The Latency Thread?

Anyone interested in DAW latency and testing what latency starts to actually bother you?

I like it as low as possible, but 13ms on an electronic drumkit is pretty damn playable.

Most people start to notice latency around the 15-20ms zone.
 
If you've got a Vdrum kit, load something like Superior Drummer 3 and play with 10ms RTL.
People who buy e-kits regularly have to put up with 10+ ms, and that is entirely in the box, keeping the computer out of it. The TD50 quotes 7ms, but I'm not sure how accurate this is, or even if it is consistent across all kits or DSP loads.

You typically get 3ms just because of the scan time required to track the voltage, convert it to digital, then convert that to something usable that the synth engine can translate into a midi note.

If you're using a DAW and the soundcard is giving you 2ms round-trip (doubtful, but depends on the gear) then you've still got to add the scan-time and the MCU conversion time to that figure.

Even the polarity of the piezo's can affect the final latency.

No-one is achieving 2ms round-trip latency with electronic drumkits, AFAIK.



Here's an interesting comment:
Ages ago I set up an experiment where I had a drum module that had different latencies on different pads (it was surprisingly easy to do). And in my experience, it was only at the higher limits that drummers could tell which was slower. At fast latencies, drummers couldn’t generally tell difference between 2.5ms and 15ms.



Also:
We have presented a study that investigated the impact
of latency and jitter on the temporal accuracy of perfor-
mance and judgments of instrument quality for two
groups of participants; professional percussionists and
non-percussionists (with varying amounts of musical
experience). The studies involved quality assessments
of a novel percussive instrument with variable latency
(zero, 10 ms, 10 ms +3 ms, 20 ms), temporal accuracy
tests and structured interviews.
In terms of judgments of instrument quality, we
found that both groups showed a preference for 0 ms
in comparison to 10 ms +3 ms and 20 ms latency.
Importantly, the 0 ms and 10 ms latency conditions
show no significant difference in rating for either group.
This suggests that a stable latency of 10 ms is acceptable
to performers of a DMI where 20 ms is not. The 10 ms
+3 ms latency condition was rated in a similarly neg-
ativemannerto20mswhencomparedtothezero
latency condition, suggesting that the addition of a ran-
dom jitter of +3 ms is enough to negatively effect the
perceived quality of an instrument. Our results support
the recommendation put forward by Wessel and Wright
(2002) that DMI designers should aim for a latency of
10 ms or below with a jitter of 1 ms or less. However, our
findings cannot tell us exactly what the minimum
threshold of acceptable latency is, except that it must
be somewhere between 10 ms and 20 ms

 
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Then there is a factor of whether you can hear your instrument acoustically along with the processed sound.
I have noticed that I can tolerate longer latency times if I can't hear the guitar acoustically.
 
When I was using a laptop rig and could monitor latency I found I started to notice it when it got above ~7ms and it started to bother me when it was above ~10ms or so.

I also remember walking the house with a wireless 100 feet from the stage during sound check and thinking it felt a little weird but I could still easily play with the band.


The general rule of thumb seems to be the more you focus on it the more you notice it, and the more you look at and talk about raw numbers the more you start to worry about hypothetical problems you likely will never actually experience
 
Another thing to add - don't trust the numbers your DAW reports. Always measure the input output latency by looping the playback system output to an input, recording it, and comparing the offset to your source signal.
 
No-one is achieving 2ms round-trip latency with electronic drumkits, AFAIK.
Point taken, but my point is still valid.
Lower latency is always going to feel more responsive/immediate.
I think there's a propensity to want to "buck" this idea... because it can be expensive/difficult to achieve effective (glitch-free) low RTL (especially under decent loads).
Part of why playing an acoustic drum kit is more "alive" is precisely because the drum heads and cymbals are immediate/responsive.
BTW, I no longer have a TD-50 based Vdrum kit. Part of what I didn't like was the lag.
Playing the TD-50 wearing headphones, pull either the left or right off your ear... and play.
The lag is pretty obvious.

Years ago, when I created Purrrfect Drums and then Purrrfect Drums 2... I did a lot of triggering drum samples via DrumKat, TrapKat, Vdrums, etc.
There's zero doubt in my mind that when the DAW was set to ~3ms RTL, it felt a whole lot more immediate than when it was set to ~10ms RTL.
To your point, there was also latency from the trigger process itself.

Some have perfect-pitch.
Some are more sensitive to latency.
Some are distracted by fan noise. (drives me crazy - I've sent back an expensive Neve Summing Mixer because of fan noise).
Neve refers to the fan as quiet.
Some can tolerate the noise, some can't.
 
Another thing to add - don't trust the numbers your DAW reports. Always measure the input output latency by looping the playback system output to an input, recording it, and comparing the offset to your source signal.

And before doing that, you need to do a loopback test... to make sure the audio interface driver itself isn't causing an offset.
Many ASIO drivers don't report their latency accurately.
This causes a time offset.
Take a short spike (high transient) audio signal, route it from analog output to analog input... and record the result.
Zoom-in fully... and you'll see if the two spikes are perfectly aligned in time.
If there's any difference, you need to add a manual offset (in samples) in the DAW.
 
it has everything to do with human performance limits.

Not all humans have the same exact performance limits.
Watch Stan Lee's Super Humans
Some can climb much faster than a typical human... due to muscle composition.
Same with world-class sprinters.
Some have accuracy with aim that most people can't achieve.

For someone with perfect pitch, an A4 a few cents out may drive them crazy.
Doesn't bother most folks.

While it's fun to poke at Steve Vai's 2ms comment, he may be able sense the additional lag.
 
While it's fun to poke at Steve Vai's 2ms comment, he may be able sense the additional lag.
Which we can't really prove without putting Vai to a blind test. It's possible he can spot it, he is from another planet after all! :alien: But for us mere earthlings it's not a problem.
 
Which we can't really prove without putting Vai to a blind test. It's possible he can spot it, he is from another planet after all! :alien: But for us mere earthlings it's not a problem.

I can confidently state he cannot. And this is not guessing either: no one can. No more than no one being able to run 100mph, or seeing infrared light, these are inherent limits to how all of us are built.

Thing is, i'm also 100% certain he believes he can spot it... and that makes all the difference in the world. Psychoacoustics are weird; if one is convinced to hear something, you will hear it. So if Vai is convinced a few ms can throw his playing off, and builds his rigs accordingly, well... that benefits him, and all of us.
 
Vai has later said he can't feel the latency of running his Synergy preamps through his Fractal, so he wouldn't using amp models either.
IMO I think he really meant to say he didn't like the feel of the modelling vs the tube preamps..

 
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Point taken, but my point is still valid.
You're a science guy yourself. I follow your posts on computing and gear because you're a knowledgeable guy. But let us present the facts.

For example, you made no mention of scan times nor microcontroller processing times, both of which typically add up to 3 or 4ms themselves. This means that the moment you hit the pad, to the moment you hear a sound, there is a baseline 3ms latency response as standard. Even if you're getting a round-trip of 2ms - more or less only possible via Thunderbolt at 88.2khz sample-rate, with specific devices - your total latency will be 5ms.
 
To illustrate (and yes, for yourself, if you like) how sensitive we are when it comes to latency: Double a signal in your DAW of choice, pan both hard L/R, add some samples of delay (using track parameters, a 100% wet delay /w 0 feedback or whatever) just to one of the signals. That's how incredibly great at detecting the smallest latencies the combination of our ears and brains is.
It's likely also a matter of evolution. When the leaves of the jungle behind you were rustling back in the days, you simply had to know whether it was slightly left or right because that might've easily become the deciding factor for that approaching sable-toothed fellow to have a nice meal or to stay hungry.
For more scientific explanations, one could always look up the Haas effect.

Note: No, this isn't all too directly related to amp modeler and DAW latencies - it just goes to show that at least in general, our senses are absolute suited to deal with extremely small latencies.
 
To illustrate (and yes, for yourself, if you like) how sensitive we are when it comes to latency: Double a signal in your DAW of choice, pan both hard L/R, add some samples of delay (using track parameters, a 100% wet delay /w 0 feedback or whatever) just to one of the signals. That's how incredibly great at detecting the smallest latencies the combination of our ears and brains is.

That is, by definition, not latency. You're experiencing phase misalignment - and yes, your ears are incredibly sensitive to it.

It's similar to how our brain picks up minuscule differences in relative pitch, and the reason why you can tune instruments by ear very precisely.
 
That is, by definition, not latency. You're experiencing phase misalignment - and yes, your ears are incredibly sensitive to it.

It's similar to how our brain picks up minuscule differences in relative pitch, and the reason why you can tune instruments by ear very precisely.
Comb filtering is latency based. Generally detectable by people at around 6ms of differentiation.
 
Perceiving latency isn't exactly related to musical time, groove or awareness of either. Sure, there's overlapping areas, but they're defenitely not the same.
You're right, I should have added more info in my post.
Point was to give another perspective of how short of a time 10ms is in relation to music and how even trained musicians are never that accurate.

I believe that hearing the instrument acoustically together with the processed sound is responsible for a lot of the hullabaloo about latency.
 
I said this before (and received a ban for it elsewhere) but, when I was testing the iPhone iPad stuff with a few hardware companies, pretty much the lowest Round Trip Latency we could get was around 20msec....And this was with chains where people who complained about the "much too high" round trip latency of like 4-8 millisecond on a DAW, yet somehow were just fine playing onstage at 20 ms.

Pretty comprehensive studies have been done for musicians tracking (when there wasn't an audible parallel path within 6dB of the amplified level at the same time) came out with some Paredo principle type numbers of like 75-80% being ok until 11-13 milliseconds. Among the leftovers were people who were really finicky even at 5ms and people who could discern it but didn't bother them until that 11ms or so
 
Comb filtering is latency based. Generally detectable at around 6ms of differentiation.

Yeah, but "differentiation" is the keyword here: you need two sources at play. Comb filtering emerges from phase interactions on two identical signals propagating, and merging back, at different speeds.

Not what people refer to when discussing latency, and particularly digital latency, which is, by its very nature, frequency- and phase-invariant.
 
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