Att: Mike Rivers re digital latency vs. need for mixer

On Sat, 14 Jun 2008 17:11:38 -0400, "Soundhaspriority"
<[email protected]> wrote:

>Echo Audio has provided the following figures for latency through the
>Audiofire 12:
>
>48 kHz: 1.83 ms
>96 kHz: 0.77 ms
>192 kHz: 0.38 ms
>
>The figures, particularly at 192K, are much better than the 3 ms that is
>typically bantered about.
>In air, sound travels approximately 4 inches in 0.38 ms. This is
>approximately the air path between mouth and ear. Since sound travels
>through bone much faster, a comb filtering effect already exists between
>bone and air conduction.
>
>The above suggests that a vocalist MIGHT be satisfied with a 0.38ms delay,
>eliminating the need for an analog mixer. I am in no position to assert it
>as fact, but it's worth a try.

Oh, most vocalists are perfectly happy with 5ms delay. Certain
styles, usually those who want minimal level in their cans, have
problems.

Laurence Payne wrote:

> Oh, most vocalists are perfectly happy with 5ms delay. Certain
> styles, usually those who want minimal level in their cans, have
> problems.

This is an alternate solution. 5 ms delay is well outside the annoying
comb filter effect range, so there are cases where more monitoring
latency can be better than less.

--
If you e-mail me and it bounces, use your secret decoder ring and reach
me here:
double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers
([email protected])

Soundhaspriority wrote:
> Echo Audio has provided the following figures for latency through the
> Audiofire 12:
>
> 48 kHz: 1.83 ms
> 96 kHz: 0.77 ms
> 192 kHz: 0.38 ms
>
> The figures, particularly at 192K, are much better than the 3 ms that is
> typically bantered about.

That's not unreasonable. I think I measured about 1.5 ms from line input
to monitor outputs on the Echo (Event at the time) Gina, the first
external computer audio interface that I reviewed. The greater-than-half
reduction as the sample rate doubles is a bit unusual with modern
converter chips, however. Intuitively it would seem so, but they don't
work quite as intuition tells us. There's a certain amount of fixed
delay in the fliters that's present at all sample rates.

No matter what Echo writes in their performance data, the real test is
for you to measure it in a real (yours) system. Have you done that? What
results have you achieved?



> In air, sound travels approximately 4 inches in 0.38 ms. This is
> approximately the air path between mouth and ear. Since sound travels
> through bone much faster, a comb filtering effect already exists between
> bone and air conduction.

> The above suggests that a vocalist MIGHT be satisfied with a 0.38ms delay,
> eliminating the need for an analog mixer. I am in no position to assert it
> as fact, but it's worth a try.

I'd say "try it" but so many people essentially have tried it and said
"I don't notice any comb filtering" that without some controlled
listening and training as to what to listen for, it's really not a valid
conclusion.

I'd take a room full of people, a mixer, and a variable delay, and let
them listen to some recorded examples with the delay added in over a
fairly wide range of time and relative level, starting with an example
that everyone should be able to hear, then changing each parameter (time
and amplitude) to determine just what it takes to make a perceptible
difference when the delay is mixed in and when it isn't. I think that
most people who say they don't have a problem with monitoring their own
voice through a 1-3 ms delay are simply listening to the headphone level
loud enough so that the comb filtering effect is minimal. Remember, you
only get complete cancellation at equal levels (and everyone's "internal
gain setting" is different) so you can't generalize even to a specific
SPL at the headphones.

So the problem has a solution that's practical for most users as long as
we can accept that the short delay doesn't affect how we play an
instrument, and I think that's reasonably well established by the guitar
amplifier or on-stage monitor examples.




--
If you e-mail me and it bounces, use your secret decoder ring and reach
me here:
double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers
([email protected])

On Jun 15, 10:10 am, Mike Rivers <[email protected]> wrote:
Snip
> So the problem has a solution that's practical for most users as long as
> we can accept that the short delay doesn't affect how we play an
> instrument, and I think that's reasonably well established by the guitar
> amplifier or on-stage monitor examples.

Well, the flipside is a small ensemble playing in a large space for
the first time can certainly be thrown off by the "live latency". I
have observed both as a musician and live sound guy that problems with
the added delay in larger venues is not uncommon for younger
performers. Add in reflections and slap echoes and this can cause not
only timing problems but pitch problems as well.

As a musician, you really have to learn the relationship between
the space and where you exist in time, initially you might find visual
cues are helpful as reference points/timing aids. But you do adapt
and learn to filter certain things out, and sometimes it's not always
as pleasant a reality performing as it is being in the audience. But
experienced players do learn to listen differently, and have a
stronger internalized sense of place I think.

Live latency - just saying.

Will Miho
NY TV/Audio Post/Music/Live Sound Guy
"The large print giveth and the small print taketh way..." Tom Waits

WillStG wrote:

> Well, the flipside is a small ensemble playing in a large space for
> the first time can certainly be thrown off by the "live latency".

They can't hear themselves on stage without monitors other than what
comes back to them from the hall, or from the sides or back of the PA
speakers, which are quite some distance away and are very far off axis.
That's what gave us monitors (which nobody seems to object to even
though they might be 10-20 feet away.

> have observed both as a musician and live sound guy that problems with
> the added delay in larger venues is not uncommon for younger
> performers.

That's because they play so loud they can't hear themselves. There's no
conductor and no music, so they HAVE to hear each other as well as
themselves in order to play together. But this isn't related to the
"latency" problem that we're talking about here.



--
If you e-mail me and it bounces, use your secret decoder ring and reach
me here:
double-m-eleven-double-zero at yahoo -- I'm really Mike Rivers
([email protected])

On Jun 15, 4:26 pm, Mike Rivers <[email protected]> wrote:
> WillStG wrote:
> > ....have observed both as a musician and live sound guy that problems with
> > the added delay in larger venues is not uncommon for younger
> > performers.  
>
> That's because they play so loud they can't hear themselves. There's no
> conductor and no music, so they HAVE to hear each other as well as
> themselves in order to play together. But this isn't related to the
> "latency" problem that we're talking about here.

Guy who play too loud, a lot of the time they expect monitors to
act as a substitute for actually listening to each other - then they
engage in "parallel play", they play at the same time somewhat, but
don't truly interact. Not very musical.

What helps is giving up "shoegazing" and actually looking at each
other, playing to the time you *see* (body language) and learning the
relationship between that and what you are hearing. It's a skill akin
to learning to play ahead of/right on/behind the beat, or how to
groove with a click track.

This has nothing to do with latency in recording of course, but the
stated assumptions that delays in live performances due to distance
between performers and/or their amps is without it's problems, I just
couldn't let that go by.

Will Miho
NY TV/Audio Post/Music/Live Sound Guy
"The large print giveth and the small print taketh away..." Tom Waits