On Tue, 20 Jan 2015, Leonardo Gabrielli wrote:
Thanks Len for the useful feedback.
I was going to say 10ms is about where I start to feel disconnected
from
what I am playing, but I realize I am thinking one way and round trip
would be about 20ms. So 16ms might be well workable.
Psychoacoustic tests (many of them from the CCRMA SoundWire team and others)
shows that 21-25ms is acceptable to keep a steady tempo. Larger values make the
performers slow down, very low values make them accelerate! AFAIK in musical
Hmm, I think though there is a difference between "keep a steady tempo"
and comfortable. Someone (those with money for whatever they want) who has
to play every night will choose less clean if it is more comfortable to
play with. It is hard to market something the "pros" don't use. What that
means to me is that it has to be not perceptible. I have to not notice it.
It is one thing to be able to play with it, but I think my playing (or
artistry) hurts when I have a feeling of something "not right".
I also think there is a difference depending on which instrument is played
and what effects are used. A guitar through an amp suffers only the delay
the sound takes to get from the amp to the ears pretty much. A pipe organ
will have a much longer delay from key down till sound hits the palyers
ears, both from mechanically opening the pipe but also the pipe will be
farther away (16 foot pipes just seem to take a while to get going too).
The synth player will also be used to some delay and may even add it for
some swelling kinds of sounds. But the synth player is probably closer to
his monitor too. So different instrument players will expect a different
amount of delay.
instrument design the 10ms figure is taken as a threshold for your instrument
response (e.g. keypress --> sound), but in performance the values can be higher
as said.
That would be another problem. designing for higher latency means that the
player who adds digital effects (or the synth player who has play delay at
10ms already) will be looking at that delay added to the delay of the
monitor delay. As a player, my feeling is that "if this new digital system
makes it harder for me to play, I will stick with my analog one thank
you." So the bar to meet is not what is good enough, but rather what will
not be noticed. The various wired digital transports are all under 5ms
latency, 3ms is common for one direction, but "as low as 1ms" is what most
claim. With a digital mixer, the delays add up. mic in to pcm is .5ms, to
FOH is 3ms. Mixer effects will add time because they use 3 or 4 DSP boards
for eq/comp/whatever in each mix strip as well as whatever the master
throws at it. Every time the audio goes to another DSP, there is a buffer
worth of time lag. I would assume each strip stays within the same DSP,
but will move to another one for master or monitor. SO there may be delay
there. then it has 3ms back to the speakers/inear along with DAC delay.
A keyboard player may already be looking at noticable delay at that point,
or at least on the edge of it with the delay in their synth thrown in.
In such a case, it would not be possible to add a WIFI that went to AP
then snake then FOH mix then monitor as there would be close to 30ms. The
WIFI would have to be a dirrect input to the FOH mix.
In the case with the three players all on different boats, it may have
actually been easier because of the separation of things. There would be
little audio from an acoustic path... plus there was less choice to make
it work :)
SOme digital FOH mixers do not actually have the PCM travel all the way to
the mixer and back. The "mixer" is really a controller and the mixer is in
the box that looks like the stage end of the snake. So the only delay is
the network to monitor box delay. This kind of system may be more
workable... it is just out of my price range so I don't think of it too
much :)
So my feeling is that WIFI is not worth using in the studio anyway where
cables are easy and sure. To work on stage though (where they would be of
good use) they need to have lower latency. I think shooting for minimal
AES67 compliance latency would be something that is not noticable.
Minimum AES67 is:
48k/24bit 48words per packet with 3 packets worth of buffer. (I can't find
my paper just now so I can't check, but axia's page says that their
compatibility mode uses 48 sample packets which they call 1ms, but the
AES67 paper suggests buffering 3 packets at a time for stable operation)
This is 3ms transport delay. If WIFI can hit that, things will work. I
know all the blurbs say that AES67 is less than 10ms, but the reality is
that it is 3ms at the first compatibilty mode. This is what gives the
widest range interconnectivity as there are already some products out
there where this is the only mode available.
It would be interesting to see what happened to two AES67 devices
connected by a fast WIFI link with various amount of radio noise and audio
channels. I would imagine the maximum number of channels in any WIFI link
would be 4 (2 in each direction) though 1 or 2 might be more common.
It would, of course, be possible to use larger buffers in the
WIFI link itself and smaller buffers in the AP to wired end... in other
words use something other than AES67 and then convert. However, I think it
would add more latency where less is needed.
A system approach is needed where the whole system where WIFI will be
used, is analysed as a whole and the max latency is based on the system
rather than just the transport itself.
--
Len Ovens
www.ovenwerks.net
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