Is it possible to totally eliminate latency?

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JJrecording

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I use Reaper on my perfectly fine Asus laptop with the Alesis io2 as my input box for guitar, mics, exc.. The Alesis uses USB to connect to my computer. And yes, I use the Asio driver. It works just fine, i have tweeked the driver the best i can. But there still seems to be a very small amount of latency, 1ms to 3ms, maybe less...idk, like I said its very small. Is it possible to eliminate this? if so, how? or am i being to picky? because I have read that low latency is what you shoot for. No latency is impossible..but i dont know if that is true.
 
Most people would say you're being too picky. I say if you hear it and it affects you then you probably have a more precise sense of timing than most people.

You can reduce latency to the sub-microsecond level by going to an all-analog monitoring path. Some 2-channel interfaces have that built in; a simple knob on the interface is a good indicator. Otherwise you need an analog mixer with sufficient quality and features.
 
sound moves at 1ft per millisecond, so 1-3ms is the difference between 1ft and 3ft from the speaker.
you are NOT humanly going to be able to tell the difference.

In blind medical studies MOST humans can't percieve below 20-30ms. Cranking it down below what
you can actually percieve only stresses the cpu more and is counter-productive. Anything below
10ms should be fine.

Dont chase numbers, it's not a peeing match....
 
sound moves at 1ft per millisecond, so 1-3ms is the difference between 1ft and 3ft from the speaker.
you are NOT humanly going to be able to tell the difference.

In blind medical studies MOST humans can't percieve below 20-30ms. Cranking it down below what
you can actually percieve only stresses the cpu more and is counter-productive. Anything below
10ms should be fine.

Dont chase numbers, it's not a peeing match....

There are different kinds of studies, some of which show sensitivity to intervals of significantly less than 1ms.

If he's just looking at the numbers and not going by what he hears then I agree with you. If he hears a problem regardless of the numbers then the latency might actually matter.

A while back I was tracking bass to a looped drum part (which had dead perfect timing). It sounded great during tracking, but during playback I thought the feel had changed in some way. I was about to write it off as my imperfect perception until I looked at the bass player who also seemed to be perplexed. We realized we both heard it. I started testing things in the system and found that the inputs were being recorded 46 (out of 48/ms) samples late. That's just under 1ms off. I adjusted the record offset control in the DAW and everything locked in perfectly.
 
I am not chasing numbers haha, that was a funny analogy though. I honestly can hear that little bit of latency during tracking and playback though. Its all good though. If it is not fixable, then I can deal with it because i can always fix that little tiny bit of latency by lining it up by ear with the tracks recorded previously. I just want it to be as perfect as possible, thats all.
 
Don't confuse latency (the time it takes your playing to get through the system back to your ears during tracking) and record offset (the audio being recorded ahead or behind the existing tracks).

You can check for record offset by recording the output of your interface connected to an input and comparing the two tracks by eye and ear.
 
Most people would say you're being too picky. I say if you hear it and it affects you then you probably have a more precise sense of timing than most people.

You can reduce latency to the sub-microsecond level by going to an all-analog monitoring path. Some 2-channel interfaces have that built in; a simple knob on the interface is a good indicator. Otherwise you need an analog mixer with sufficient quality and features.

Thank you for your advise. On the Alesis io2 interface there is a knob that gives you the option of running it Direct or USB. However, it is going to connect through the USB chord no matter what, because that is the only option for connecting it to a computer. Now my question is, which one should I use? because i have been turning the knob all the way to USB. By the way, when i first got the Alesis I gave the Direct option a try but it didnt sound right so I have used the USB option ever since and it sounds perfectly fine..with that very small amount of latency though. Maybe i should fiddle with the Direct option more.. Idk. As you can probably tell, I don't know much about the difference in the two options. reading the manual on the Alesis io2 would probably help.
 
Perhaps when you use the direct monitoring you're still hearing the inputs through the USB path as well. That could make the inputs sound bad. You probably just need to turn off input monitoring in your recording software.
 
Don't confuse latency (the time it takes your playing to get through the system back to your ears during tracking) and record offset (the audio being recorded ahead or behind the existing tracks).

You can check for record offset by recording the output of your interface connected to an input and comparing the two tracks by eye and ear.

Ok that gave me some insight to a bit of my problem. Thanks, now i know the difference in record offset and latency. Although I have a small bit of latency when tracking, my main issue is the record offset. So how would I fix that in Reaper or with my Asio driver? because I have a little delay when i record a track next to a track previously recorded in the same project. In other words when I listen back to them played together, they are a little off timing wise, enough for the naked ear to notice. I have been lining them up manually because of this.
 
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I haven't done it in Reaper, but it looks like you go Options > Preferences > Recording and uncheck "Use audio driver reported latency" and type a value into the "Input manual offset" field.

Reaper's uses the term latency where Vegas 6.0 (that I'm used to) uses the term record offset.
 
Thank you for your advise. On the Alesis io2 interface there is a knob that gives you the option of running it Direct or USB. However, it is going to connect through the USB chord no matter what, because that is the only option for connecting it to a computer. Now my question is, which one should I use? because i have been turning the knob all the way to USB.
There's your problem. That knob mixes what you are playing/singing live with what is coming out of the DAW. If you are monitoring through the DAW as well as directly, you are going to hear the phase of the two signals conflicting.

Turn off the monitoring in your DAW and turn the knob towards direct until you have a good enough mix to perform your part and you should be good to go.
 
sound moves at 1ft per millisecond, so 1-3ms is the difference between 1ft and 3ft from the speaker.
you are NOT humanly going to be able to tell the difference.

In blind medical studies MOST humans can't percieve below 20-30ms. Cranking it down below what
you can actually percieve only stresses the cpu more and is counter-productive. Anything below
10ms should be fine.

Dont chase numbers, it's not a peeing match....

I'm not chasing numbers, but 10ms (very approximately a 512 buffer size in most daws) will SERIOUSLY bother just about any drummer, pianist or guitarist I know, and I don't know any picky primadonnas.
 
There's your problem. That knob mixes what you are playing/singing live with what is coming out of the DAW. If you are monitoring through the DAW as well as directly, you are going to hear the phase of the two signals conflicting.

Turn off the monitoring in your DAW and turn the knob towards direct until you have a good enough mix to perform your part and you should be good to go.

Exactly. The purpose of that knob on the IO2 is to mix the playback from your DAW with the material being recorded live BEFORE it goes through the DAW. If you're mixing your voice (or whatever) live and via the DAW, it will sound a bit strange.

I agree with the comments about "chasing numbers" though. If you're playing as part of a live band, even on a small stage there's going to be 8 or 10 feet between the person on the right and the one on the left--and a varying lesser amount for the people in the middle. If you're a guitarist and your big amp is 5 feet behind you, there's 5 and a bit ms between you picking a string and hearing the amplified sound.

The same applies to the audience who hear various parts of the band at different times--and it doesn't sound out of time or un-natural. The only trick is you don't want to hear YOURSELF out of time, and that's why any half decent interface offers some form of hardware direct monitoring with zero latency.
 
Most people would say you're being too picky. I say if you hear it and it affects you then you probably have a more precise sense of timing than most people.

You can reduce latency to the sub-microsecond level by going to an all-analog monitoring path. Some 2-channel interfaces have that built in; a simple knob on the interface is a good indicator. Otherwise you need an analog mixer with sufficient quality and features.

Latency is unnoticeable through my Focusrite 2i2 when running GuitarRig4.

If I set up a complex FX chain and try and monitor vocals then I start to notice it.

I recommend using a crappy reverb on vocals when monitoring and dont have any multiband compressors on your master bus. ;)
 
Perhaps when you use the direct monitoring you're still hearing the inputs through the USB path as well. That could make the inputs sound bad. You probably just need to turn off input monitoring in your recording software.

I just had that issue today, somehow it was a dual playback in the WIndows Mixer. I muted the "copy" line and problem was gone. It was a very subtle delay, until playing fast notes then it would sound like a delay effect was on intentionally.

Easy to check out.
 
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