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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.

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.

8" is not a huge driver. The point with these is to avoid the need for a separate subwoofer for most program material.

It's the biggest model in the Yamaha monitor series and if you care about bass that much you probably also want a subwoofer ;)

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.

> 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.

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.

But surely the difference is 16 vs 24 and nothing to do with the sample rate?

The larger problem with 44.1 is

But it depends what you're sourcing from. If you source 44.1 then you will have a worse recording if you change it to 192. If you source at 48k then you just waste samples. If you’re recording analog inputs at 192k in a crappy adc then you will have a worse outcome than a good adc at 48k (or 44.1k)

Same with bit depth - the adc is far more important.

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.

    $ sox source.flac -b 16 dithered.flac
    $ sox --combine merge source.flac dithered.flac loud-difference.flac remix 1,3i 2,4i norm -1 spectrogram  # assumes stereo input
    $ open spectrogram.png
    $ open loud-difference.flac

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.

    $ sox source.flac -b 16 quiet.flac gain -100
    $ sox quiet.flac loud-again.flac norm -1
    $ open loud-again.flac

> 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 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.

> 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

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.

Any high end audio dac is internally running at a much higher sample rate-- that's what it takes to get their delivered performance with the silicon that's available. Its up-sampling process was designed by the designers of the DAC with intimate knoweldge of its analog properties.

Second guessing it by upsampling in front of it seems dubious to me. It might help in some cases where the DAC designers were thinking of different objectives or just didn't do a great job. It might also help with some other issues, like if the dac is timed off the input clock and the input clock sucks and the upsampler retimes the signal.

Of course the upsampler designers could also get it wrong, be aliasing the hell out of the results, and happen to like the sound of the corrupted audio. :P

The effects are all objectively measurable however-- with expensive equipment at least. I think I'd want to set test results with a particular hardware combination before sticking an upsampler in it. OTOH, if there already was one there because it's just some built in feature of some kit I wanted to use otherwise, I wouldn't worry much about it. Particularly if that kit has been reviewed by people with proper test gear and they didn't decide that it was broken.

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..)

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?

You're missing a few words. To be correct you would have to have said 'if nobody's capable of hearing the difference every single time beyond a statistical doubt, does it matter?'

You can say that and be correct, while also sounding a little more silly than perhaps you'd like.

edit: rather than go even harder, I'm instead going to suggest it's perfectly fine to care about things you don't hear every single time, but still like or dislike :)

My pet example is sand in the lettuce for a salad. If you dislike that particular cronch against your teeth while eating salad, it has a spectacular ability to ruin your enjoyment of your salad, even though you don't perceive it every single time. Digital distortions are like that for some of us, things like wow and flutter and vinyl surface noise are like that for others. People vary. (which is also why not to generalize about what 'people can hear')

Oh, I care about audio quality a lot more than most people, but not into 'magic cable' territory.

There's no reason you can't gather statistics about a representative part of the population. It doesn't make sense to make the entire world pay for better audio because some guy somewhere might be a bit more sensitive and he thinks it ruins everything.

I don't know if such a person exists, but especially in the 'magic' audio territory there's a big amount of bullsh*t going on. There's a reason the James Randi price was never claimed.

There's a huge difference between digital 'distortions' caused by sampling at CD quality and things like tape flutter which most people can actually hear. Even then, some people like the imperfections like vinyl or tape artifacts. Some people even prefer MP3 compressed music.

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.

https://news.ycombinator.com/item?id=48774112 <-- Here is a test. I would have said the same and I played all samples quite often. It is extremely subtle, but even with a cheap headset it is possible to hear differences, where the high quality version is just a little bit clearer.

> 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.

> With a 24/192 Master in FLAC or ALAC, I can downsample to whatever the destination form factor is. I can transcode to a 320kbps MP3, or a 16/48 WAV stream for a smart speaker, or a 24/96 stream for the theater.

I used to think the same. But I realized that downsampling hi-res music to 16/44.1 isn't a transparent conversion. So now I prefer the one downsampled to 16/44.1 by an expert in production env. I almost always download 16/44.1 flac files because of this.

I can't hear the difference between 128 kbps opus and FLAC.

And that's fine! I've got a flatmate who loves 320kpbs MP3s on studio monitors, I've got musician friends who swear by CD-audio and Sennheiser HD200s, and others who love how vinyl uniquely degrades over time on big speakers.

The takeaway from these sorts of posts, at least in my opinion, should be two-fold:

* Understand the physical limits of human senses and perceptions to help inoculate yourself against outright scams and grifts

* Liberate you from the "tech grind" and allow you to enjoy what you like, how you like it.

> Understand the physical limits of human senses and perceptions to help inoculate yourself against outright scams and grifts

Also understand that while there is an upper limit, we are all different within that. I can hear the difference between 128Kbps and FLAC, at least for some content, but not 256Kbps, maybe not 192. For some content (spoken word etc.), 64Kbps, sometimes less, is perfectly acceptable (to me). There was a time I could hear the difference between some encoders, but that was decades ago and anything in active use is pretty damn good (and my ears are not what they used to be) unless you really crank the bitrate down or tweak other options daftly.

> I can hear the difference between 128Kbps and FLAC

You've established this with double bind testing, correct?

Some time ago, yes. For 128 and there-about & below, not 192+ hence I'm less certain about that (but I'm pretty sure I wouldn't be able to tell the difference).

Not recently, so it is possible that improvements in encoding methods, and changes in my ears, could mean that I'd get a different result now.

I’ve not tried encoding my own MP3s in at least a decade, but when I was doing so, 128 kbps was instantly distinguishable to me on anything with cymbals, especially hi-hat: it loses that shimmery sound. At 192 kbps I could tell if I really, really tried, but it was so minute I didn’t really care. I was never able to reliably tell the difference between 256 and 320 kbps rips.

The thing I didn't understand with higher quality music files is that it's not like the entire song is different and better when you go from 64 to 128 kbps opus, it's just these super minor details that get changed. It was enlightening doing an abx test, but I still use flacs because it's nice not worrying about the quality mattering.

> I can't hear the difference between 128 kbps opus and FLAC.

A reasonable definition of transparency for high bitrate compressed audio is "Can the worst files be distinguished by a listener trained in what artifacts sound like". Maybe also add in having to use a high discrimination listening setup, including not running excessively loud (increases masking).

If that's not the test you're doing, it's unsurprising. At moderately high bitrates no one can reliably distinguish them on arbitrary samples: most inputs are easy.

If you test on known-difficult "killer samples" you'll probably easily distinguish them, even without first being shown what to look for, and certainly after.

During the development of Opus I created many 'trained listeners' and selected many killer samples, and I don't recall* ever encountering a tin ear that couldn't be taught to ABX any high rate samples, though some people are obviously much better at it.

I'm not sure I'd recommend it though: learning to identify artifacts has a frequent side effect of making low rate audio like the HE-aac used in SirusXM absolutely intolerable. I'm bothered by it even when I hear cars driving by using it. :)

[*] My memory for such things sucks, so I could be wrong-- but my point that it's not expected remains.

I did the ABX test extension in foobar2000 with Octopus's Garden. It was on nice headphones.

You're right it's just minor details.

That's not really how down sampling works. You already have all the information with 48 kHz sampling.

Higher rate sampling is just like storing integers to 3 decimal places, or archiving an upscaled DVD.

I recommend you actually read the article. I vaguely recall they did it in video form too.