there is a slight delay...according to my aardvark and cubase drivers together about 5 ms...so unfortunately it looks like it doesn't set the compensation automaticaly and therefore I have to figure it out and punch it in... ohwell, I'll figure it out...I've used DP3 for midi at my college's recording studio but never tested their latency issues...no need

[xstatic]

It still sounds to me like you are looking at the whole delay compensation thing in the wrong way. It isn't meant to make your delay a value of 0. It is really meant to make your outputs of all your individual audio, midi and VST's be lined up when you are running plugins etc.... Not on the inputs, but on the output signal. 5 ms is a pretty small delay between two programs.

[TheNewKid]

another question is: how is AMD holding up these days? I'm looking to buy a recording-only computer this summer, and AMD has great deals, but can it handle multitrack? Can anyone confirm that these delay issues has nothing to do with AMD? Thanks!

[StrykeBack]

the delay that i had a problem with was only a one time thing so i'm not sure what the case was but i've had no problems otherwise recording up to 6 or 7 tracks at a time using an aardvark and working with over 35 to 40 tracks in one song. My amd is a 2600+ or xp chip same thing i think but its overclocked to 2.1 gighz and runs smooth. I think some of the programs are now optimized for Intel's hyperthreading but its more of a matter of budget. If your gonna go with an AMD you can get a mid level chip for around 100 to 150 and make sure you get the nvidia chipset, preferablly an asus with the dual caching ram feature.

[djui5]

I love AMD....

I'd much rather get an AMD than an intel....and if building a computer for recording you're going to want an AMD...they rock.

Delay has nothing to do with processor speed of your computer, it's caused by the circuitry of the A/D/A and other factors such as internal bussing and plug-in's, also the more outboard gear you have in a single chain the greater the delay will be..

[xstatic]

I just upgraded my cpu this week. I moved from an AMD xp1800 to the new 64 bit AMD 3000. My old system kept up pretty well, but I am starting mixdown right now on a few larger projects where the track count got really high. I was having to lower my buffe when mixing because many of the songs are loading 36+tracks with LOTS of plugins. I was experiencing some pops and cracks during playback since my CPU was heading up to about 85% during mixdown. With the new AMD those same songs when loaded are at 35% tops now. I starting mixing another song yesterday on it and purposely used a couple of high cpu load plugs just to see how it handled. Even using convoution verbs my load still isn't at 40% yet. I even setup extra busses just for fun, still not even a hiccup. Not only is the new processor much more stable for playback (stable as in audio clicks), but i am only running 512 megs of ram instead of the 1 gig that was in my old machine (ran out of money and RAM prices are up right now). Also, this is the fourth system in a row that I have used the VIA chipset with never even 1 conflict with my RME. The new motherboard even does dynamic cooling and overclocking. It constantly changes my cooling system and clocking speeds based on the currentl needs of the system. So now my tower is faster, smoother, and quieter:smokin:

[laboroflove]

So if there's a audiosuite plugin on a track thats causes delay just process it in place and turn it off on the insert and it'yll be in time as if no plug in was there?
I've heard all audiosuite processing causes some sort of latency even processing a 1 band eq during a word of vocals.

[yeahforbes]

I have the luxury of using Pro Tools HD at work which displays the accumulated plugin latency for each channel, and I can tell you that the number of samples of latency for the 1 band EQ is in the single digits. 4 or 7 or something. Totally negligible, except perhaps for parallel processing (see related thread).

If by "turning it off" you mean making it inactive, then yes, it is essentially not there. Bypass mode, on the other hand, keeps the latency.

P.S. Be sure to distinguish AudioSuite from Real Time AudioSuite... I think you are referring to the latter.

[laboroflove]

What I was wondering was a hefty plugin like say,l1 ultramaximizer,was on a kick drum channel and I've got the setting's on the money where I'm lightly brickwall compressing it.
If I process it in place and pull the plug-in off.gone.Would I have to move the file back for it to be in time when I originally played the drum beat?

Was that 4 to 7 samples or millisecond's that you were talking about on the 1 band.?

[laboroflove]

nevermind I went back and read sample's.sry