Basic Noob Question

B

barnesbabb

Member
My setup.

Guitar > AXE-FX >IMAC via usb from the back of the AXE FX.

I use Logic Pro X.

The latency is poor. I play a riff and it sounds amazing but as soon as I replay it on my DAW its so slow to the point of being embarrassing.

I basically would like to play and record and then when I replay it, it sounds as fast as I played it.

Its clear that things aren't moving fast enough. I have two basic questions.

1) When I watch videos of guitarists that I don't like, they plug their guitar into a box of some sort which has guitar jacks in and out. Do I need one of these sorts of things? Like an interface between my Axe FX and Mac so I don't have to use shitty USB.

2) Is there anything I can do with my AXE to help this or is a USB connection a basic, moronic cretins idea the like of which I spew everyday?

Any guidance would be most appreciated.

The Axe is immense but my skills with technology are poor. I can play a guitar but its really hard when you have to plug it into Skynet to understand it (my limited skills)
 
barnesbabb said:
as soon as I replay it on my DAW its so slow to the point of being embarrassing.
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you probably have your Logic Project set to the wrong Sample Rate. the Axe-Fx runs at 48kHz. set your Logic project to that.

this has nothing to do with "speed" of the communication.

Like an interface between my Axe FX and Mac so I don't have to use shitty USB.
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those interfaces mostly use USB as well.
 
Try the analog outputs to your ADC and monitor thru your mixer. Even a latency of 1ms messes with my ears. Now if the Axe Fx is all you have to your computer then set your DAW to the lowest I/O buffer level that you can and free up as much memory as possible.
 
hochha said:
Try the analog outputs to your ADC and monitor thru your mixer. Even a latency of 1ms messes with my ears. Now if the Axe Fx is all you have to your computer then set your DAW to the lowest I/O buffer level that you can and free up as much memory as possible.
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So you have to play with your ear a foot or less from your amps speaker because otherwise its taking well over 1ms just for the sound to travel from your speaker to your ear, and having supernatural auditory resolution you can't play with more than 1ms.....
 
lqdsnddist said:
So you have to play with your ear a foot or less from your amps speaker because otherwise its taking well over 1ms just for the sound to travel from your speaker to your ear, and having supernatural auditory resolution you can't play with more than 1ms.....
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Yes, thank you for your celestial guidance, clearly from on high. Never said can't. Learn the difference between latency and room acoustics. Now go join the MSM your a perfect example of that mind set.
 
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hochha said:
Chris, do you understand the difference between latency in digital recording and room acoustics?
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Could you please explain? You keep mentioning it, but I don't know what you are referring to specifically. Are you saying there's a difference in 1ms digital recording vs 1ms room acoustics?
 
The great thing about standardised units of measurement like seconds (and their subdivisions, milliseconds), is that they take as long time to pass whether it's between you and a speaker, or a DSP and a source.

You can't hear 1ms. But you're almost definitely not dealing with 1ms, it's more likely to be 10-15ms.

48kHz is about five samples per millisecond. Have a look at your buffer size.

It's okay to backpedal.
 
For those interested, ADDA manufactures only list output latency as far as I have seen. Full round trip is always twice what is listed. When you encode audio even with the best gear there is always latency in that process. Room acoustics is not equitable to latency in an audio recording process, one is analog the other is digital. Now Ed DeGenaro has said he is fine with up to 7-8 ms latency, dosen't bother him. I find it interesting that so many want to jump on me because I say that 1ms messes with (my ears), but please tell me how that has anything to do with you.
 
hochha said:
For those interested, ADDA manufactures only list output latency as far as I have seen. Full round trip is always twice what is listed. When you encode audio even with the best gear there is always latency in that process. Room acoustics is not equitable to latency in an audio recording process, one is analog the other is digital. Now Ed DeGenaro has said he is fine with up to 7-8 ms latency, dosen't bother him. I find it interesting that so many want to jump on me because I say that 1ms messes with (my ears), but please tell me how that has anything to do with you.
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why do you bring up room acoustics? i don't see what that has to do with the latency discussion.

i think people bring up your 1ms statement because of the scientific discovery that humans can't discern 1ms, that's all. we can take it as an exaggeration though. i'm honestly trying to understand what you're sharing with us.
 
Latency detection between our left and right ears is extremely sensitive as it gives us directional awareness in our environment. Latency detection in our hand-ear coordination while playing is a different story.
 
chris said:
why do you bring up room acoustics? i don't see what that has to do with the latency discussion.

i think people bring up your 1ms statement because of the scientific discovery that humans can't discern 1ms, that's all. we can take it as an exaggeration though. i'm honestly trying to understand what you're sharing with us.
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Chris I mentioned room acoustics because [Iqdsnddist] is throwing speakers at me claiming it takes much more than 1ms to reach my ears from the speaker cone, that is the(room acoustics) part. Now specifically, using Logic Pro X and my current ADDA converter I get an output latency of 1.6 ms and round trip of 2.6 ms. Not bad numbers, not great either in my little world, but I could work with it if I had to. However, I don't have to work with it. I monitor in the analog domain when overdubbing because my superstitious little ears lie to me left and right. Now if 1 or 2 or 3ms isn't a problem when recording and overdubbing completely in the digital domain then please call up all the manufactures of ADDA converters and tell them to stop all their research and expenditures in improving these numbers because you could save them alot of time and money, it's all superfluous as you have pointed out! The ops post had to do with the latency he was experiencing, I suggested he monitor on a mixer and avoid the latency. Now not everyone has a mixer, some just have a computer and Axe Fx so they have no other choice. How about getting back to the ops question and stop attacking me about my (Supernatural hearing) per (Iqdsnddist). One more thing Chris, scientific proof is not as cut and dry as you suggest but if it can help sell a product have at it.
 
hochha said:
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I think you missed the point, with the analogy to "room acoustics" - we're talking about the speed of sound, which is just under a foot per millisecond.

In other words, you've added 3ms of "latency" (delay between the source and your hearing the source) just by sitting three feet from the speaker in question.

That's not including any latency before the sound is processed and supplied to the speaker, how long it takes you to process the sound once you've heard it, etc.

I just think you're an order of magnitude off, regarding what you can perceive.
 
I don't think 1ms in any respect can be perceived by humans. And I'm not selling anything :)

We can agree to disagree perhaps. But I can't disagree with the speed of sound.

The OP's post seems to be about sample rate. I think his playback is at 1/2 the speed and pitch due to an incorrect setting. Pretty sure the OP has nothing to do with delay-style latency.
 
chris said:
The OP's post seems to be about sample rate. I think his playback is at 1/2 the speed and pitch due to an incorrect setting. Pretty sure the OP has nothing to do with delay-style latency.
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Good point.

@barnesbabb : when you say that e playback is slower, does that mean that the notes are playing back at reduced speed, and it takes longer for the song to finish?
 
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I believe Logic up/downsamples to match the project sample rate. At least in my experience when I've accidentally recorded at the wrong sample rate I've been surprised to find it wasn't an issue. But maybe that's not always the case.

The other thing to look at OP is the buffer size in Logic. A smaller size will mean lower latency, but there are limits as the smaller the buffer the greater the burden on your computer so you might need to experiment a little.
 
Interesting.

Let's get this thread back on topic. At this point we need to wait for the OP to respond with more details.
 
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