@hyc As you'd maybe know if you watched the Julian Krause video, this is intended to be used for professional audio production. In other words, it is assumed that you are recording and then applying a post-production step where you rebalance the sound. The advantage is that you can record and then adjust the sound afterward, instead of knowing how loud the thing is, setting a gain and recording. It potentially saves time. It's probably more for measuring very quiet things than very loud.

@hyc @mcc the idea is to change the gain down (or up, if recordng low volume sounds) in a daw afterwards. you're not gonna clip or lose any sounds, because the recording has such a high dynamic range. this makes it so you don't have to mess about with a recording devices gain before recording (the new Zoom field recorders like the H1e don't even have a gain knob for the microphone, because it isn't supoosed to be necessary.)

- posted by Lumi

I think there are even DAPs (and DACs too) which will play back such audio? I see some DACs sold on aliexpress for under $500 USD which use the AK4493 chipset which can do 32bit 784KHz PCM and even 512DSD (I have no idea what records that, since even Sony discontinued their DSD recorders, the ADC chipset in the Zoom UAC-232 only supports 256DSD as an example) playback supposedly.

But then, I already own a portable recorder which is capable of 32bit 384KHz recording, so Zoom's non portable UAC-232 which does 32bit 192KHz recording, doesn't seem too impressive to me personally.

It is at least cheap? (Maybe around $199 it seems from what I can find online) but higher bit rate and sample rate recorders (and playback systems) exist. They tend to cost a lot more than $200 though some are at least under $1000 USD. Going above 384KHz (for recording, not playback as playback devices are invariably more economical as far as I have observed) seems as if it was more than $2000 USD last I checked a couple of years ago, but presumably that will come down in time.

I agree though the "never clip again" thing is total BS.

Moreover, it's extremely rare still to find any place which sell lossless audio music that is better than 16 bit 44.1KHz.

Even Bandcamp which gets way too much love these days (especially after Epic bought them and fired everyone and basically pulled a complete dick anti-union move before selling them off again to Songtradr) has file size limitations.

So you'll encounter emails from Uwe Schmidt (behind various projects, e.g. Lassigue Bendthaus, Atom™, Señor Coconut, Stereonerds, etc.) apologizing for reducing the length of some of his higher bitrate recordings, so that they are available on Bandcamp in truncated form to fit into Bandcamp's file size limitations rather than not at all at higher bitrate. He has done the due diligence in his own recordings and I definitely hear a difference, but I've found I prefer to source his lossless audio more directly from his own website when possible as a result.

Most music producers, as far as I have observed, tend to use higher bit rate (even the marginal gains 24bit 96KHz as an example) for their "mastering" if at all since it gives them a little more wiggle room; but I still encounter some sticking to 16bit 44.1KHz and even huge commercial deejays have posted videos of themselves outputting shitty lossy MP3s to their USB sticks before going to play some sold out stadium shows, so I know that few actually give a crap about high bit rate lossless recording as nerdier audio sensitive types such as I do.

If I had to guess, whatever audio chip set (the https://www.akm.com/us/en/products/audio/audio-adc/ak5534vn/ by the looks of things thanks to Julian Krause's opening the device) Zoom is using in their UAC-232 just has 32bit 192KHz codecs as a given, so they might as well expose it to their users?

It's probably because even sourcing hardware audio chip sets that don't support such things in 2024 is increasingly rare.

If you look at the data sheet for the AK553x series ADCs, they claim to supports up to 32bit 784KHz sampling, so while I am not a Zoom customer, I would be wanting to know why they only expose 192KHz to users and not 384KHz or 784KHz.

There's even 64bit audio recording stuff out there; but I haven't owned any such gear personally and even if it's worth it (to someone like me, it probably would be, but I can't afford it currently), I can only imagine that the recording industry and distribution channels (and most likely, a lot of DAWs) aren't ready for it given that most of them barely seem ready for 24bit 96KHz, still.

CC: @mcc@mastodon.social

@mcc the cheaper way to go is to stick an AGC IC on the frontend with really wide reference rails and read the attenuation level from that as an inverse scaling factor when translating the linear values. but that's a bit more fraught and honestly doesn't seem all that beneficial versus just having an ADC do 24-bit PCM across wider rails. or 32-bit if you want fairly pointless number increases.

@mansr If 1.0 is "normal", and the effective true high range is in the range of something like 256.0, then from an end user UX perspective it might be inconvenient if every single signal were coming in at 1/256 power and your first step in the *normal* case was to bost by 256x.

@pedromj @foo @mcc like if I was designing a language I’d make you use an unsafe-style keyword escape hatch every time you wanted to force it to use a floating point, with compiler warnings explaining to them that it’s almost certainly not what they want because it almost certainly isn’t