[RWP] Latency, is shorter always better?

[Jim Snowbarger]

Now and then, I feel like a slight departure from topic..  And, this is one 
of them.  So, stand bye with your delete key ready as I carry on.

This probably belongs over on MidiMag.  But, I don't feel like joining just 
so I can post this once in a blue mooner.

One of the great things that digital audio processing has brought to us is 
so-called latency.  You might just call it delay.  but, in the 21st century, 
we like to use clever names.  It makes us feel smarter.  So, let's co-opt 
the term latency, which had a totally different meaning before the 
techno-gods got hold of it.  And, let's now define latency as the act of 
being late.  But, however you slice it, it comes down to delay.

Digital devices impose delay mostly because data consumers, like sound 
cards, or recording devices, have learned to be defensive, knowing full good 
and well that data providers, such as input sound cards, or other streaming 
devices, can not be counted on to keep up a steady stream of data.  Internet 
congestion, or scheduling congestion inside your own machine, can 
temporarily block the normal flow of things.  Sound playback requires a 
rock-solid comsumption rate of the data. The sampels need to keep flowing. 
You might not get that next buffer load of data in time.  so, it pays to 
keep a backlog.  The more backlog, the safer you are.  But, if the backlog 
is too great, you get, latency, that annoying delay.

I recently picked up one of those fine Computers Chris is always talking 
about from StudioCat.com.  That is one very fine box.  And, now that I also 
own Chris's Delta 1010, I was enjoying fine-tuning my latency down to 
acceptable levels,  not carefully measured, but clearly less than 10 
milliseconds.

Most of the recording work I do involves a microphone and headphones.  I am 
quite typically listening to my own voice as I speak.  If you have listened 
to the Snowman Radio Broadcasts, you know the kind of multi-track microphone 
work I'm guilty of.
When living on machines where such short delays were not possible, my habit 
was to listen to my own foice direct out of the mixer, and not going through 
Reaper.  So, I kept the reaper monitor off.  What was annoying about that is 
that, if I panned my various character voices in the stereo  mix, then, my 
direct microphone sound would not be panned the same as the character voice 
track I was recording into.  So, when it played back, it came from 
elsewhere, and was more than a little bit confusing.

But, with delay this short, I find that I switch off the direct sound, and 
now can monitor the signal coming back from reaper with the monitor turned 
on.  So, I'm now listening to a delayed version of my voice, and it is 
panned to the same place where that character voice sits, which helps me 
keep track of who I am supposed to be right now.  And, I can more easily 
tell now whetehr a track is armed, and even if one is armed that should not 
be.  It's nice to be able to work like that, just listening to reaper's 
output.

But, here is the cool thing.  The exact amount of latency you provide 
affects the quality of what you hear in your headphones.

No matter how good your phones, the sound that you hear when you are 
listening to yourself speaking live into a microphone, is actually the 
composite of at least two signal paths, and maybe more.  Yes, there is the 
direct signal coming through Reaper.   Then, there is bone conductivity, the 
sound of your own voice coming through the structure of your head, which 
will very somewhat with density.  If you don't get any of that, you might 
wonder about that density stuff.
And maybe even, there is leakage around the ear muffs.  In all, it is a 
complex sond that actually reaches your ears.  And, the phase relationship 
between all of the various contributors will affect the frequency response 
of the final signal that you hear.

In the old days, we knew about the affect that phase would have on such 
things.  Having your head phones out of phase with your microphone left you 
feeling empty headed, due to the phase cancellation that took place.
But, since delay was in the nanoseconds, we didn't get to know so much about 
the effect that delay would have, despite our compulsive preoccupation with 
tape delay.

Phase is mostly a frequency independent phenomenon.  Yes, we know that some 
systems, especially mechanical transducers, or even cheap equalizers, which 
will have a reactive component to their impedance,  introduce a variable 
amound of phase shift, depending on frequency.  But, usually those effects 
are at the far ends of their usable range.
In general, especially in mixer land, where things are nice and linear, and 
where impedances are strictly non-reactive, if you put something 180 degrees 
out of phase, you will get perfect cancellation, all across the frequency 
band.

Enter the digital age, and the new innovation, latency.
The relationship between signal phase, and a delay is frequency.  For 
example, a delay of 4 milliseconds is one full cycle of a 250 Hertz tone. 
But, it is only half a cycle of a 125 hertz tone.   It is all still a 4 
millisecond delay.  But, the phase impact depends on the frequency. 
Combining the pre and post delays of these two tones with that 4ms delay 
will have completely different effects.  The 125 hertz tone would be nulled 
out.  The 250 hertz tone would actually see a 6 db increase.


The result is that, if you put a delay in front of your headphone mix, you 
will cause what is referred to as a comb filter effect on the perceived 
headphone signal.  It is a filter that has a frequency response curve that 
looks like a rola coaster, with hills and valleys.  If you were listening to 
an audio tone sweep, one that you would actually need to sing, in this case, 
in order to get that bone conductivity thing happening as well, As you move 
steadily up in frequency, the sound would be much stronger at some 
frequencies, and much weaker at others.  As the tone rises, you would hear 
rising and falling of the net response.  And, changing the amounbt of delay 
slides that comb up and down the audio spectrum.
Depending on several things, the frequency range of your voice, response of 
your headphones, your ears, the density of your grey matter, your 
preferences, and on and on, you might have preferences about the optimal 
position of that comb.  What frequencies do you like to accentuate?  And, 
which to attenuate.

The cool thing is that, by fine-tuning your headphone latency, you can 
position that comb how you like, and can optimize your headphone experience. 
The latency needs to be short enough to not give you a delay echo effect. 
But, beyond that, the shortest possible latency may not give you the 
headphone experience you like.  Instead, relax it a little, and see what 
enriching tones come your way.
Silly you.  And you always thought shorter was better.  And now you know.
TROTS.