Capturing high-quality audio in a meeting room for videoconferencing is a notoriously complicated problem.
Microphones are crazy sensitive and pick up things like footsteps and conversations outside the door, shuffling feet and tapping on keyboards, and construction and HVAC noise like you wouldn't believe.
So filtering those things out, and then capturing the best quality audio from the current speaker, and trying to get everyone's voice at roughly the same volume whether they're sitting directly across from the microphone or are piping up from the corner of the room...
...and do this all while cancelling 100% of the echo that might be coming from two or three speakers at once...
...it's an insanely hard problem. Beamforming microphones absolutely help in a huge way, because if you know the speaker's voice is coming from 45° then knowing that any sound coming from any other angle can be removed is a really helpful piece of info.
Now, with beamforming microphones, the precise relative location and direction of each mic is known. The idea of creating one big beamforming mic for the room out of people's individual mics is... insanely hard, but super cool.
It's interesting to me that this article is about measuring the quality of voice transcription, rather than about the quality of audio in an actual meeting. But I suppose the voice transcription quality measurement is simply a proxy for the speaker audio quality generally, no?
This could actually be a huge step forward in not needing videoconferencing equipment in meeting rooms. So far, one of the biggest reasons has actually been dealing with echo and feedback -- when people are in the same call with multiple devices in the same room, it tends to end badly. But if the audio processing is designed for that... the results could actually be quite amazing.
And it's well-known that the "bowling alley" visual of meeting participants (camera at the end of a long conference table) isn't ideal. If each participant has their own laptop camera on themselves, it could be a vastly better experience for remote participants.
My company pushes us to have any conference that will include remote people from our desks, even if some or most of the attendees are in the same physical local. It means that no audio is dropped bec of too much cross-talk and that all attendees are on the same footing. Only real issue is that we don’t automatically get headsets, you need to request/expense it.
I've been in many meeting rooms where there's a single projector/tv, and the person controlling it only shows either the remote cameras OR their screen (while they're sharing), so that isolates the remote people even more. (I've also been the remote person in this situation, and it definitely feels more like being an occasionally noisy fly on the wall then a full participant).
Everyone also gets their full desktop (big/multiple monitors, full keyboard, etc).
It'll be interesting to see what happens post-lockdown.. will the people miss the benefits of "one remote = all remote" and have more empathy for remote people, or will we go back to the same old?
This is great for the participants, but absolute hell for everyone else in open offices, or even shared offices.
I enjoy team based offices, 7-10 man rooms. Even in there, this would probably be a nightmare unless you had this tech running in real time so you don't get microphone crosstalk/echo.
None the less, I really like the spirit of the system.
Not from my experience of 20 years ago setting up VC systems, biggest issue was video and making sure lighting was good, and plane wall behind (sky blue was good colour for that).
Audio wise, was many desk standing mic's (can't recall main brand) but was a few.
Did have one issue once with setting up a connection to a remote french company, was no audio from there end - turned out that the technician at the other end was sat on the end of the table in front of the camera and also was sat upon the mic that was on the table. Soon solved but still, most funny.
Back then we had VC systems that could roll into a room on a cart and worked well - picturTel IIRC being one solution back then and PolyCon being another that soon overtook them as well as doing wonderful conferencing microphones.
But as bandwidth got cheaper and more accessible, many meeting rooms that would be too noisy visually for VC became accessible and the need for dedicated rooms drifted away for more client usage.
Though audio from my experience back then was the easy part.
If we have beautiful, well-lit video feeds if every participant, but no one can hear what they're saying -- that's a deal breaker. The other way around, if we have clean, crisp audio from everyone and inconsistent video, at least the conversion can still move forward.
> And it's well-known that the "bowling alley" visual of meeting participants (camera at the end of a long conference table) isn't ideal. If each participant has their own laptop camera on themselves, it could be a vastly better experience for remote participants.
It seems to me that these days the simpler solution to both these problems is to just have people use airpods.
And a lot of meetings have most (e.g. 10) people in a single room, with another handful (e.g. 5) of remote participants.
15 people around the table in a conference room and each person wearing airpods (thus being connected to their own device) is an expensive solution with a lot of points of failure.
Not much: each participant has a pair of (amplitude, phase) values for each microphone. Filtering human voices and correlating sources to find the phase is not new.
Filtering sources isn't new, of course, and doing it manually fiddling around with a recording you already made is one thing.
But doing it in automatically in real-time on equipment with unknown characteristics that leaves zero ghost signal behind... is, yes, insanely hard.
The specific improvement Microsoft are touting is blind beamforming, without knowing where the microphones are located relative to each other. Regular beamforming is already in use in some products.
Think of all those shitty little video clips people take at a concert. Could all those be combined to make some high quality panoramic video? Probably a lot of other cool applications that I can't even comprehend for now. What a time to be alive.
People definitely have created very cool, close to seeming professionally-produced, full-length concert films out of tens of separate YouTube uploads. It takes a lot of editing skill though.
An actual 3D panorama is vastly harder, but remember Microsoft PhotoSynth? That rendered a 3D point cloud out of hundreds/thousands of tourist photos, and positioned the photos from where they were taken.
One of the applications for a thing like that would be creating 3D environments of concerts and other historical events that were fairly accurate from any angle, though, which could have some pretty interesting effects (could you imagine how interesting it would be if you could watch old concerts of dead artists, or a politician's speech from a hundred years ago, with 6 degrees of freedom, "accurate to the millimeter!" or something?) and outcomes and so on. Much more interesting historical record-keeping.
https://en.m.wikipedia.org/wiki/Awesome;_I_Fuckin'_Shot_That!
Also, it's been done by sound engineers for decades.
It's like webex - it turns on 24x7 microphone access "to detect nearby video devices"
This technology was already available in the 60s with Kalman filters.
The innovation here seems to be primarily about making this available in circumstances where it wouldn't otherwise be.
Gets down to matching a single sound and working out the timing of that sound from the multiple sources. Then you also need to factor in the frequency response as well.
That last part would be important to handle things like the table the devices are sat upon picking up vibrations from the desk. Remember that phones don't have a rubber base to isolate them from the table so any vibration of that surface would propagate into the device and microphone. Then the whole aspect of varying devices and with that, varying microphone quality and device housings. So calibrating at some level would be key for this to work, though doable and processing wise you could even run a master device and handle the processing there and remove the server aspect with some of the processing done upon each local device and passed onto the main device for correlating. Certainly some phones have the power to handle this type of affair to replace the server aspect. But that would be more work/effort and something that may well see later on. Though makes it harder to sell a bit of server processing software then.
Though one test I'd like to see this system handle would be how well it filters out those vibrations.
After all you don't want to hear somebody writing or putting a cup or other object down whilst somebody else is talking.
I'd also wonder what type of jitter tolerances they are working with across those devices and how that scales with devices/jitter - does jitter increase after so many devices.
However the initial greeting at start of the meeting would be good enough to cover that. Though some feedback and constant recalibration would be ideal and doable, That covers things like people entering the room and briefly changing the rooms acoustics with the door open briefly. Then somebody closes a blind and things like that, even somebody moving a coffee cup on the table would have (whilst small) an impact upon the acoustics. Though in that last instance, somebody moving a cap nearer a device would have a bigger impact upon that single source.
Though easiest way would be having a sound source on the main camera that did a simple frequency sweep - if you wanted to use a reference point sound source for calibration. You may even get away with single calibration then, though dynamic calibration and using the meeting itself to constantly recalibrate, whilst more effort, would give a better result.
But be interesting seeing this in action and how they handle aspects like that.
Indeed, thinking it thru you could have each device as it joins into the meeting do a calibration tone sweep that the other devices would pick up. That approach may well be better as you could get a more accurate map of all microphones in relation to each other that way. So initial login/join of the devices would handle that aspect nicely.
"While the idea sounds simple, it requires overcoming many technical challenges to be effective. The audio quality of devices varies significantly. The speech signals captured by different microphones are not aligned with each other. The number of devices and their relative positions are unknown. For these reasons and others, consolidating the information streams from multiple independent devices in a coherent way is much more complicated than it may seem. In fact, although the concept of ad hoc microphone arrays dates back to the beginning of this century, to our knowledge it has not been realized as a product or public prototype so far."
Thoughts:
There's something deep here, not with respect to microphones and speech transcription (although I wish Microsoft and whoever else attempts to wrestle with those problems the greatest of success!)
There's a related deep problem in physics here.
If we consider signals that emanate from outer space, let's say they're from the big bang, or heck, let's just say they're from one of our past-the-edge-of-this-solar-system satelites -- that wants to communicate back to earth.
Well, due to the incredible distances involved, the signal will get garbled in various ways...
So here's the $64,000 question:
When that signal from deep space gets garbled, isn't it possible that it turns into various other signals, at various different other frequencies and wavelengths?
In other words, space itself, over long distances, acts as a prism (not really, but as an easy way to wrap your mind around this concept), for radio, and other electromagnetic waves...
Now, if you want to reconstruct the orignal message at these long distances, you must be able to reconstruct garbled radio (and other em) waves, which are moving at different frequencies, and may even arrive at the destination at different rates of speed with various time shifts...
Basically, you've got to take those pieces -- move them to the correct frequency, time correct them, speed them up or slow them down, sync them, and overlay them -- to reconstruct the original message...
That's the greater question in physics -- the ability to do all of that, with em signals from a long way off in space...
The article referenced -- is the microphone/audio/slow speed equivalent -- of that larger problem...
https://github.com/introlab/odas https://github.com/introlab/manyears https://github.com/introlab/16SoundsUSB
Website of the team behind these:
In theory if the entire 3D layout and material properties were known known in advance you could get clear audio analytically. But reverse-engineering the 3D layout and materials from existing audio is essentially impossible.
So machine learning is used to find approximate solutions that work.
The idea was that at a gig loads of people would record and you could reconstruct a much better recording.
Edit: This would be cool if I trusted Microsoft to properly handle privacy.
