Comment by saltcured
4 days ago
Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart.
I also spent a lot of time ripping my old CDs to FLAC and trying different MP3 and AAC encoder settings to get playback that felt transparent enough to me. I could never tolerate Sirius/XM radio streaming due to the horrid compression I heard with every futile attempt. I still seem to have more sensitive hearing than most people around me, but in my 50s I know it isn't what it once was.
I never had huge budgets, but did strive for hi-fi in my limited ways. I used things like toslink and HDMI to send raw PCM data from Linux to my Yamaha A/V receiver's DACs + amplifier to drive somewhat nice Polk tower speakers. But then COVID-19 happened, and this stuff was packed up to move house.
Nowadays, music playback is streaming with mundane "subwoofer + satellite" PC speakers or MP3 playback with a mini-SD card permanently parked in my car's infotainment system.
> Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart.
As referenced in the article, a common explanation for those audible differences is that the high-resolution version of the album is sourced from a different master.
> As referenced in the article, a common explanation for those audible differences is that the high-resolution version of the album is sourced from a different master.
In this case, it was my brother's own 24/192 recording, down-mixed by him to CD format with the intent that it be transparent. I believe he said his software was supposed to be dithering, but this was ~25 years ago and I can't really confirm the details anymore.
Even more likely, high frequency ringing in the higher res file, caused by the converters, has the same effect analog distortion via tubes does creating the perception of clarity where there is none.
No one can hear the difference between properly mastered high res files. I will happily put money on it.
This is easy to disprove by downsampling from a 24/192 source to 16/44.1 Even if the downsampling is (close to) ideal there are obvious differences.
In fact if you can't hear the difference between 24/192 and 16/44.1 you shouldn't be working in audio. (Doesn't apply to consumers. Does apply to musicians and engineers.)
It's like being colour blind.
And if you don't understand the math behind quantisation, you shouldn't be posting pseudo-scientific videos where you use an oscilloscope and a cheap spectrum analyser - both tools with very limited resolution - to "prove" your point.
16 bit isn't enough for hard, objective reasons. One is that the noise spectrum of quantisation is not simple. Most people assume it's something close to plain white noise, but it really isn't. It's actually a very complex spectrum with some prominent peaks at specific subdivisions of the sample rate. Those frequency peaks are significantly above audibility. 24-bit quantisation shrinks them below audibility.
The other is that most people can hear dither/noise-shaping at 16-bits. That adds a single bit of noise which should - if you're being very literal - be far below the threshold of audibility. But it clearly isn't.
These two facts are related.
The more complex reason is that listening is an active perceptual process. The brain does a huge amount of processing to separate sources and place them in a perceptual field which includes information about perceived object type, distance, and ambience cues. Some of those cues are very quiet, and we don't hear them linearly.
So using sine waves as some kind of perceptual reference for audibility is nonsensical. We hear much more complex signals in an active way, and if there's information missing in the quiet parts - which there is with limited quantisation - then the signal simply isn't accurate.
I agree with most of your points, but saying you shouldn't work in audio if you can't tell the difference between 192khz and 44.1khz is a bit elitist imo. And saying you're color blind if you can't tell the difference is like saying you're blind if you don't have 20/20 vision and shouldn't draw. You can always use meters to check for aliasing artifacts.
It's not like all of your samples and virtual instruments are 192khz or even 96k. Many are 48khz or even 44.1k.
I think there are many cases where people never need to go above 44.1khz unless you maybe have saturation on the master bus. I agree that good dithering is important though and think that there hasn't been enough research on that so far.
> 16 bit isn't enough for hard, objective reasons. One is that the noise spectrum of quantisation is not simple. Most people assume it's something close to plain white noise, but it really isn't. It's actually a very complex spectrum with some prominent peaks at specific subdivisions of the sample rate.
What you are describing is the result of blunt truncation. If you use the most basic (“uniform” or “rectangular” a.k.a. “RPDF”) dither, the spectrum is in fact flat, as demonstrated by the video you are likely alluding to and calling “pseudoscientific” (https://youtu.be/cIQ9IXSUzuM?t=12m50s). If you sum two uniform dithers together, you get what pretty much everyone uses (“triangular” or “TPDF” dither) which, in addition to decorrelating the mean quantisation error from the signal, also decorrelates the standard deviation, eliminating noise modulation and leaving a correlation only in still higher-order moments like skewness and kurtosis.
You can even try it for yourself with SoX. Find a 24-bit track, quantise it with dither to 16-bit, calculate the difference between both tracks, blow up the difference and take its spectrogram and it will be completely flat. Or listen to the difference (mind the volume) and see if you can make out anything meaningful.
And then remember that this difference would normally sit at roughly -93 dB FS, so to hear it in a typical room, you would have to be listening at deafening levels. You claim that it “clearly isn’t” below the threshold of audibility but it’s not clear how you arrived at that conclusion. You then claim that the audibility of that noise floor is somehow related to what you said before about the effects of undithered quantisation, even though those effects stop being relevant the moment you apply any sort of dither.
> We hear much more complex signals in an active way, and if there's information missing in the quiet parts - which there is with limited quantisation - then the signal simply isn't accurate.
It’s not missing. You can do a similar test where you “bury” your source material in the 16-bit dither noise floor, blow it up again, and you’ll be able to detect it under the noise.
But surely the difference is 16 vs 24 and nothing to do with the sample rate?
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One possibility (pure speculation) is a bad antialiasing filter. The Nyquist frequency at 44.1ksps is 22.05kHz, which is only ~10% above the audible band. This means that you need a rather sharp filter both when downmixing and when playing to avoid potentially audible aliasing into the audible band or attenuation within the audible band.
If you look at a site like audiosciencereview.com and pull up measurements of a DAC or ADC, you can find graphs of the antialiasing filter response. Some are great and some are not.
One could think of 16/44.1 PCM as being a codec that is potentially perfect but requiring some degree of care to encode and decode correctly.
That would be how you'd go about telling, sure enough. You can't go by 'frequencies' or distortions or anything like that, these analog departures from convincing reality aren't how digital failings manifest.
You try to hear the brickwall by the muffled, enclosed quality and possibly by the weird pre-ring blurriness of the filter making things sound more vague than they have to be, and you hear the truncation not because it is audible 'distortion' as we know it, but because depth collapses and it sounds like it's coming from the speakers and not being a separate space behind/around the speakers. At no point will it be the most glaringly obvious thing but it'll never be 'distortions' as we imagine them, it's more a 'pod people' lack of personality thing.
Like a much subtler version of listening to AI music :)
I'm quite happy with 24/96 as suitable overkill for anything I might want to hear or do. Neil Young went hard on the proposition that 192 was necessary. Sold the Ponoplayer, I had one but it died on me, battery failed eventually. It really did sound awesome beyond just about any other listening device I've ever heard…
24 > 16 is not debatable. Sample rates are more complex because then higher the clock rate the more you get distortions from jitter and the design of the DAC/ADC. Most converters introduce different artefacts at different sample rates, especially at the prosumer end, so you're not comparing like for like.
The last couple of generations of converters have gotten a lot better, so 192kHz today is likely to sound cleaner and smoother than it did ten years ago, where there was a good chance the clock was quite jittery.
Personally I don't think it's worth the extra bandwidth for playback, but I can understand why some people might want it.
Generally all of these "debates" come down to people who think math > circuitry. All real designs are imperfect trade-offs. They all have issues, and arguing as if converters are perfect when they never are, and the imperfections can be benched objectively, is... not very scientific.
>Generally all of these "debates" come down to people who think math > circuitry. All real designs are imperfect trade-offs. They all have issues, and arguing as if converters are perfect when they never are, and the imperfections can be benched objectively, is... not very scientific.
There is one purely objective benchmark: a true blind test. You can believe if something is different or not, but if nobody's capably of hearing the difference, does it matter?
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This is an extremely hard comparison to do well. I'll give a few examples as to why:
Small differences in gain are ABX able much more readily than differences in noise at the 16 vs 24 bit level. So if the signal chain gives even a small difference in gain between the samples that's what you'll track. A reasonable conversion path to 16 bits for mastering will also apply dithering and some kind of brickwall limiting (you have to limit after the dither or as part of the dither as dither can change levels!), and this can result in gain changes. The DAC may behave differently or have outright bugs for some configurations too.
This is particularly true wrt reconstruction filters for sample rate differences. And if you were comparing 44.1k and 192k then the physical DAC itself was likely running at a different rate and its _analog_ filters are probably better optimized for one vs the other (this is less true for 48k vs 192k, as the hardware likely runs at the same rate for both). So one answer to this comparison can be "on this particular hardware this rate is better than that rate"-- but that's a implementation property not a property of format choice.
You might think, "okay I'll use a mathematically perfect down and up conversion process and run the DAC in the exact same configuration for all cases". But even then you run into issues like after reconstruction the _inter sample_ peak levels will be higher than the levels of the samples, so you have to handle that and in a way that doesn't produce a gain difference between the two configurations. (probably by running your perfect process and finding the gain level that results in no limiting, then making the gain of the original match).
And then for the high rate vs non-high rate you have to deal with the fact that most amplifiers are not particularly linear (compared to well constructed software at least!) and that any real speaker is very far from linear. This means that the presence or absence of ultrasonics will change the audio in the 0-20khz band.. Before you think "well that could be a reason that high rate is better" observe that if there was some consistently good effect from the ultrasonics you could just bake it into the low rate sample.
> but in my 50s I know
Yeah if you're in your 50's you're absolutely not hearing differences way up above 20khz (especially if you're male), I bet you can't even hear CRT flybacks from 100 yards anymore. :P Most people have no idea how much their high frequency hearing degrades as they age because it plays approximately no role in your life, but it's real, dramatic, and as far as I know happens to everyone.
I don't mean to discount your experience: I don't really doubt that it was real. But answering the general question of the necessity of low vs high rate probably takes a team of experts, armed with test gear and the designs of the HW/SW in question, to vet the test configuration. Testing a _particular_ configuration without the ability to distinguish its implementation quirks from format-fundamentals is much easier and that's what most attempts to test this question are actually testing.
By testing in a recording studio you were doing far better than most such comparisons. Usually people try comparing different files and they're comparing entirely different mastering processes. Files made for the "high res" market will often have much less compression and limiting then files made for commercial radio play / casual listening... and truly do sound obviously much better. Some of my favorite recordings are rips from vinyl. Vinyl is an awful format from the perspective of audio fidelity, but it's also pretty intolerant of excessive compression and limiting because the record will skip if the needle is bouncing off the rails. And more recently I suppose they also avoid over compression there because of the difference in target listener/environment.
Yes, perhaps the amplitude was subtly different.
This was supposed to be running the DACs to match the source configuration, not resampling into some common format. I think that is an unavoidable part of the whole end-to-end ABX test concept.
Maybe it would be interesting to up-sample back into 24/192 and play both in that mode. But then people would argue about what type of up-sample to use.
I was in my mid 20s for this test. I understand my high-band hearing was better back then.
Speaking about up-sampling, Im curious to know your opinion on the benefits of it. Im sending CD resolution audio as well as web streams from soundcloud.com to cambridge audio azur 840C and its not clear if its the up-sampling that makes the difference or their per channel wolfson dac arch. The iPod Video with their dac sounded amazing with just normal AAC files compared to the iPods before or after it.
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> Small differences in gain are ABX able much more readily than differences in noise at the 16 vs 24 bit level.
This was common knowledge at least as far back as the mid 80s, when every hifi shop and salesguy knew to ensure the bit of gear with the highest profit margin got played an almost imperceptible bit louder than the gear the customer came in to buy during back to back testing.
It's also a reason why double-blind testing is important. If someone doing the setup is expecting one piece of kit to sound better, if it doesn't they'll check the configuration more, and difference in gain can come from many sources. So errors that result in higher gain in favor of the "better" candidate go uncorrected, while ones that favor the worse tends to be fixed.
Point being: it doesn't even require an unscrupulous sales person to get similar results to an unscrupulous sales person! :P
Ah, 20kHz and CRT flybacks.. when I was a child I could of course hear that (in Europe that would be 15625 Hz), when I studied electronics and TV repair we could all hear that, and because we had the equipment we "tested" what we could hear using a function generator. The limit for conscious hearing for me was somewhere around 17kHz. Or not 18kHz for sure.
But I think I lost the ability to hear the flyback not long after I passed twenty. The world turned silent as far as that's concerned (before, you could hear it anywhere and everywhere, in shops, homes, some workplaces..)
The "20kHz" thing is kind of a myth for most people, at least that's what it looked to me after all the testing we did at school. I think it can influence what you hear, somehow, but in any case it's for very young people.
> Most people have no idea how much their high frequency hearing degrades as they age because it plays approximately no role in your life, but it's real, dramatic, and as far as I know happens to everyone.
I agree completely. I recall some discussions a long time ago on RMMGA (Usenet: rec.music.makers.guitar.acoustic) where some distinguished and experienced, but middle-aged guitarists got practically angry when a young guy described the sound of a certain type of newly-introduced strings "harsh" and "like fingernails on a blackboard" when used on a particular guitar.
The difference was, of course, that what the young guy could hear is something which stopped existing at least when you had passed 30.. I was at an age where I too couldn't hear that kind of sound from strings, but it was still not that long ago and I remembered and had noticed the difference, i.e. that I could not hear what I could hear before. For example the huge difference between fresh strings and week-old strings (and that fact has, over the decades, saved me tons of money which I would otherwise have spent on replacing strings all the time..)
Even for PC, I recommend some cheap studio monitors.
https://www.bhphotovideo.com/c/product/964752-REG/yamaha_hs8...
Those things have huge drivers and are probably too big for a lot of rooms. Unless you (and your neighbors) absolutely want to have that thumping sound and you go out of your way to kill unwanted bass reflection you're probably better off with the HS5 or something similar.
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Yeah I'm just lazy about dealing with the room. If I find the motivation, I'll pull the original equipment back out of storage.