Back in high school, I built (with some parental assistance) an apparatus to measure how quickly the pressure would drop (in a pressurized cylinder) when a very small hole allowed air to leak out.

Turns out, not only can you measure temperature that way, but can extrapolate the graph out to find absolute zero (IIRC my result was out by about 20 kelvin, which I think is pretty damn good for a high-school-garage project).

>>>> Reminds me of the electronics adage: "all sensors are temperature sensors, some measure other things as well."

A corollary that's one of my rules to live by: Never measure anything over time without also measuring the ambient temperature.

I love these kind of inadvertent measurements. One of my favorite examples is that a sufficiently accurate IMU can get you relatively accurate longitude measurements from the Coriolis effect.

Is there one saying “All electronic devices are smoke machines, some can compute too”?

I just learned how the Duracell Powercheck© worked, which was done with temperature.

https://youtu.be/zsA3X40nz9w?si=oGg2wdUlLXSDxpsN

a colleague of mine spent months analysing fluctuations in narrow band signal from a geophone only for a more senior colleague to get fed up with it and demonstrated that actually the fluctuations simply correlate with the air temperature and do so within the spec sheets reported temperature tolerance.

> Reminds me of the electronics adage: "all sensors are temperature sensors, some measure other things as well."

I wanna say that’s a Bob Pease quote but I can’t find an attribution to it.

The highest grade gauge blocks use laser interferometry from Mitutoyo have a measured coefficient of thermal expansion AND a uncertainty of that coefficient. And they have a size variance of plus or minus 30nm. That is only about 410 oxygen atoms.

Oh yeah. I realised this the day I discovered my fancy digital SLR was a thermometer: https://entropicthoughts.com/does-my-dslr-have-dead-pixels

It does act as a thermometer, if and only if the altitude remains constant. The speed of sound fluctuates with both temperature and altitude

Reminds me of Intellectual Venture's Optical Fence developed to track and kill mosquitoes with short laser pulses.

As a side-effect of the precision needed to spatially locate the mosquitoes, they could detect different wing beat frequencies that allowed target discrimination by sex and species.

I did a similar project at 18. Needless to say I didn't have enough HW and SW skills to do much since I implemented the most naive form of the TDOA algorithms as well as the most inefficient way of estimating the time difference through cross correlation. I still learnt a lot and it led me to eventually getting a PhD in SAR systems, which are actually beamformers using the movement of the platform instead of an array

What were the results of your study? I’ve heard that bat lungs are so sensitive that when they fly across the pressure differential of large turbines their capillaries basically explode

I would love to do something like that to track the bats in my garden, how feasible would it be for an amateur to do as a personal project? Any good references on where to start.

A nice mention about this is the outstanding and quiet work of the Cosys-Lab of the University of Antwerp. They once put a microphone array below a scorpio, and showed how bats moved their ultrasonic beam to scan for a scorpio. Incredible stuff [0].

[0]: https://www.youtube.com/watch?v=57ScSPWhGqU

I had no idea they were mammals until this comment. I thought they were furry birds!

That sounds super interesting. Is there a write up somewhere of the project?

That sounds like a fun project. Was it part of a research grant?

> bats (the flying mammal)

As opposed to?

Honestly, that sounds like amazing work. I wish I could afford to get out of enterprise software engineering and just do academic software development like that.

(OP here) tverbeure hit most of the main points, but mostly cost ($2/mic vs $0.5/mic adds up when there are 192 microphones), difficulty of finding things with enough i2s interfaces (even with 16 way daisy chaining, thats still more than most/all things will have). The FPGA/custom hardware was part of the fun as well!

Not OP, but I looked in to this a few years ago. It was more expensive then, and only went to 20 kHz. Higher frequencies are helpful if you're listening for the hiss of leaking gas, or corona discharge of an electric arc.

The Orin has 6xI2S ports internally, so that would work up to 16*6 = 96 microphones, which is a good number. But it looks like maybe only 3 are brought out & on different dev board connectors [1]? As with a lot of design, the devil is in the details. An FPGA could be easier to configure if you need more than 96 microphones.

My notes:

ICS-52000 $3.50, 20 kHz

ICS-41350 $1.05, 40 kHz

SPH0641LU4H-1 $1.45, 80 kHz+

[1] https://docs.nvidia.com/jetson/archives/r34.1/DeveloperGuide...

I've considered making a phased array myself, but never got around to sending out the PCB. But here are two reasons by I2S is not the best option:

* I2S requires 3 instead of the 2 pins of PDM. However, in the datasheet that you provided, it shows how you can daisy-chain microphones which is really cool (even if not standard I2S.) So that argument goes away.

* PDM gives you access to way higher sample rates which in turns gives you more flexibility in choosing the delay for a delay-and-sum operation. For example, if the PDM clock is 2MHz, you could theoretically delay with a precision of 0.5us. In practice, you'll do that with lower precision, but with I2S, the clock will typically max out at 192kHz.

* PDM microphones then do be cheaper.

I am very interested in how small these arrays can be. From talking with a friend with cochlear implants, I would assume this could help dramatically with the right signal processing to help him hear.

I do my PhD in in-air ultrasound with phased arrays and talk to the medical guys at conferences/labs that we talk to and it's soooo much harder in solids/liquids. The frequency is significantly higher, think 1-10MHz instead of like 40khz, so any normal electronics are out the window.

One problem is that the speed of sound is not constant (or approximately constant) across the bandwidth you're interested in when the sound wave is traveling through solids and liquids.

Then, why not be a grad student again?

I may be the FUS grad student you seek. Reach out via profile email if you want to chat. Cheers!

Medical applications would presumably require contact coupling and not through air?

For the hard of hearing like me the killer application would be live transcription in a noisy setting like a meetup or party, with source separation and grouping of speech from different speakers. Could be life-changing.

(Android's Live Transcribe is very good now but doesn't even try to separate which words are from different speakers.)

I believe modern macbook pro’s already have multiple microphones that probably do some phase-array magic.

This is known as the Cocktail Party Problem. It turns out or brains do an incredible amount of processing to allow us to understand a person talking to us in a noisy room.

https://en.wikipedia.org/wiki/Cocktail_party_effect?wprov=sf...

In general the position of the microphones in space must be known precisely for the phase shifting math to be done well, and also the clocks on the phones would need to be in sync at high precision like 10x the highest frequency sound you're picking up. In other words within 10s of thousands of a second. Also if the array mic locations is not a simple straight line, circle, or other simple geometry the computer code (ie. math) to milk out an improved signal becomes very difficult.

It's already kind of implemented.

Xilinx tool chain is also no-cost.

You might look into OBS and/or VoiceMeeter to see how streamers selectively route audio while livestreaming/recording video/audio streams.

https://obsproject.com/

https://voicemeeter.com/

Loud show noise and your online friends' nearby audio is going to be reflected around the room as well as off of your bodies.

What you want isn't microphone or beamforming tech, it's echo cancellation the same as every videoconferencing software uses.

You just need to feed the show audio and friend audio in, and apply echo cancellation to each.

From the article "The simplest method of beamforming is delay-and-sum (DAS)". Measure distance from a point (couch) to each microphone, delay the signal in time domain by the time the sound takes to travel from point (couch) to microphone, and add up the signals. Pretty trivial. Basically you want the microphones receive the couch signal at the same time, even though they are different distances away.

Make sure there is enough variation in microphone distances for this method to be effective.

Pcb manufacturing is cheap. I put 20 parts 1.5 inch by 24 inch into pcbway and ended up with final delivered cost of 240 dollars.

Not having to deal with wiring that many individual boards and all days of headaches tracking down issues is well worth it in my book.

I believe it can, there's a demo under the "Directional Audio" section, unless I misunderstand you.

I’m still sad these didn’t become a thing. I don’t need a 48MP camera phone. No seriously. I do not.

A similar technique is very popular in industrial automation to spot leaks in compressed air pipes and their connections from far away. These leaks are extremely loud in the ultrasonic range. It's overlayed with a camera picture.

That's ultra expensive gear.

I've always wanted this for videoconferencing room. A microphone array around the screen should be able to dynamically focus on the active talkers and cancel out background noise and echos to get much better sound quality that the muddy crap we usually get.

If there were a speaker array around the screens too, you might be able to localize the audio for each person so that it seems like the sound is coming from where their head is on the screen.

I wish could rent one to figure out which device in my office has a squealing capacitor. I can hear it well enough to be driven crazy by it, but not well enough to find it. I start disconnecting things to narrow it down but then convince myself that it's my ears ringing.

I'm unsure if I'll age out of this problem, or if worse hearing will just recreate it at different thresholds.

The tech is beamforming .. the applications are AV conferencing, camera tracking, voice lift, or sound reinforcement

At a rough guess from the audio samples, that array is producing an acceptance angle much narrower than any Soundfield mic is capable of. The noise source is only 45 degrees off-axis; I'd say any first-order microphone polar pattern (i.e. those a Soundfield mic is capable of) would capture more of the noise than is demonstrated here.

Of course, you can improve on the rejection of off-axis sound by instead using a microphone with a more specialized polar patten (e.g. a shotgun mic), but then you lose the property of the pattern being steerable merely by signal processing.

Lastly, such an array of dirt cheap pressure sensitive mic capsules with some clever computation behind them strikes me as the sort of thing you could throw Moore's law at, if you could justify the quantity. Whereas, Soundfield mics don't make much sense unless you're working with very precisely machined pressure-gradient capsules.

Still, I get the feeling it'll be a while yet before this technique starts looking viable for audio production work, but it's very interesting.

This beamforming effect only works well when each sensor is getting a dramatic enough "different angle" on the signal that each one can use phase shifting to cancel out other noise, but with a laser there's not really any noise to cancel out (i mean you're just monitoring a vibrational spot on a window), and you also don't have a far enough "different angle" to shine from, if you're monitoring from one spot.

However having multiple lasers from multiple different locations might be able to create an improved signal if all signals are averaged, but it wouldn't really be due to the phase shifting that's used in beamforming.

Didn't Israeli students show that you can recover audio from the vibrations of bulb filament with a fast photo diode?

I'd test that with a CCD line sensor plus a wide aperture lens and reading it out with 8kHz. Then you have 128 audio pixels that can cover an entire city.

afaik, it really depends on the spatial structure of the audio field.

think nyquist sampling rates, applied to space, and you can't apply a low-pass filter just because you don't care about higher-order signals. that means that for any given audio environment, there will be some "spatial spectrum" of signal, and you need to sample it densely enough to avoid aliasing.

Because the distance between the mics needs to be 1) large and 2) consistent. It would work with a grid but the mics near the middle would be "underutilized" (not maximally taken advantage of), and also in a grid the mathematics is horrendous, but with a circle it's simple.

(OP here) Primary reason is that you can make a big array with only 2 boards, a small board in the middle and a bunch of long boards around it.

Radial pattern of linear arrays with exponential spacing should also be pretty close to optimal for the distribution of pairwise microphone distances to maximize the gain with a fixed number of microphones.