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24-bit/192kHz music downloads and why they make no sense (2012)

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211–220 of 274 posts

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#211
What's really interesting to do with all of these people arguing over audio formats (as always happens on HN) is to point a frontier model at this thread.

In a nutshell: nullc, rahimnathwani, zamadatix and vor_ know their shit, and geraldmcboing and PaulDavis are technicially correct but talking past each other. speak_on and TheOtherHobbes are confidently wrong.

And also: 44.1 kHz captures the entire human audible spectrum with room to spare, and 16-bit already goes beyond anything useful for listening. The higher resolution / sample size format is useful for production or archival purposes only.

The two main reasons why you hear a difference between the two formats: (1) it's likely a different master, (2) tiny gain differences in the signal (salesmen use this trick, but it's also easy to do it by mistake).

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#212

Earlier quoted context omitted.

No, the central point is that the analog signal handling before AD introduces vastly more "artifacts" than the AD or DA does. In addition, nobody cares about "measurable" artifacts (or rather, they should not). What matters are "audible" artifacts. We have measuring equipment that is vastly more sensitive than human ears (e.g. your recording equipment that can pick up signals far above 22kHz). What's measurable is no…

Hm, no. The discussion was never about analog artifacts vs AD conversion artifacts. Both are present. And not sure why you use "artifacts", do you not believe the artifacts are real? How can the lowpass filter not introduce artifacts? And absolutely - I blind tested coverters extensively. Mbox2, Black Lion Audio upgraded converters, UA, Prism.

Note: blind testing is not double blind testing. Scientists evolved double blind for a reason: blind testing doesn't remove bias.

Yes, the discussion was "never about analog vs AD". But my point is that I see little point wasting time on one set of artifacts (in the digital realm) that are tiny compared to those introduced in the analog realm. If there's a mouse and an elephant about to enter your home, you focus on the elephant, no?

The big difference, of course, is that "everyone" has convinced themselves that most/all of the analog artifacts, as big as they are, are somehow "tasteful" or "artistic", whereas the digital ones are just "math errors". I don't think is too helpful.

And look, if lots of people could get through double blind tests and still show they can hear aliasing or whatever the digital artifact du jour is, then I'd say "yes, absolutely, we need to be very aware of this and do everything we can to reduce or eliminate it". But as far as I can tell, this just isn't the case.

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#213
post #204

Earlier quoted context omitted.

It's discontinuous. You have some low noise amplifier. There is a signal. You split it. The result on each side has >=1 bit worse noise floor, probably somewhat worse as we're not using superconductors :P-- as you expect: there is no free lunch. Now: take one copy and attenuate it 48dB, further degrading its noise floor. Sample both. The attenuated copy is mostly useless, except when the input goes high enough that i…

> So the tradeoff is that you lose a small amount of noise floor constantly-- out at the 20th bit So, basically, no better than the best AD converters we already have? My understanding of the fundamental limit to AD performance is that the brownian noise level is around the 22nd bit level. So even if you come up with techniques to successfully measure down to that level, you're basically picking up .. inevitable, irr…

Yeah people describe the benefits incompletely/inaccurately. This approach has a worse theoretical SNR, but an effect that improves the delivered SNR in real usage: Without the clipping protection the user would massively lower the gain, hurting the SNR.

A common experience for someone doing field recording of performers (my experience is music) is you twiddle your setup to get the gains reasonably high to get good SNR even for quiet parts. ... and then you record the actual performance, and you find that the tuba player really got into it for the real performance and the new peaks are 10dB over where they were in the practice. And now your recording is screwed up with a bunch of hard clipping you have to deal with. So then experience tells you in the future to take whatever you thought was safe and lower gains another dozen db.

The multi-ranged recorders eliminate that problem and the result is that you don't need to use precautionary gains, and you get a better SNR in your recordings. You probably don't need to adjust gains at all: The gain can be whatever makes the self-noise of the microphone dominate the SNR of the process, ... which would be too high for the loudest samples, but the clipping handling deals with that.

The samples that need to use the extended range have worse SNR (and probably poor linearity due to mismatches between the converters), but human hearing is much less critical to noise with loud signals anyways.

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#214

I decided to test for myself, downloaded Lacinato ABX and tested a 32-bit 352.8Khz flac I had lying around, to the same file downsampled to 16-bit 44.1KHz. I couldn't tell any difference. Then I tried 192k mp3... still no difference. Couldn't reliably differentiate 128 or 64kbps mp3 either. I had to go down to 32k before I could be certain which was which, and even then I still had to listen carefully. Think I need t…

same here. seems all the years of q-tip use is saving me money by not needing to buy expensive Hi-End Audio gear.

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#215

Earlier quoted context omitted.

Hm, no. The discussion was never about analog artifacts vs AD conversion artifacts. Both are present. And not sure why you use "artifacts", do you not believe the artifacts are real? How can the lowpass filter not introduce artifacts? And absolutely - I blind tested coverters extensively. Mbox2, Black Lion Audio upgraded converters, UA, Prism.

Note: blind testing is not double blind testing. Scientists evolved double blind for a reason: blind testing doesn't remove bias. Yes, the discussion was "never about analog vs AD". But my point is that I see little point wasting time on one set of artifacts (in the digital realm) that are tiny compared to those introduced in the analog realm. If there's a mouse and an elephant about to enter your home, you focus on…

This is a more philosophical take… And I totally agree with you. I mix at 16/44.1 just for the record. I do not buy into the idea of gold plated connectors or 96 kHz mixing. My point was never about quality - I can hear the difference (the point!), doesn't mean for me personally > 44.1 is "better" or "worse".

To your main point: yes, all artifacts are just our learned, cultural, developed preferences. In the exact same way major/minor thirds were considered dissonant just a few hundred years ago - it's all a learned perception, not an absolute judgment.

I would go even further, doesn't matter whether people perceive aliasing as a major issue, it's no different from the U47 "warmth". You can't afford this, probably, as a software developer in a way, but at the most fundamental level any sound's - or artifact's - judgment is based on our our current diagram of "sounds nice" vs "sounds bad".

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#216

I decided to test for myself, downloaded Lacinato ABX and tested a 32-bit 352.8Khz flac I had lying around, to the same file downsampled to 16-bit 44.1KHz. I couldn't tell any difference. Then I tried 192k mp3... still no difference. Couldn't reliably differentiate 128 or 64kbps mp3 either. I had to go down to 32k before I could be certain which was which, and even then I still had to listen carefully. Think I need t…

Yeaaaaaaaaaahhhh.... you might be a tiny bit deaf. OTOH, we know nothing of your audio equipment nor how its setup.

HiFiMan Sundara headphones, focusrite scarlett 2i2 interface. Although, it turns out the Scarlett is set to 48KHz anyway, and I can't seem to change it easily under linux. Not that it seems to matter for my ears, lol.

EDIT: Did some more ABX testing with a CD-quality track that I'm much more familiar with ('Introduction' from the Mirrors Edge soundtrack, which has been my go-to for comparing audio gear for the last decade). I could sometimes distinguish 128k mp3 this time, though interestingly, I got it consistently wrong rather than right. For some reason the compressed version seems to be my preference. Dropping to 96k mp3, I got it right 100% of the time - though only because there was a very noticeable difference in the stereo positioning of the first sound, rather than a difference in the quality of the sound. I think if it were mono I would still be unable to tell.

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#217

Earlier quoted context omitted.

Correct. I've paid for Tidal for a decade because I just like the peace of mind that it's closer to the original recording. I'm sure it's mostly placebo, but I like it.

I tried Tidal nearly a decade ago, and the audible fluttering effect caused by their audio watermarking totally ruined certain types of music, like choral recordings, strings and such. It was obviously apparent on $20 ear buds driven by any device, far beyond the more stereotypical audiophile gripes. I opened a support ticket but they never responded. After that it was difficult to take their lossless claims seriousl…

They don't have anything like that anymore

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#218

Earlier quoted context omitted.

Correct. I've paid for Tidal for a decade because I just like the peace of mind that it's closer to the original recording. I'm sure it's mostly placebo, but I like it.

The original recording of almost all music on Tidal was done with equipment that was very, very far from the 192kHz "fidelity" it claims.

I still prefer lossless over MP3

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#219

I cannot hear the difference between 16/44.1 (and by extension, 16/48) and High-Res Content generally, be they HDCD, SACD, or just straight-up Masters from Qobuz. This is on multiple sets of equipment, ranging from El Cheapo earbuds all the way to HD800 cans and full-fledged tower speakers being bi-amped. That’s not why I go for High-Res stuff, though. It’s all about archival, at least for me. With a 24/192 Master in…

> I absolutely prefer anything with classic instruments (Jazz, Classical) in higher-quality formats High-dynamic-range material benefits from lots of bits.

This. It may be a niche. But music that works with volume as part of the composition can be amazing.

But most music today has heavy compressors in the pipeline that kills dynamic range in favor of allowing you to hear almost even whispers, even in traffic or a city with ear pods.

But if you're from the first group, as you said it's more noticable the benefits of having better codecs and bit depth vs heavily compressed top billboard songs where even listening the master track from the studio, falls into diminishing returns.

Re: 24-bit/192kHz music downloads and why they make no sense (2012)

#220

Earlier quoted context omitted.

"Nobody uses 32 bit float for recording" - you are just displaying total ignorance here.

My comment should have been more emphatic that: nobody uses AD converters that generate 32 bit floating point values natively when recording, or anywhere close to the resolution that format implies. I am extremely aware that as a data format in DAWs and other recorders, 32 bit floating point is completely common.

While the best that ADCs can provide is linear 24-bit audio samples, the following audio processing is better done after converting the samples to FP32, and keeping this format until the final 16-bit encoded audio suitable for listening is generated.

For the same reason, video processing is preferably done on FP16 samples of the color components even if both the input ADCs and the output video signal may use only 10-bit or 12-bit per sample, at most.

Moreover, most high-resolution audio ADCs do not really sample the input audio at a 24-bit resolution, but they use only a sigma-delta method where the actual samples have only a few bits, possibly only even 1 bit.

Then DSP techniques are used to convert the audio stream with a high sampling frequency and a low resolution per sample into an audio stream with a low sampling frequency and a high resolution per sample, which is the external output of the ADC.

If you had access to the raw audio bit stream as actually captured by the ADC, you could modify the decimation algorithm to really output FP32 samples, though no existent ADC could actually have a so high dynamic range (except if the output bandwidth would be reduced a lot, to filter the input noise).

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