At my current job we were looking into delay timings of video systems.
We were doing end-to-end measurement by putting a time display in front
of a monitor
and have the camera show both the time display and the monitor.
It looks a bit like the old infinite mirror.
If you arrange things right it shows two images of the time display
one that lags the other. And the difference is the round-trip delay.
When I'm on Skype and my co-worker shares his screen, I can see my own
camera image come back to me in a similar way.
So if I were to point my camera at a rolling time display, he shares
that image back to me, I could take video (with a second camera)
of both the live time display and the delayed.
The particular system we were measuring ended up having a multi-frame delay
due to the video codec having an optional frame re-ordering feature which
required buffering a group of frames at startup and carrying that delay
forward.
We found out how to turn that off.
Chris
On 5/6/20 9:00 AM, jimlux wrote:
Given that there's a lot more people spending time zooming, webexing,
teaming, skype, facetime, etc. these days, I'm curious if anyone has
figured out to *quantify* the issues of lag, desynchronization, etc.
How would one go about instrumenting it (without access to the source
code or servers involved)?
There's two areas of some interest to me:
1) there's several studies that say that when voice and image aren't
perfectly synchronized, particularly if it's not a consistent delay,
or if there are gaps and jumps, that it is more stressful and creates
a cognitive workload that does not exist with actual in-person
meetings (the "why am I more tired after a day of telework than the
real thing")
2) If you wanted to do group music playing or singing, relative timing
among the streams is critical. Is there a threshold where it all
breaks down? For instance, in an orchestra or choir, one has visual
cues from the conductor, but most people do not sing or play using the
conductor as a metronome triggering the next measure's notes. They
also listen to the players around them (or perhaps on the other side
of the stage, some 30-40 milliseconds late)
I can think of ways to "test" a given teleconferencing system
(blinking LEDs in a pattern, tone bursts on audio), but I think
there's some challenges in things like compression algorithms (do they
have constant latency?) and highly structured test signals might not
measure the same as actual video and audio.
I will note that there are subjective difference among the various
tools, and there's differing effects from compression artifacts and
bandwidth/packet transport.
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