But regardless here the methodology is very weak, just playing a sine sweep with a spectrum recorder open and eyeballing the frequency magnitudes.
Obviously this will only work if there's a passive crossover network, and nothing active in the speakers.
This is sadly not very well documented process, but some open source implementations exist. E.g. https://github.com/maj4e/pyrirtool
They were talking about a home. /s
Like with 2 of each, or 3 of each, where you play the same waveform through every possible pair of speaker and microphone, you can solve some kind of system of matrix equations to determine the only possible combination of responsiveness at each device at each frequency?
Or do you just need a reference microphone with known characteristics, period, end of story, because math can't do it?
(Obviously from a practical perspective you want the reference microphone... I'm just curious about in theory.)
You don't do it every day, which is why an outfit like Bruel & Kjaer can charge a lot for their gear. ;-)
However, if you ignore tolerances and assume that every microphone of a given model number has equal response, then it's simply a matter of having that known response information available, similar to a hypothetical brand of ruler being known for coming up short.
I'm guessing that there must be a way to either cancel out skews and work around this, but I guess it's just easier to start with a calibrated microphone
Measurement mics (the professionally targeted kind) can with some models be purchased from the manufacturer with a calibration file that can be used by various audio measurement software packages. There are also places that will calibrate your mics for you. In the end what this really means is just figuring out the frequency response characteristics of a given mic relative to a known source. This could just be a case of “we match it to this mic over here.” As long as all the mics someone is using are calibrated to the same standard then it’s less important that they’re perfectly flat in frequency response. I have some really good mics but the bulk of my measurement mics are inexpensive and not calibrated. They work fine though because they all measure close enough to the same that once you figure in all the external parts of a system (the room for instance) the mics minor variations become insignificant. The important thing is that I can put 8 mics around a room and pull reliable data to tune the room and make it sound musical. At the end of the day all the measurement stuff is just there to help you scientifically quantify why a room doesn’t sound right and then you can fix it. Like the last room I tuned that had 2 blown HF drivers and the polarity flipped on some LF drivers. The ears easily said there was a problem. The tools told me right away what they were. Sure I could have found them without the tools and by tweaking settings till everything was right, but the proper tools (and knowing how to use them) makes a huge difference.
Also, generally microphones are sensitive to orientation, so to have repeatable results be sure to control that. Position in a room is also essential for repeatability.
I have spent a lot of time measuring rooms and you can't underestimate how they can mess with frequencies, but it's also true that good speakers sound pretty good in almost any room, and bad ones will sound bad.
Going the USB route sidesteps the separate DACs, which either add to the problem or run into the thousands.
Why not for it's always interesting to experiment but one is lossy (all bluetooth codecs are lossy AFAIK) and the other is analog. Probably on top of an already lossy source too. They "sound" the same the same way a pixelized color print of Mona Lisa next to the real Mona Lisa looks identical if you're far enough. You may or may not be able to tell the difference but lossy Bluetooth and analog aux-in don't sound the same.
I decided to go for a simple setup: a Yamaha fully integrated amp that does it all, including a network streamer. And I stream from Qobuz (lossless streaming) and from my own collection of CDs I ripped to FLAC (lossless and bit-perfect rips even though you're ripping from an audio CD, verified with an online DB of hashes from other people who did ripped the same CDs).
So I know that up to amp, it's all lossless. Then the amp does its magic.
It's simple really: even though I can't tell the difference between a pixelized color print of Mona Lisa and the real thing from far enough, I'd still prefer to know I'm actually looking at the real thing.
A Qobuz (lossless) subscription doesn't cost more than a Spotify one (lossy although they announced they'd move to lossless IIRC).
It's 2024: FLAC files are tiny compared to, say, even just a 1080p movie. Bandwith is plenty to stream lossless.
Why even bother with lossy? Lossy audio is tech from a quarter of a century ago.
I know I'm rehashing the same argument that has been had around and around for decades but - because it often doesn't matter. In 2024 a 320Kbps or whatever high quality lossy source over a recent bluetooth codec into a Chinesium amp + DAC into a mid-range $500 pair of speakers sounds _awesome_ and the amount of time and money you spend going above that may be a fun hobby but it's really not worth the effort for most people.
I overlaid those two images [0], and they seem significantly (though not hugely) different to me.
Wouldn't speculate as to why that is though, without checking the consistency of passes with the same setup.
0 - https://imgur.com/a/7GUSmPW (with hue shift for comparison)
It’s not my desire to slam the author but it’s also important that people don’t take this correct methodology.
`Please don't fulminate. Please don't sneer, including at the rest of the community.`
As a curious person with a degree in Electrical Engineering, and one in Computer Science, very interested in quality audio, etc, I know what you mean.
But please, don't be the person who kills curiousity. This person was curious, did some experiments, and posted the result. If you have time, write some of the major ways that such a measurement is inadequate. If not, link to some resources.
`Please don't post shallow dismissals, especially of other people's work. A good critical comment teaches us something.`
That aside though, I did mention a number of issues with the write up. Anyone interested is welcome to dig into them and come back with questions if they desire. This is not however an introductory level topic and I elected to not try to thumb tap out a distillation of a very advanced and multi-faceted topic here.
I would not expect someone with graduate level qualifications and years of experience in software development to try to distill their work down to my level either.
Everything was implemented using transistors, so it involved a lot of calculations, and simulation in LTSpice.
It was a very cool project that spanned multiple disciplines as we built a phased microphone array, a system to tilt and rotate the speaker, and programmable logic and software to generate and analyze signals.
It was proof-of-concept quality, but was later made real and reduced analysis time from dozens of hours to about 45 minutes. Two of the project members were even hired by the company.
When you use Bluetooth, your speaker is functioning as the DAC (digital to analog converter), but when you use the aux, your computer is functioning as the DAC and also amplifying the signal, so it's reasonable to expect them to sound different.
moreover, i'm not sure if/how something like THD relates to accuracy of reconstructing more complex naturalistic (i.e., non-sine wave) signals.
LOL. Why is this even on the front page? He measured it with a phone's mic, then he goes on to "eyeball" the results, which are clearly different even to the eye. He then declares them the same. From top to bottom, everything is wrong with this.
I used to be in-principle anti-EQ, but the AutoEQ project for headphones completely changed my opinion (though obviously headphones are far easier to EQ).
You can still tell that from the coloring of a local reference mic.
I don't even know where to begin addressing everything that's wrong with this article, but assuming linearity isn't one of those errors.
Stopped reading
> The music video is fantastic too and so, so beautiful except for the rap part which I don't like.
But that's fine for microphones -- the question here isn't to determine their absolute volume, which is of course unsolvable. It's to determine the relative "volume" (response) at each frequency. It's the shape of the curve that matters, not its offset.
And again, I'm not looking for a practical solution (like getting the info from a manufacturer) -- I'm just curious about it in theory. If it's inherently solvable or not.