How Big is too big ?
Hi Guys and Girls
I have a brief question. I have to adjust my DMA buffer size on my soundcard to 512 to eliminate clicks and pops when i record anything over 7 or 8 tracks on N-track. This should give me about 10 millseconds latency input/output at 48khz. Will this be a problem when I record tracks ? Will I hear a delay ? I have just implemented this solution and I am
uncertain of the consequences. If anyone has experience with this I would sure appreciate your input.
Thanks,
Ted
How much latency is too much? When you can measure it with a stopwatch… 
The consequences will be subjective. If you’re a professional musician, 10 ms may bug you alot, particularly if you’re playing VST instruments. If you’re a punter like me, a 10 ms delay is nearly unnoticible, and worst case when you’re mixing down, you can shift a track back n forth to tighten it up.
Try it out for a week or so. You can always reduce the buffer, once you’ve found ways to eliminate the last of the glitching.
Latency is totally irrelevant unless you monitor through the computer, such as when using the live button to hear plug-ins in real time. Use the buffer size that give no pops and clicks and do monitoring of whatever you are recording external to n-Tracks and you’ll have nothing to worry about…assuming things are working right.
Thank You Phoo and Archimedes 
That is exactly what I needed to know.
Ted
I’m a bit confused by the analysis that latency is irrelevant unless you monitor through your computer. I guess it’s OK if you are recording all your tracks at once. However, when laying down successive mulitple tracks I believe latency is indeed an issue. Record latency is the delay in time it takes between when a sound occurs and when its actually recorded. If one is multi-tracking, then one is neccessarily monitoring previously recorded tracks (through the computer) and “playing to” them. So, the higher the latency, the longer the delay between where you wanted to place your note/sound, and where it actually “lands” in the recording. For example, you strike a note while playing to a recorded drum beat, and you hit the note “dead-on” the beat. If your latency is high, the sound gets recorded some amount of time (the latency time) after the drum strike took place. Thus you wind up playing “behind” the beat by and amount of time equal to the latency. Is this not correct ? Whether or not the delay is acceptable, or even noticable, is indeed subjective. This can be compounded if you also have playback latency issues which is, as I recall, a separate setting. Anyway, “just say’n”…all comments and opinions welcome…
I’ve found by changing mixboards pops and clicks get eliminated… …that was my problem in the past… i had a mackie 24 track mixboard then i changed to a PA mixboard by peavey, both gave me pops and clicks… especially when i’d peak out… then… i got this 22 track behringer mixboard… comes w/a built in compressor and gate… thing is the most smooth board i’ve ever owned… just thought maybe you could try that… I don’t know much abt Latencies or anything like that… i’d like to learn, but that didn’t end up being my problem
-Reflex-
www.soundclick.com/reflexrx
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If one is multi-tracking, then one is necessarily monitoring previously recorded tracks (through the computer) and "playing to" them.
Not exactly. The previously recorded waves are not going through the computer, but being play by the computer. What goes into the input and is processed and is then play out the computer in real time is going through the computer. Latency is the time for a full real time trip. Simple playback does not make this real time trip.
When tracks are played they are queued to play in advance - buffers are filled in advance then are played on cue. The app knows how much time to offset and it compensates every track at playback so they stay in sync as previously recorded. Recording tracks know about this and the next time you play your song those tracks you just recorded will playback just like you played them. You don't have to think, "I have 50 milliseconds of latency so I need to play early."
If this kind of latency was not compensated for at record time automatically then no one would be able to lay down multiple tracks, ever. Even 2 milliseconds out of sync is noticeable if the source is right.
Some folks have problems with sync. That's not withstanding, but it is a problem.
Not exactly. The previously recorded waves are not going through the computer, but being play by the computer. What goes into the input and is processed and is then play out the computer in real time is going through the computer. Latency is the time for a full real time trip. Simple playback does not make this real time trip.
When tracks are played they are queued to play in advance - buffers are filled in advance then are played on cue. The app knows how much time to offset and it compensates every track at playback so they stay in sync as previously recorded. Recording tracks know about this and the next time you play your song those tracks you just recorded will playback just like you played them. You don't have to think, "I have 50 milliseconds of latency so I need to play early."
If this kind of latency was not compensated for at record time automatically then no one would be able to lay down multiple tracks, ever. Even 2 milliseconds out of sync is noticeable if the source is right.
Some folks have problems with sync. That's not withstanding, but it is a problem.
Yes, I realize that a computer based recording system has built-in latency by it’s nature, and that it is compensated for hardware/software. However, I did not realize that it’s always compensated for, not matter what it’s length (dma buffer setting). So then, it only becomes an issue when, as previously stated, applying real-time, computer-based effects and such ?
I’ve been finding out as much as I can on such things while tracking down my own set up “demons”. I’m using a 2496 with ASIO drivers, and I find I must increase the recording buffer size to it’s maximum allowable size to avoid dropouts in my recordings. This hasn’t been an issue since I’m merely dubbing tracks from mini disc via the analog inputs. Still, I feel as though I shouldn’t have to increase the size so extremely just to get a clean recording. But I guess that’s a topic for another thread. Thanks for the enlightenment…
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