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88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)
Old 20th February 2020
  #1291
The edge of perception

Quote:
Originally Posted by earlevel View Post
It its interesting indeed, but a little slippery. I wonder if it can be repeated. The differences were relatively small, not like cost tossing where there are two opposite outcomes. And not valid for ABX testing.
The whole subject is very slippery, and quite difficult to study. I remember an informal test that was offered at AES many years ago. (I can't recall whether it was AB or ABX, but it was blind.) My reaction during the first few examples was "oh, that's quite obvious!" I felt so sure of my results that I started trying to figure out how I was making the identification. As soon as I started doing that, I couldn't hear the difference anymore. Unsurprisingly, the results for these later trials were random. This present study makes me wonder whether "attentive" listening caused me to engage "conscious" neural pathways that were exactly the wrong ones.

Quote:
I believe you stated it correct in terms of error due to chance, but I know almost everyone will think that means it's a 95% chance the results are correct, and overwhelming powerful vindication, and that's not what it means ("the p value fallacy"). I'd like to see a follow up to this...
Indeed. Much of the rigor in scientific method comes from replication.

David
Old 20th February 2020
  #1292
Gear Maniac
 

Quote:
Originally Posted by monomer View Post
Hmm. Musically relevant is a relative term. They boosted the ultrasonics by a considerable amount.
What would the results be if they hadn't boosted these frequencies?
Also hmm, and they didn't test whether blasting arbitrary high frequencies elicited the same response...I can see it now, a new audiophile product that blankets your listen area in ultrasonics to let you enjoy your old CDs better—based on "science"

Hot dam, heat up the solder iron, I have a new product idea...
Old 20th February 2020
  #1293
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sax512's Avatar
 

Quote:
Originally Posted by earlevel View Post
Also hmm, and they didn't test whether blasting arbitrary high frequencies elicited the same response...I can see it now, a new audiophile product that blankets your listen area in ultrasonics to let you enjoy your old CDs better—based on "science"

Hot dam, heat up the solder iron, I have a new product idea...
And of course what you eat probably affects your mood more than that.
You get the ultrasonic 'relaxing device' done, I will take care of the audiophile sandwich, ok?
Old 20th February 2020
  #1294
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Quote:
Originally Posted by Yannick View Post
When I was looking for a natural sounding 44.1K DAC, even the Bricasti with its 15 filter settings did not have one setting which made me relax. All sounded more digital than eg. Benchmarkmedia DAC2. The linear fase steep filters were amongst the worst sounding !
This should give you some insight into the difficulty of doing 44.1k properly, it's a ridiculous standard that only makes sense when minimizing data rate is a primary concern (which it was for the CD).

Funny that you mention Benchmark, they get around the problem by upsampling internally to 110 kHz.
Old 20th February 2020
  #1295
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Quote:
Originally Posted by David Rick View Post
The whole subject is very slippery, and quite difficult to study. I remember an informal test that was offered at AES many years ago. (I can't recall whether it was AB or ABX, but it was blind.) My reaction during the first few examples was "oh, that's quite obvious!" I felt so sure of my results that I started trying to figure out how I was making the identification. As soon as I started doing that, I couldn't hear the difference anymore. Unsurprisingly, the results for these later trials were random. This present study makes me wonder whether "attentive" listening caused me to engage "conscious" neural pathways that were exactly the wrong ones.
This is interesting, but not necessarily surprising. Sometimes conscious attention can change an outcome. A parallel might be an athlete who doesn't think about mechanics during a game, if they do performance often declines.
Old 20th February 2020
  #1296
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Quote:
Originally Posted by johnnyc View Post
This should give you some insight into the difficulty of doing 44.1k properly, it's a ridiculous standard that only makes sense when minimizing data rate is a primary concern (which it was for the CD).

Funny that you mention Benchmark, they get around the problem by upsampling internally to 110 kHz.
Nope, it only gives insight in playing back 44.1 K properly.
The filter is a big issue.
The dac2 and dac3 actually upsample to 211 KHz !

In my listening tests the dac2 is far superior to many other dacs at 44.1 K.
It is so good the difference between 96 and 44.1 is just not there.

IMO it shows that the adc side is probably much less critical, as this is maybe far more simple, since everybody has been using oversampling the last two decades.

Probably it remains extremely impossible to design a synchronous, non oversampling dac at 44.1 K

But that still does not prove anything is wrong with capturing and storing at 44.1 ?

Interestingly, I am invited in another studio for a dxd 384 Khz surround demo. I think I will need to visit. Maybe I will change my mind completely !

Anyway, with my gear, in my mid-priced budget, there is no difference.
Old 20th February 2020
  #1297
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Quote:
Originally Posted by Yannick View Post
But that still does not prove anything is wrong with capturing and storing at 44.1 ?
Benchmark operates at a frequency that they determine to be most optimal for the chips they are using. 110k might be the older model, if it's now 211k that means the new chips work best at that rate.

You can't assume going up and down from 44.1k will have no impact vs doing everything at the best rate to begin with.

Whenever I've measured complete systems, they are always more accurate at higher sample rates.
Old 21st February 2020
  #1298
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Quote:
Originally Posted by johnnyc View Post
Whenever I've measured complete systems, they are always more accurate at higher sample rates.
What did you measure? And how?
Old 22nd February 2020
  #1299
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Quote:
Originally Posted by sax512 View Post
No, it isn't. It depends on the digital filter design.
Anyway, even with less than top notch digital filters, the amount of IMD in even the best audio chain is about 60 dB higher than aliasing, in the part of the audio band that we are most sensitive to.
All-right, but you seem to assume there is plenty of content above 22 kHz in the signals only because you record in 88.2. But this is not the case in most of my recordings. Already at the mics there is roll-off. Of course there are few issues, that I've encountered anyway, recording in 44.1 and mixing in 88.2/96. I can hear aliasing in some channels, which is why I've addressed that when needed and I did that by mixing in 88.2/96 (I'd say that resulted in 90+% less aliasing).

As many producers, also on this thread, are recording in higher sample rates you are effectively saying that all their work is very likely to contain IMD issues. So according to you these producers have no idea what they're doing.

Last edited by Mikael B; 22nd February 2020 at 12:00 PM..
Old 22nd February 2020
  #1300
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Quote:
Originally Posted by Mikael B View Post
All-right, but you seem to assume there is plenty of content above 22 kHz in the signals only because you record in 88.2. But this is not the case in most of my recordings. Already at the mics there is roll-off. Of course there are few issues, that I've encountered anyway, recording in 44.1 and mixing in 88.2/96. I can hear aliasing in some channels, which is why I've addressed that when needed and I did that by mixing in 88.2/96 (I'd say that resulted in 90+% less aliasing).
IMD is there even without considerable ultrasonic content. The majority of it comes from the audio band content. It is, at best, 40 dB lower than the actual content. With many speakers, I would be willing to bet it is actually somewhere in the -30 dB range.
Now, this might seem catastrophic, but it is so only in respect to the level of aliasing. Typical aliasing is -100 dB or less, so 60 dB lower than the IMD created by the audible content in the audible band.
It could be that, with less than well thought out digital filters, you get some aliasing higher than -40 dB, but it is limited to the upper range of the audible band, where people usually can't hear it, and requires high amount of ultrasonic content to be generated.
However, considering the capability of your ears AND the flaws of speaker design AND the fact that you listen to them in a room which reflects your first wave-front and gives it back to you sometimes at +10/15 dB (yes, that's a plus sign), even -30/-40 dB is usually undetectable.

Quote:
Originally Posted by Mikael B View Post
As many producers, also on this thread, are recording in higher sample rates you are effectively saying that all their work is very likely to contain IMD issues. So according to you these producers have no idea what they're doing.
Not necessarily (but not even necessarily that far fetched). There might be that plug in that they like that works better at higher sample rates. There might be concerns about latency. There might be practical reasons for the need to record at higher sample rates (marketing being a big one).
However, these reasons are not indication of something inherently wrong with 44.1 kHz.
When you consider sound reproduction, your best bet for accuracy is to feed your amp and speakers the least amount of frequency content necessary, as it MIGHT be that the extra IMD (the one on top of the unavoidable IMD already there from the 0-20 kHz range), could further audibly deteriorate the accuracy (it always does and it's measurable, but it doesn't mean that it's necessarily audible).

What is important to understand is that there is NO WAY, as of today, to get a signal accurately reproduced with less than -40 dB distortion on top of it.
That's the theoretical limit with today's best speaker drivers.
Real listening in real rooms is waaaaay worse than that.

So, all in all, I wouldn't say that I am that much AGAINST recording and listening at higher sample rates.
As I said, there are way bigger fish to fry than extra IMD (aliasing is NOT one of them).
If one is that much paranoid about plug in design, I would say that using higher sample rates is probably an acceptable evil.
One could always downsample at the very end of the production chain with a precise SRC program and generate their CD tracks that way.

What grinds my gears is that people keep saying that 44.1 is not as accurate as higher sampling rates, when there is measurable proof that, if filters are done right and there's no broken plug in in the chain and latency (we're talking 1ms, by the way) is not a concern, 44.1 is actually the only standard sample rate capable to squeeze the last bit of accuracy out of your audio chain.
Which is not saying much given the amount of crap your signal goes through for other reasons before hitting your eardrums.
But it's a matter of principles to state that 44.1 is potentially the most accurate way to record and reproduce sound, and if one thinks they can hear aliasing on top of all that crap I was saying, either they need a better designed DAC/ADC, or they're full of even more crap.
Old 23rd February 2020
  #1301
Gear Guru
Larger doesn’t mean better.....
No one I’ve talked to has ever said there is a problem with 44.1 being accurate and quite the opposite...
Old 23rd February 2020
  #1302
Apologies if this is a bit off topic - I set up a Native Instruments Audio DJ 2 soundcard (the old one) a few days ago and when it is set to anything other than 44.1 it makes a high frequency ringing on low bass.

I don't really understand what it could be but it sounds like bit distortion (obviously it isn't). It can be clearly heard playing low sine tones from Audacity.

I also noticed the same thing on the built in soundcard on my old HP laptop.

I'm aware these are not high-end, but in the case of NI especially it is bizarre you can set it to "too high". Even I can't get too high ffs.
Old 23rd February 2020
  #1303
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Quote:
Originally Posted by ardis View Post
Larger doesn’t mean better.....
No one I’ve talked to has ever said there is a problem with 44.1 being accurate and quite the opposite...
Meet johnnyc (one of the many)

Quote:
Originally Posted by sax512 View Post
Quote:
Originally Posted by johnnyc View Post
Whenever I've measured complete systems, they are always more accurate at higher sample rates.
What did you measure? And how?


Old 23rd February 2020
  #1304
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Quote:
Originally Posted by sax512 View Post
IMD is there even without considerable ultrasonic content. The majority of it comes from the audio band content. It is, at best, 40 dB lower than the actual content. With many speakers, I would be willing to bet it is actually somewhere in the -30 dB range.
Now, this might seem catastrophic, but it is so only in respect to the level of aliasing. Typical aliasing is -100 dB or less, so 60 dB lower than the IMD created by the audible content in the audible band.
It could be that, with less than well thought out digital filters, you get some aliasing higher than -40 dB, but it is limited to the upper range of the audible band, where people usually can't hear it, and requires high amount of ultrasonic content to be generated.
However, considering the capability of your ears AND the flaws of speaker design AND the fact that you listen to them in a room which reflects your first wave-front and gives it back to you sometimes at +10/15 dB (yes, that's a plus sign), even -30/-40 dB is usually undetectable.



Not necessarily (but not even necessarily that far fetched). There might be that plug in that they like that works better at higher sample rates. There might be concerns about latency. There might be practical reasons for the need to record at higher sample rates (marketing being a big one).
However, these reasons are not indication of something inherently wrong with 44.1 kHz.
When you consider sound reproduction, your best bet for accuracy is to feed your amp and speakers the least amount of frequency content necessary, as it MIGHT be that the extra IMD (the one on top of the unavoidable IMD already there from the 0-20 kHz range), could further audibly deteriorate the accuracy (it always does and it's measurable, but it doesn't mean that it's necessarily audible).

What is important to understand is that there is NO WAY, as of today, to get a signal accurately reproduced with less than -40 dB distortion on top of it.
That's the theoretical limit with today's best speaker drivers.
Real listening in real rooms is waaaaay worse than that.

So, all in all, I wouldn't say that I am that much AGAINST recording and listening at higher sample rates.
As I said, there are way bigger fish to fry than extra IMD (aliasing is NOT one of them).
If one is that much paranoid about plug in design, I would say that using higher sample rates is probably an acceptable evil.
One could always downsample at the very end of the production chain with a precise SRC program and generate their CD tracks that way.

What grinds my gears is that people keep saying that 44.1 is not as accurate as higher sampling rates, when there is measurable proof that, if filters are done right and there's no broken plug in in the chain and latency (we're talking 1ms, by the way) is not a concern, 44.1 is actually the only standard sample rate capable to squeeze the last bit of accuracy out of your audio chain.
Which is not saying much given the amount of crap your signal goes through for other reasons before hitting your eardrums.
But it's a matter of principles to state that 44.1 is potentially the most accurate way to record and reproduce sound, and if one thinks they can hear aliasing on top of all that crap I was saying, either they need a better designed DAC/ADC, or they're full of even more crap.
As this thread is now repeating posts/ideas, I'll repeat this.....yeah, at tracking, I believe plugs operate better at 96 or maybe 192.

So...I get my mix happening on the main daw (at say 96) and once a mix is sort of happening.....I do the old fashion thing....with daw #2 synchronized, I play the mix out of the main daw....via analog quarter-inch...into daw 2 (the two-track mixdown machine)...and it's quarter-inch connections...complete with the 2 trips of d/a then a/d (on Laylas)......with daw #2 set for whatever I want. Maybe a pass in at 24/48....32float/96....32/192....16/44.1.

Realtime. No dither. Personally, I think this way sounds great...via Laylas. The discussion could become, gee...so much a/d, d/a conversion.

I like this approach...perhaps because the workflow is what I'm historically used to. Plus...I detect a result that I like.

Of course, this has nothing to do with mastering, which, in this thread, i don't give a f*** about.

There....a repeat of a repeat of a repeat.
Old 24th February 2020
  #1305
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Quote:
Originally Posted by Mikael B View Post
All-right, but you seem to assume there is plenty of content above 22 kHz in the signals only because you record in 88.2. But this is not the case in most of my recordings. Already at the mics there is roll-off.
As an example, here is a string chamber orchestra recorded at 96kHz, from two DPA 4006TL's, The horizontal cursor is 20kHz.

But just because there is energy up at 37kHz doesn't mean we can ever hear it and therefore that we need to encode it.
Attached Thumbnails
88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)-aco_hf.jpg  
Old 24th February 2020
  #1306
Gear Addict
 

I agree with what you say, but this thread won't end them all. I know how these discussions go. Hahaha


Quote:
Originally Posted by Flippy Floppy View Post
This is a post that I just made it a recent thread. One of a plethora of high definition audio threads out there. Hopefully this thread can help end this silly debate over sample rates. And hopefully dissolve a few misconceptions about recording above 44.1kHz.

If you can hear audio far above 20kHz, high sample rates are for you. Unfortunately, very few humans can hear much above 16kHz. And yes, I know the argument about how high def will make the frequencies we do hear sound better. This is completely untrue. 1000Hz at any sample rate should sound the same. In fact 20-20,000Hz should sound the same at any sample rate. If the sound changes at different sample rates, you probably don't have very good converters.

I think this misconception of high def sample rates is due to people using photography as an analogy. Meaning the higher the resolution, the more detailed photograph. Digital audio dosen't work like that. Think of audio as light instead. Light has a spectrum that we can see. And like audio, there is a spectrum above (infrared) and below (ultraviolet) that we can't see. Much like how most people can't hear past 16-18kHz or under 20Hz. A quality converter samples the audible frequency as close as possible to the source, regardless of the rate. If 1000Hz is a blue ball and a camera takes a picture of it, a quality camera will produce a picture of a blue ball like the source. Now if there was a setting on the camera that could capture light all the way up into the infrared spectrum (96kHz) and you took a picture of the blue ball again, it should be exactly the same color—even though there are additional infrared frequencies we can't see in the picture.

Another misconception is analog tape speeds compared to sample rate speeds. With tape, it's achieving the same noise floor of about 12 or 13 bit digital audio. Speeding up tape to 30ips will lower the noise floor to achieve a better S/N ratio... Maybe 14 bit. Also the frequency roll off shifts up. Meaning, at say 15ips, the lowend roll off will start at 30-40hz. At 30ips it will start rolling off at 50-60hz (depending on the machine and calibration). With tape speeds, it's always a trade off between: 15ips—bigger low end, softer highs and noisey or, 30ips— less sub low end, crisper highs and less noise. From 44.1khz to 96khz, we don't get this frequency shift like tape. With digital, the frequency responce is flat from end to end of any given sample rate. You just are just recording inaudible frequencies. Analog tape is a very crude way to record an audio signal. It's noisey, it's frequency responce is all over the place and harmonic distortion evrywhere... But it is super vibey and the artifacts are pleasing!

Here's the kicker... Guitar amps start rolling off at 5khz, most all dynamic microphones start rolling off at 16khz at best, most condenser mics start rolling off at 18-20khz, same with most preamps. What are you guys recording up there?!?!

I've never heard a hit song that was ruined by a bad choice of sample rate!!
Old 24th February 2020
  #1307
Here for the gear
 

I have to disagree, there are many things involving direction and perception as well as harmonics that are carried in the real world out of the range of what is perceived as human hearing. Digital audio at best is an approximation of real audio, if your sampler is "looking" at a waveform at say 44 thousand times per second and this waveform is modulating at 20 thousand times per second it's guessing at what happens between the samples. It's bad enough with a simple sine wave, imagine the detail that is lost with a complex musical waveform.

It's a pity that we are now generations removed from analog audio and the acceptance of mediocrity is becoming more widespread. This follows with the plug-in myth. So many are convinced by nice graphics ad visual simulations of "vintage" processing gear, when the bottom line is that any kind of digital desk or processor is nothing more than an over priced calculator, all they can do is add and subtract numbers... period. These simulations are just achieved by algorithms that degrade the performance of the A/D conversion. I would give them more credibility if they just had screens that said what they were doing.

I've wandered off track ;o) but it's just a statement on the trend to lower the bar. We should strive for the best that is practically possible, memory is cheap nowadays and if it's a matter of time... if it's not worth the time to do it right, it's probably not worth doing.
Old 24th February 2020
  #1308
Quote:
Originally Posted by David Spearritt View Post
As an example, here is a string chamber orchestra recorded at 96kHz, from two DPA 4006TL's, The horizontal cursor is 20kHz.

But just because there is energy up at 37kHz doesn't mean we can ever hear it and therefore that we need to encode it.

This looks very much like aliasing, and likely not part of the original audio content.

Amplitude of harmonics falls with frequency, this HF section looks as if it does the opposite, it probably mirrored at Nyquist.

IMHO, this is good design, and to be expected. This ultrasonic range is beneficial at reducing the necessity for aggressive filtering near the audible band, but in audio, it's really rarely meant to capture the whole range. What counts, to just any field of audio engineering, is the integrity of audible content.

All our products tolerate aliasing outside the audible band (never inside it, though), to make the process behave better over the audible band.

Last edited by FabienTDR; 24th February 2020 at 04:54 PM..
Old 24th February 2020
  #1309
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sax512's Avatar
 

Quote:
Originally Posted by ecaltd View Post
I have to disagree, there are many things involving direction and perception as well as harmonics that are carried in the real world out of the range of what is perceived as human hearing. Digital audio at best is an approximation of real audio, if your sampler is "looking" at a waveform at say 44 thousand times per second and this waveform is modulating at 20 thousand times per second it's guessing at what happens between the samples. It's bad enough with a simple sine wave, imagine the detail that is lost with a complex musical waveform.

It's a pity that we are now generations removed from analog audio and the acceptance of mediocrity is becoming more widespread. This follows with the plug-in myth. So many are convinced by nice graphics ad visual simulations of "vintage" processing gear, when the bottom line is that any kind of digital desk or processor is nothing more than an over priced calculator, all they can do is add and subtract numbers... period. These simulations are just achieved by algorithms that degrade the performance of the A/D conversion. I would give them more credibility if they just had screens that said what they were doing.

I've wandered off track ;o) but it's just a statement on the trend to lower the bar. We should strive for the best that is practically possible, memory is cheap nowadays and if it's a matter of time... if it's not worth the time to do it right, it's probably not worth doing.
Here's another one, ladies and gentlemen

Old 24th February 2020
  #1310
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Plush's Avatar
44.1 apologists can bite me. Such garbage is written here by rank amateurs and s-talkers.

How to test it??

Certainly not by doing an A/B test switching back and forth between 44.1 and higher sampling rates.

Instead, mark out a week to only work in hi-res at 96k or 192k--both in recording and playback / editing. Work for one week exclusively in the higher sampling rate at your studio.

Then switch back to using 44.1k or 48k and feel the collapse of the stereo picture. Feel the unease creep back into your playback. Hear the garbage filtering of 44.1k.

An easy test and one that is convincing.
Old 24th February 2020
  #1311
Quote:
Originally Posted by Plush View Post
44.1 apologists can bite me. Such garbage is written here by rank amateurs and s-talkers.

How to test it??

Certainly not by doing an A/B test switching back and forth between 44.1 and higher sampling rates.

Instead, mark out a week to only work in hi-res at 96k or 192k--both in recording and playback / editing. Work for one week exclusively in the higher sampling rate at your studio.

Then switch back to using 44.1k or 48k and feel the collapse of the stereo picture. Feel the unease creep back into your playback. Hear the garbage filtering of 44.1k.

An easy test and one that is convincing.
this has been my experience,
working in 44.1 for a long time, and changing to 88.2 when i upgraded my DAW/PC setup.
Old 24th February 2020
  #1312
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sax512's Avatar
 

Quote:
Originally Posted by Plush View Post
44.1 apologists can bite me. Such garbage is written here by rank amateurs and s-talkers.

How to test it??

Certainly not by doing an A/B test switching back and forth between 44.1 and higher sampling rates.

Instead, mark out a week to only work in hi-res at 96k or 192k--both in recording and playback / editing. Work for one week exclusively in the higher sampling rate at your studio.

Then switch back to using 44.1k or 48k and feel the collapse of the stereo picture. Feel the unease creep back into your playback. Hear the garbage filtering of 44.1k.

An easy test and one that is convincing.
How I missed your nonsense, Plush.
Old 24th February 2020
  #1313
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Quote:
Originally Posted by batsbrew View Post
this has been my experience,
working in 44.1 for a long time, and changing to 88.2 when i upgraded my DAW/PC setup.
Sorry, just poor anecdote with no repeatable, peer reviewable, scientifically acceptable testing.

Your brain and sensory organs are designed to take short cuts to save power, given the high cost of running the brain. Interpolation is very common in hearing and vision.

So unless you have extraordinary proof, it's just, like, your opinion man.

Smoke some weed, it sounds even better
Old 24th February 2020
  #1314
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Quote:
Originally Posted by Plush View Post
44.1 apologists can bite me. Such garbage is written here by rank amateurs and s-talkers.

How to test it??

Certainly not by doing an A/B test switching back and forth between 44.1 and higher sampling rates.

Instead, mark out a week to only work in hi-res at 96k or 192k--both in recording and playback / editing. Work for one week exclusively in the higher sampling rate at your studio.

Then switch back to using 44.1k or 48k and feel the collapse of the stereo picture. Feel the unease creep back into your playback. Hear the garbage filtering of 44.1k.

An easy test and one that is convincing.
I wouldn't normally duplicate a response, but this needs it:

Sorry, just poor anecdote with no repeatable, peer reviewable, scientifically acceptable testing. And nonsensical claims given the proven math.

Your brain and sensory organs are designed to take short cuts to save power, given the high cost of running the brain. Interpolation is very common in hearing and vision.

So unless you have extraordinary proof, it's just, like, your opinion man.

Smoke some weed, it sounds even better

Last edited by diggo; 24th February 2020 at 05:59 PM.. Reason: spellcheck error
Old 24th February 2020
  #1315
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Quote:
Originally Posted by Plush View Post
44.1 apologists can bite me. Such garbage is written here by rank amateurs and s-talkers.

How to test it??

Certainly not by doing an A/B test switching back and forth between 44.1 and higher sampling rates.

Instead, mark out a week to only work in hi-res at 96k or 192k--both in recording and playback / editing. Work for one week exclusively in the higher sampling rate at your studio.

Then switch back to using 44.1k or 48k and feel the collapse of the stereo picture. Feel the unease creep back into your playback. Hear the garbage filtering of 44.1k.

An easy test and one that is convincing.
With what exact brand/model interface?
Old 24th February 2020
  #1316
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sax512's Avatar
 

Thumbs up

Quote:
Originally Posted by thenoodle View Post
With what exact brand/model interface?
Why, this one of course!

Crane Song Solaris Quantum D/A Converter

From his own 'review':

2. tonally more accurate than other dacs--all frequencies very defined. Change from Benchmark was revolutionary. Bettered the DAD we have here. Bettered the PrismSound we have here.



How can you counter to this level of analytical demonstration?
I'm sold. 384 kHz, here I come!

Old 24th February 2020
  #1317
Quote:
Originally Posted by diggo View Post
Sorry, just poor anecdote with no repeatable, peer reviewable, scientifically acceptable testing.
dont' be sorry.
i'm not.

i do not care what you think.

i like doing what i like.

things sound better to me, at 88.2
they just do

again,
my opinion counts,
to anyone who listens to my stuff and likes it,
and that's what matters.
Old 24th February 2020
  #1318
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Quote:
Originally Posted by FabienTDR View Post
This looks very much like aliasing, and likely not part of the original audio content.
No, it's certainly original musical content. This is a 96kHz recording, and this energy is all well below Nyquist, at least 10kHz below.
Old 24th February 2020
  #1319
Here for the gear
everyone is different

I like to play vst 's at 88/97khz (roland sua)as i feel there is more clarity in my guitar sound(kazrog) plus less latency but not very practical as the required resources on the com is too high (need to upgrade even at 4770k 4.3 ghz 16 g ram ssd etc ) , but this clarity does not make a huge difference in a final mix. I think a lot of guitar players are after better latency which i think is a much more, and underestimated problem than we think. Its a subliminal thing I think......?
We need more companies coding their own drivers for their interfaces and coming up with ideas which will give us much lower latency devices sub millisecond. Rme seems to be the only one that has made any inroads with regards to latency albeit at the cost (minor) of sound quality.

some people want to go higher but when i do this i a cant hear any difference above 96khz and how would I as the equipment needed to do this would astronomical . high end Genelecs etc etc
everything we listen to in the main stream is 44khz
Old 24th February 2020
  #1320
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Plush's Avatar
Quote:
Originally Posted by thenoodle View Post
With what exact brand/model interface?
Can use DAD, or Prism Atlas, or Sonosax a/d's. d/a by Cranesong Solaris.

The demonstration works with any high quality adc / dac combination.
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