jabbr

This is not a trivial topic. My impression after just a little bit of investigation is the contrary — a well implemented, for example, DSD256 capture will have significantly less higher ultrasonic noise than a DSD64 or PCM44 signal which is upsampled to DSD256 (because of noise shaping). The caveat is that various filters might be applied, so hard to compare apples to apples but perhaps @Miska who I believe has one of those sweet RME ADI-2 ADCs can chime in with some real data. Since we know that most all DACs internally upconvert incoming PCM44 to hi-bitrate SDM, then I assume all DACs need to properly deal with ultrasonic noise (hence the output filter of course)

I'd like to expand on the idea of overspec'ing @esldude -- yes by all means get the best microphones possible and the microphone should have a range well beyond the highest frequency you need to capture, similarly your interface: it should capture well outside the microphone range (as @Miska says), similarly your power supply: it should handle well out side your maximum draw, same for your amplifier, it should handle way more than the maximum SPL you listen to. same for CPU etc etc etc same for measuring equipment, we typically try to use equipment with 10x the resolution of the signal we are measuring The point is that most every system starts degrading at the limit of its stated performance so you get the best results if the system can handle more than you expect to throw at it. Overengineering is common sense. Of course if you don't have $$$ then you cut corners and live with it.

Well I'm tired of dealing with Espressobin V7 and picked up a NUC6CAYH as my wifi NAA. Loaded Ubuntu lowlatency-hwe-18.04 and its playing DSD512 really smoothly over 802.11ac wifi ... I'm leaving it dual boot with installed Windows 10 just in case I need windows for something.

I often use headphones but there is great merit to multichannel & room correction. Likewise in photography we have raw capture. Input profiles/convolutions and output profiles/convolutions ie for each device. One goal to preserve the 24 bit headroom is that we can do more in room processing and/or even non linear ops — for example as you age and Hf hearing drops off perhaps a nonlinear op might give you the same perceptual hearing — or even transfer “air” at 24 kHz into “mist” at 18 kHz etc 😉 In any case I don’t need anyone telling me that I never want anything higher resolution than 16/44.1 because you never know what may come 😉 😉

One of my main desires in delivered music, is to allow me to close my eyes and feel like I’m at a live performance. To promote the hallucination of “being there”. In other situations the delivered music is the performance. One of the most striking experiences I had at the Radiohead concert was seeing the complex electronic effects performed live in real-time. I had assumed these were multitrack studio creations.

The definition in the bloody dictionary was smudged. In any case since we are unable to agree on the definition of mathematically precise terms such as lossy and lossless — which themselves are also defined in dictionaries — then your own interpretation of the dictionary definition of typical (here I’ve defined the group as audiophile music given our presence on an audiophile site) will likely be at odds with the English language definitions as I use them.

I defer to the judgement of the mastering engineer, regarding processing and resolution. As I said: I want a copy of the Master when possible.

Ah, so you mean typical as in mainstream mastering engineers like Barry Diament?

🤣 didn’t even notice that meaning ... I was thinking of the nit-rate as the frequency that @mansr would nitpick a post.

Like it or not your “typical” recording of Lady Gaga is not the same as my “typical” recording and certainly not the same as a typical audiophile recording by any of the discussed audiophile labels. For mainstream releases I do like Analogue Productions as another example. But if you consider “typical” as average mainstream then the intention is as you say — indeed intended to sound best on earbuds via Spotify, or a car stereo or through a TV screen ... and no doubt highly processed as you say. Again my position is that if I was recorded and mastered at a certain nitrate, then that’s the rate I want my delivery. Again your typical is vastly different than mine so there we are.

The statement above belies a fundamental difference between your and my desired music and philosophy. No doubt we will never agree. I’m fact it is exactly my desired audio experience to listen to vocals sounding as close to the mic feed as possible. For me it is supposed to be the same (or at least give me as close to that illusion as is possible)

To expand: this issue is important to me because I firmly believe in the preservation of the recorded track in all its glory regardless of whether ultrasonics are audible. I would preserve the original raw delta-sigma ADC output tracks as well as the (hopefully) DXD master. Regardless of whether it is audible. The redbook/CD downconversion is not the same to me, regardless of whether the difference is always audible. I consider that downconversion to be a loss, hence my use of the term "lossy", regardless of whether the loss is always audible. Why? The issue of ultrasonics is not settled in my mind. No doubt there are fantastic CDs. But preservation of every bit of the recording preserves the possibilities of future as yet unknown technologies. That's common sense to me.

I am asking the question because when you assert that downconversion from 24/192 to 16/44.1 is lossless then you are asserting that the redbook/CD distribution is equivalent to the digital Master. Likewise with your example, you are asserting that one can recover a 24/96 digital master by upconverting a 16/44.1 CD.

True or false: a) Downconversion from 24/192 to 16/44.1 is frequently lossy? vs b) Downconversion from 24/192 to 16/44.1 is always lossless? My position, according to the common definition of “lossy” is that (a) is true and (b) is false.

Again, its not PCM which is itself lossy, rather the process used to convert one PCM format (bitdepth-samplerate) into another. A perfect infinite length stream could be Fourier transformed and a perfect brickwall filter applied but no stream is infinite and no brickwall filter is perfect. So ... the decision to use 16/44.1kHz is based on a "psychoacoustic" model of human hearing that claims that since the cochlea does not respond to tones above ~20 kHz, than bandlimiting the signal to 22 kHz will capture everything that is heard. This is an assumption folks. No need to argue whether it is a correct assumption but nonetheless it is based on a model of human hearing. Arguably the cochlea and human hearing system is better modelled with wavelets and so wavelet compression has been implemented. Is that "psychoacoustic" because different math is used (e.g. wavelet vs fourier?) The point is that all of these formats whether 16/44 PCM or MPEG or AAC are based on human hearing assumptions (admittedly some assumptions e.g. 20 kHz upper limit, are more substantiated than others) and all of these formats are based on assumptions about what is necessary to represent audio. My definition of losseless conversion between format A and format B is very precise: There must exist a pair of transforms such that given a file f, when transformed and then inverse transformed, results in a bit identical result. e.g. f = T-1(T(f)) where fB = T(fA) and fA = T-1(fB) So of course each format is lossy in a different way, and some are too lossy, and some losses aren't audible. Sure. Nonetheless downconverting a master to redbook/CD is typically lossy except where the source data was 16/44.1.