Neglectable latency? Right.....

Halion

Halion

New member
It seems I have been misslead somewhat by the "extremely low latency" statements on soundcards like those sold by M-Audio, Terratec, Emu etc.

I have a Terratec EWX24/96. Up until now I've always used hardware monitoring so the latency wasn't a probably, but I've tried software monitoring and it is becoming a problem. I've got workarounds but I just want to explain a bit how much my total latency from the point of the microphone to the headphones is.

System:
AMD Athlon XP 2000+
1gig ram (DDR, but not the fast kind)

First the theory part:
If I record one single track at 24bit 96khz at the theoretical buffer size of 128 samples, I can get a latenly of something like 2 ms.

BUT: That's not all the latency there is. In the control panel of my terratec, there's also a DMA Latency box, which has selectable values from 1ms to 10ms. It didn't seem to make a difference where I set it (although I'm sure it has to do with stability somehow), so I set it to 1ms.

Then there's output latency. What that's you say? Yes, there's such a thing as output latency (I didn't know either). And apparently, it's bigger even than the input latency. Lets say, again theoratically, it's 4 ms.

Then there's more. When I'm looking at the black bootscreen when I start my computer, there's also shown how much latency my ram has, and apparently, it is 2.5 ms.

Quick math: 2 + 1 + 4 + 2.5 = 9.5ms. Still good enough.

However, there is no way I can run a decent project like this. First of all, I don't want to record at 96khz since it hogs my system and fills my harddrive like a speed demon. So it switch to 44.1 khz. WHOP, there goes the latency. 2 ms input latency turned into 6 ms (keep in mind, buffer is still on 128 samples). The output latency (which I can only view in Cubase) suddenly shoots up to something like 12 ms.

Quick math: 6 + 1 + 12 + 2.5 = 20.5ms Edgy.

However, after a track or 2, I'm getting pops and clicks out the butt, so all I can do is enlarge the bugger. I'm taking it slow and going up to 256 samples. Input latency is now around 8 ms and output latency is around 16 ms.

Quick math: 8 + 1 + 16 + 2.5 = 27.5ms Too much.

It sounds like a lot more aswell (well over 40 ms if you ask me, but I'll blame my ears for the time beeing).

Now if I want to run things really smoothly, I'll have to enlarge the buffer a bit more even.

So much for "extremely low latency". The ASIO4ALL drivers I found on the internet faired a little better, but still too much latency.

Just felt like sharing.

*End of Rant*
 
woah... that's really interesting.

i didn't realise it could get that big just through the normal systems, without anything actually acting slowly or whatever.

i might have to try and work it out myself.

thanks for that,

Andy
 
No problem.

It's not that thinks work slowly. When you're not tracking all is fine. You don't notice the latency. But the fact that you are playing something and hearing it back a little bit late makes you approuch the feel and groove differently. Like, if you want to make something sound really laid-back, so you're playing everything a bit late, but because of the latency it sounds like you're now really really late, so you start playing more in time, and suddenly you're smooth touch and chill groove turned into a stompin' quantizy sounding rock beat.
 
Who there, I think you are making some bad assumptions.

You can't just look at some specs and add them together unless you know how they are measured. I suspect some of those measurements are made with the latency of the RAM and the other parts of the PC's guts folded into it already.

Anything below about 10 ms you would be very hard-pressed to detect, and it shouldn't mess with your feel at all.

The latencies of players on a stage because of the difference between each other, the amps, the PA monitors, etc. is on the order of 10 ms.
 
I know, I never said I did notice 10ms of latency. But when I hear it, I hear it, and I heard it.

Ok it might be a bit fast to just add that stuff up like that. But in the end, I'm still getting way more latency than what's advertised.
 
I had the pleasure of recording with a phenomenal jazz bassist who was donating his time to a children's project I was working on. I don't know Kai Eckhardt's full bio, but I know we recorded and toured with John McLaughlin back in the 90's. Anyway, great guy and a great bassist.

He heard the latency on my Pro Tools LE system (002) even on its lowest buffer setting. I couldn't use "low latency monitor" mode because he wanted to have some reverb on his lead track. So that's around 12 or 13ms of delay?

So we switched to ADAT for the project and I used a hardware reverb. I know the ADAT still has conveter latency, but he was already comfortable with them and away we went. There were other things I could have done (monitor through the mixer and set up the hardware reverb) but it was just easier and more comfortable to switch systems. It was a good lesson- I'm not at all bothered by the latency of the system, but that doesn't mean it work for other people.

Take care,
Chris
 
10 to 12 ms of latency is actually very easy to tell. If you can't notice 10 ms of latency, than you really need some more practice and ear trainging. I have had keyboard players bothered by even 6 ms of latency. Timing is everything when recording. Thats why I do all of my monitoring in the analog realm. That way I never have to worry about latency at all.
 
I thought most software corrected for latency now... I may be misunderstanding... but if I lay down a guitar track over drums and bass, it is recorded exactly as I played it, when I played it, over the other instruments. I don't know... I use an Aardvark Direct Pro 24/96, with Acid.
 
Yeah, but when you want to hear yourself play WHILE you play it, than you notice the latency. The computer compensates the second you hit the stop button.
 
hmm, that has not yet happened for me... I might have a drum and guitar backing track and plug bass DI... and can play right along with it listening through the monitors and no latency. in fact it sounds exactly as it will sound in the mix as is.
 
No program can counter for I/O latency. That would be like the program knowing to start itself 35 ms before you hit the button to start:D I/O latency is a fact of life. If you have a card that offers input monitoring though, then you can get around it because the card itself can ereturn you the signal before it hits any hardware and software that latency can be introduced.

When recording tracks, you can't really record in complete sync, you will always have an input latency in place. However, input latencies are much easier to manage and are typically much lower. You might hear the note that you just played a little late while input monitoring through a software application, but in general, upon playback it should be very very close to where you actually played the note. This is because while you are recording it and trying to listen to it in real time, the output latency delays the signl that is sent back to your headphones or monitors as your host software processes it, but it was still actually recorded very close to where you actually played it. Basically, it is purely the output which is delayed.

During mixdown, many apps now will automatically calculate the delay times that each individual plugin and soft instrument might create. This way you can hear all of your tracks in sync while you are mixing. What happens here is that the software application does not give you an immediate response when you hit play. It internally shifts all tracks so that with the plugins that you may have running on them all of the individual outtuts remain phase accurate and in sync with one another. The bigger your mix and larger your plugin in affect on your processor is, the longer that time will take when you first hit play.

I hope some of that helped.
 
If you can't notice 10 ms of latency, than you really need some more practice and ear training. I have had keyboard players bothered by even 6 ms of latency.
Click to expand...

Oh boy...

OK, say you're standing ten feet from the drummer's kick.

Sound travels at 1116 ft/s at sea level. To find out how long it takes to travel 10 feet, divide 10 ft by 1116 ft/s. The answer is 0.009 s -- 9 ms. For less than 6 ms of delay you need to be about 6 feet away.

So how do these picky keyboard players of which you speak manage to play with other people? Do they sit in the center of a circle and the rest of the band all have their sound sources six feet or less from the keyboardist's ears?

If anything, a musician used to playing with people would have trouble if there were no latency, because they would be playing as if there were the natural spatial latency in what they are hearing and locking with, so the natural feel would be harder to achieve...
 
Its called headphones. Some people don't notice or care, and some are driven crazy by it. They just hate that lack of response when they hit a key and it isn't right there. I agree with your principle and the science of it, but when a musician notices and says something about it, then you HAVE to fix it. Otherwise they get a little nervous and tentative. Also, try running the dry source in conjunction with a 6 ms latency returning from a cpu, there is quite a noticable phasing effect. The point is that it is there, and sometimes can become an issue. That is why I never monitor live inputs through DAW software and an OS. Also, when playing live you are bombarded with other sounds and that kind of sonic smearing is expected and accepted. There is also the visual aspect when playing live together. Once you put a pair of headphones on though all of your expectations tend to change somewhat.
 
Yeah. I wasn't trying to start a discussion on wheither or not we can all hear 10ms of latency or whatever. I'm much more interested in the results other people have gotten out of their DAW, latency-wise.
 
Sorry, in trying to type as fast as my brain was communicating to my fingers, I may have inaccurately portrayed my thoughts. To see just how bad your latency actually is (without looking at the actual numbers) you can take a live input and monitor it from both the source, and through the DAW application. Then we should all be able to hear just how bad 10ms is.

Also, I am sorry if this thread did get a little sidetracked from the original topic, but I do feel like there is some good information here for many of the others who may read it. I personally could not tell you how my latency is becuase I have a large format inline console, so I do all of my monitoring in analog land before even hitting the DAW and as a result, I have no latency. OK, technically there would have to be some, but it is far smaller than any software can offer.
 
Not to start an argument, but 6 ms latency is like having your amp 6 feet away. If you can't play like that, there is something wrong.

Musicians are superstitious, If they can blame something else, they will. Especially if they can blame something they don't understand, but heard that it was bad.
 
Its not a question of right or wrong or superstitions or not. My job is to achieve the best quality recordings possible. Thta starts with performance. Performance starts with confidence, and that is directly reflected by the musicians level of comfort. We are also talking about more than one musician here that has no preconceived notion or knowledge of latency. Also, guitar players are used to having their cabs away from them, its part of their sound. Piano players are not. They are used to a different response...i.e... hearing the sound when they push the key. Wearing heapdphones calls that kind of delay more to your attention. If they hear it, its there, like it or not. Some people may care, some people may not.
 
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