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88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)
Old 1st March 2020
  #1441
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Also, there is NOTHING realistic in the way a standard recording is usually done.

It is unnatural to close mic instruments, it is unnatural to use mics that have no resemblance to how our hearing works, it is unnatural to EQ (and compress and limit) the raw audio and it is unnatural to listen to the resulting signal through speakers that most of the time have an impulse response that looks like crap (how's that for time smearing, by the way?).
Finally, it is unnatural to have that sound wave floating around a room that has no resemblance to the room in which the recording was done.

... Yet, somehow, all the make or break about a digital recording relies on that extra bit of inaudible ultrasonics for it to sound like the real thing.

Come on. This is bad logical thinking.
Old 1st March 2020
  #1442
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Quote:
Originally Posted by bogosort View Post
I just watched that video, they did a great job explaining the issues and convincingly demonstrating the advantage of using 44.1/48 project rates with oversampling plugins over 96/192 project rates.
Every video on YouTube made by Dan Worrall is worth watching. All the TDR plugins, all the Fab Filter plugins and more.
Old 1st March 2020
  #1443
"objectively broken"

Quote:
Originally Posted by sax512 View Post
Where is the proof that most ADCs/DACs and plugins are broken?
I'm not going to speak to the topic of plugins. There are thousands, some good, some bad. Most of them are "black boxes" and the only way to determine what they are doing is through painstaking "bench testing".

The proof that the average converter chip shouldn't be used at 44.1k is printed in black and white in the data sheet. Earlier in this thread I displayed the filter characteristic for a well-respected Cirrus Logic converter, but I don't wish to pick on one particular manufacturer so today we'll look at a similar part from Texas Instruments.

The PCM4220 is a 24 bit stereo ADC that uses multi-bit delta-sigma architecture to achieve an advertised SNR of 123 dB. The part prefix tells me that it was designed by some of the top-notch engineers that TI brought onboard when they acquired the Burr-Brown company. Being a 2-channel part with a selling price around $10, it's a bit too pricey for your run-of-the-mill multi-channel audio interface, but could be used in gear with lower channel counts or higher price points. Here's a graph showing the filter performance around fs/2:



The normalized frequency axis tells us that this chip uses the same filter coefficients at all sampling rates. One can see that the response at 0.5*fs is only 6 dB down. This converter has both "standard" and "low delay" filtering modes. Here's a graph of the stopband response in low delay mode.



Although stopband attenuation ultimately reaches -90 dB (-100 dB for the standard filter), it doesn't reach that figure until 25.7 kHz. Frequencies below that point can alias at much higher levels. Indeed, it's possible to have alias components appear below 20 kHz that are only 20 dB down. Don't panic though: If you run the part at 48k, the alias components are a full 90 dB down at 20 kHz. Clearly the chip designers expected that anyone who is "serious" about audio quality will run the chip at 48k or higher.

We've spent some time discussing oversampling converters in this thread, so it's interesting to look at the extended filter response.



Those of us who've done multi-rate DSP design can see a lot in this graph. It's clear that the decimation is done in multiple stages of filtering and down-sampling, with the initial stages being short recursive filters of the "cascaded integrator comb" type and the final one or two stages being much longer FIR filters run at lower sample rates.

David L. Rick
Attached Thumbnails
88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)-pcm4220-classic-transition-band.png   88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)-pcm4220-fast-stop-band.png   88.2 - 96kHz Vs 44.1-48kHz (a thread to end them all!!)-pcm4220-fast-extended-stop-band.png  

Last edited by David Rick; 1st March 2020 at 08:22 PM.. Reason: "prefix", not "suffix"
Old 1st March 2020
  #1444
Gear Nut
Hypothetical sample rate related question:
Say you record a project at 96K and artists want to add their own tracks recorded with their own interfaces. Most consumer interfaces do not support 96K.
How do you proceed?
Old 1st March 2020
  #1445
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Quote:
Originally Posted by David Rick View Post
Plush's opinion of 44.1k recording is the result of careful listening, a thing that is taught very well in European tonmeister programs. I believe his colorful way of expressing it was learned on the streets of Chicago, however.

One needn't know much (or any) math to know that 44.1k sampling is inadequate for some kinds of program material. All you need to do is listen. But if you do know some math, then it's possible to understand exactly why, and I explained it rigorously (months ago!) in this very thread. It has nothing to do with Nyquist, and everything to do with engineering. Bob O. summarized it best: "It's all about the filters."

Marco's standard reply to this point is essentially "We can design digital filters as good as we need to by simply making them longer." In the interest of avoiding another tedious argument about time response I'll say simply: The problem is that most converter designers don't make those "good" filters. The majority of ADC chips today utilize half-band filters that allow significant aliasing around the Nyquist frequency. When those chips are used at 44.1k, some of that aliasing happens below 20k. Maybe you've bought into the idea that 2x and 4x sample rates are audibly useless (or worse). Fine; then record at 48k. But don't record at 44.1k because the overwhelming majority of 44.1k converters on the market are objectively broken.

The case against 2x and 4x sample rates rests on two major arguments. One is that that letting anything over 20 kHz reach your speakers will cause horrible levels of IMD distortion. Leave aside that most IMD is caused by signals below 20 kHz. If significant ultrasonic-induced IMD is present then it is also present in analog recording chains and is by now certainly "baked into" our collective audio consciousness in much same way that we like the way an analog compressor or a U47 distorts. It's not some new artifact that converter companies are imposing on us in a conspiracy to ruin our audio.

The second argument involves digital domain nonlinearities -- mostly digital compression and limiting. The recent discussion of this -- especially the fabfilter demos -- were very enlightening and have helped me to understand why I never much liked the sound of digital compression. In modern mix practice, which typically involves multiple nonlinear devices in series, 2x and 4x rates don't fully prevent aliasing of the resulting distortion components -- it takes oversampling and judicious filtering at key points in the signal chain to prevent the build-up of digital grunge. Or you can simply do what I do and perform all your compression in the analog domain.

Which still begs the question: Why would one want to record at higher sample rates, especially after sax512 has spent the last several months browbeating everyone with Fourier theory and the horrors of intermodulation. If reproducing upper partials simply causes our speakers to distort, why reproduce them? Here's an idea: Because our ears also distort. They've been doing it for thousands of years and we probably have evolutionary adaptations to the fact that help us to recognize when a sound is "real" and when it isn't. Audio reproduction at 1x sample rates doesn't sound "real" to me, nor to Plush either. I can't hear a darn thing above 16 kHz anymore, but apparently I don't need to: When I down-sample critical material from 96k to 48k I can still tell the difference and I don't like it.

David L. Rick
Seventh String Recording
What you and Plush keep forgetting is that I or we ALSO have done extensive listening, and comparisons, even blind.

I spent a lot of time with ADC and dac converters, and you are right, a lot of highend brands sound like complete crap at 44.1 K.

With what I use, I cannot hear a difference between 44.1 and 96.
Period.

As a last note, your last sentence about downsampling 96 to 48 completely undermines the rest of your post.

Maybe we can reach a consensus: whenever you use (albeit very expensive highend) mediocre ADDA conversion, it is better to stay at or above 88.2K

No way you can sell me the notion that a 8K dac should sound significantly better at 96K than at 44.1K. It proves there is a design flaw.
Old 1st March 2020
  #1446
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Quote:
Originally Posted by David Rick View Post
The proof that the average converter chip shouldn't be used at 44.1k is printed in black and white in the data sheet.
Buy better converters. Simple.
There are people (the ones we should really be grateful to, instead of the self proclaimed golden ears) that spend a lot of time measuring DACs responses. I wish there were more.

Of course, the chip response itself tells us only part of the story. The real story is told when the whole DAC response is measured.
However, with half band filters, it is impossible to not get some aliasing (although, it is only in the very upper frequencies where nobody can hear it, as opposed to IMD which is spread throughout the whole spectrum. You seem to forget to compare the power and distribution of these two separate sources of noise to fit your narrative, in my opinion).
But in any case... -20 dB distortion at 18 kHz bothers you more than a decrease in S/N throughout the whole spectrum (with noise not evenly distributed, by the way)?
Buy a better converter. Simple.
Old 1st March 2020
  #1447
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Here's one of the first google search hits I got.

One can do their own homework for other DACs/ADCs, but for this one, ballpark figures at hand, there's a rejection of more than 100 dB (120, actually) at 25 kHz, for 44.1 kHz sample rate, and the highest harmonic is sticking out barely above -90 dB.
Which means that aliasing doesn't affect anything below 19.1 kHz to a remotely audible degree. And likely stays well below IMD right up to 20 kHz.

This is a $300 DAC, which I hardly consider as a lot of money. Most of you fancy studio guys probably paid a lot more for that lava lamp on the corner.

Let's keep things in perspective, please.

https://www.stereophile.com/content/...r-measurements


EDIT: Please don't take the link above as an endorsement for snake oil selling rags the likes of Stereophile. Sometimes, though, they make measurements. Which almost makes up for the rest of the BS they spread.
Old 1st March 2020
  #1448
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Quote:
Originally Posted by Yannick View Post
With what I use, I cannot hear a difference between 44.1 and 96.
Period.
What do you use?
Old 2nd March 2020
  #1449
On the audio philosophy of Blaise Pascal

Quote:
Originally Posted by sax512 View Post
The fact that our ears distort audio themselves is basically saying that the eardrum IMD is what gives higher sample rates the 'obvious edge' you claim they have.

Seems a little far fetched, but I did ask you if you had any study that points at a shred of indication that this is the case.
To which you never replied, because there is no such study.
I happen to think the difference between 44.1k and 48k is obvious on certain program material (as do many others) and I've now posted more than one technical analysis giving bench-measurable reasons for that. I personally find the difference between 48k and 96k considerably more subtle (though worthwhile for my kind of work), but you're correct: I don't have an equally rigorous argument for why there's any audible difference there. All I have are some tentative hypotheses, which may or may not prove true under careful experimental study over the course of many years. We're not going to know the answer anytime soon.

So what should we do in the meantime? Probably whatever carries the least risk of harm to musical posterity. As Bob Olhsson said in this very thread, "You never know when you're recording history."

Let's examine the alternatives in the spirit of Pascal's wager. You argue that the key problem is that frequencies above 20k produce such serious IMD on playback though loudspeakers that they should be proactively filtered out in the recording chain. I counter that numerous golden ears think there's something above there that's worth preserving. Our choice is simply whether to record that extra octave or not. Well, if we do record it but you turn out to be correct on the IMD issue, then we will always have the option of removing that top octave some time in the future. With more and more speakers becoming digital, they could even have the required LPF built in. On the other hand, if we don't record that extra octave and one day learn that there is some perceptual utility to the information we stripped out then nothing we can do later will bring it back -- it's gone forever. I'll freely admit that, as an archival recordist, I have a problem with that.

In summary, 2x recording may not be easily justifiable based on today's understanding human perception, but there's little harm in doing it and we keep our options open for the future.

David L. Rick
Old 2nd March 2020
  #1450
The cheapest converter upgrade

Quote:
Originally Posted by sax512 View Post
Buy better converters. Simple.
C'mon Marco! Are you seriously suggesting that people rush out and replace their converter sets solely to get the same performance improvement that is achievable just by switching their existing boxes from 44.1k to 48k? Where's the logic in that? Running 48k gives no significant difference in CPU load or file size, but it makes many widely-used converter chips work better. Why not just do it?

The ADC chips I cited in my examples are used in a lot of pretty expensive boxes. There's a long-running thread on Gearslutz that enumerates which interfaces use which chips.

The CS5381 ADC I discussed in an earlier post is used in dozens of well-regarded products, including Apogee Sympony, Avid HD, Crane Song HEDD, Lynx HiLo & Aurora, and Prism Orpheus.

The PCM4220 ADC that I just discussed is used in Focusrite ISA428, ISA428, and that basic filter topology appears in a number of cheaper multi-channel TI chips which are probably more commonly used.

David L. Rick
Old 2nd March 2020
  #1451
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I have to say, aside the bad engineering principle of using a lot more resources than necessary, I see your point.
It's vastly based on paranoia and disregard of all the studies conducted so far on the matter, but to each their own, I guess.

And in fact, strictly limited to the digital part of the chain, I repeatedly said that it's not that I am against recording at higher sample rates, but rather against who claims that recording at 44.1 is detrimental.
To this last point, I consider the long list of highly priced converters that rely on non optimal chips as further proof that the ridicule aimed at the boutique audiophile crowd is fully deserved.

Luckily, there's examples of reasonably priced devices that do things just right at any sample rates. I just linked to one.
Old 2nd March 2020
  #1452
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Analogue Mastering's Avatar
Also, 44.1 being the main destination, at least for audio, one would like to prevent unnecessary SRC, as this will harm the audio more than any filter beyond hearing range.
Old 2nd March 2020
  #1453
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doom64's Avatar
Quote:
Originally Posted by Govier966 View Post
Hypothetical sample rate related question:
Say you record a project at 96K and artists want to add their own tracks recorded with their own interfaces. Most consumer interfaces do not support 96K.
How do you proceed?
Plenty of consumer interfaces have supported 96 kHz for years.

Assuming the musicians didn't go with that option, I would upsample. Unless more than about 80% of the tracks are the musician provided tracks then I would consider doing a high quality sample rate conversion to 44.1 kHz.

Whenever I record pop or rap artists, 99 times out of 100 they bring 44.1 kHz instrumental files but I'll track their vocals at 96. The instrumental is usually finished and a stereo mixdown at that point so it's a 50/50 decision.

Like @ David Rick mentioned above, you never know when you're recording history that millions of people will care about in the future. The Who could have recorded to U-matic tape but went with film instead. And in 2019, they scanned their concert footage at 4K resolution.



24/96 or 24/192 is my "4K resolution". Since the year 2007, I've been recording 24/96 and haven't looked back.

Quote:
Originally Posted by Analogue Mastering View Post
Also, 44.1 being the main destination, at least for audio, one would like to prevent unnecessary SRC, as this will harm the audio more than any filter beyond hearing range.
What's harming audio is the perpetual loudness war. SRC with high quality software such as iZotope RX, r8brain Pro or SoX is a lot less damaging vs. digital clipping.

Seriously, what is wrong with some of you mastering engineers releasing music like this:

http://dr.loudness-war.info/album/view/172252
http://dr.loudness-war.info/album/view/172221
http://dr.loudness-war.info/album/view/171708
http://dr.loudness-war.info/album/view/171837
http://dr.loudness-war.info/album/view/171404

????

Last edited by doom64; 2nd March 2020 at 01:19 PM..
Old 2nd March 2020
  #1454
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Quote:
Originally Posted by doom64 View Post

What's harming audio is the perpetual loudness war. SRC with high quality software such as iZotope RX, r8brain Pro or SoX is a lot less damaging vs. digital clipping.

Seriously, what is wrong with some of you mastering engineers releasing music like this:

http://dr.loudness-war.info/album/view/172252
http://dr.loudness-war.info/album/view/172221
http://dr.loudness-war.info/album/view/171708
http://dr.loudness-war.info/album/view/171837
http://dr.loudness-war.info/album/view/171404

????
I disagree with pretty much all you said, but I back you up 100% on this.
Loudness war is what killed sound quality. Lack of musical talent did the rest.

It's all a matter of quantifying each source of distortion in respect to the others. Look at the power and frequency distribution of each kind of noise/distortion and rank them in order of priority.
If chasing after the potential necessity of ultrasonics, at the expense of degrading the quality in the audible range, is that important for somebody, then let them use ever incresing sample rates.
However, with yours and David's line or reasoning, I can see no limit to the sample rate/bit depth ever increasing specs.

Do we really want to kick audio fidelity down to the ground even more, by following these fool's quests? I think it's doing a more than egregious job at it with magic wooden blocks, small resonance absorbing bowls, cable raisers, directional cables and what not.

I feel like I'm in that scene in Casino, when they beat Joe Pesci's brother to death in front of him, and he begs to let him be because he's still breathing.

Let audio fidelity gain some more credibility first. It's still (barely) breathing. Is that too much to ask?
Old 2nd March 2020
  #1455
Lives for gear
In rock music, the audio fidelity is a nonsense. The sound quality is a part of the works. A picture of the moment.
Remastered the Beatles is an artistic crime.
Old 2nd March 2020
  #1456
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Quote:
Originally Posted by sax512 View Post
I disagree with pretty much all you said, but I back you up 100% on this.
Loudness war is what killed sound quality. Lack of musical talent did the rest.

It's all a matter of quantifying each source of distortion in respect to the others. Look at the power and frequency distribution of each kind of noise/distortion and rank them in order of priority.
If chasing after the potential necessity of ultrasonics, at the expense of degrading the quality in the audible range, is that important for somebody, then let them use ever incresing sample rates.
However, with yours and David's line or reasoning, I can see no limit to the sample rate/bit depth ever increasing specs.

Do we really want to kick audio fidelity down to the ground even more, by following these fool's quests? I think it's doing a more than egregious job at it with magic wooden blocks, small resonance absorbing bowls, cable raisers, directional cables and what not.

I feel like I'm in that scene in Casino, when they beat Joe Pesci's brother to death in front of him, and he begs to let him be because he's still breathing.

Let audio fidelity gain some more credibility first. It's still (barely) breathing. Is that too much to ask?

A bricked mix doesn't mean the multitracks aren't pristine. Marketing 101 indicates.....if "history" is what that recording is....it can be remixed 30 years from now for a more sedate crowd.

It's guaranteed though, that the original listeners that made the recording a hit via sales....will scream for the bricked version they knew and grew up with.
Old 2nd March 2020
  #1457
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Quote:
Originally Posted by David Rick View Post
I happen to think the difference between 44.1k and 48k is obvious on certain program material (as do many others) and I've now posted more than one technical analysis giving bench-measurable reasons for that. I personally find the difference between 48k and 96k considerably more subtle (though worthwhile for my kind of work), but you're correct: I don't have an equally rigorous argument for why there's any audible difference there. All I have are some tentative hypotheses, which may or may not prove true under careful experimental study over the course of many years. We're not going to know the answer anytime soon.

So what should we do in the meantime? Probably whatever carries the least risk of harm to musical posterity. As Bob Olhsson said in this very thread, "You never know when you're recording history."

Let's examine the alternatives in the spirit of Pascal's wager. You argue that the key problem is that frequencies above 20k produce such serious IMD on playback though loudspeakers that they should be proactively filtered out in the recording chain. I counter that numerous golden ears think there's something above there that's worth preserving. Our choice is simply whether to record that extra octave or not. Well, if we do record it but you turn out to be correct on the IMD issue, then we will always have the option of removing that top octave some time in the future. With more and more speakers becoming digital, they could even have the required LPF built in. On the other hand, if we don't record that extra octave and one day learn that there is some perceptual utility to the information we stripped out then nothing we can do later will bring it back -- it's gone forever. I'll freely admit that, as an archival recordist, I have a problem with that.

In summary, 2x recording may not be easily justifiable based on today's understanding human perception, but there's little harm in doing it and we keep our options open for the future.

David L. Rick
If I record a project at 96, run the mix realtime to a second synchronized 2-track daw (via Laylas)...with the 2-trk set for 48k....and give the mastering engineer that 48k mix ....what horrors do your math indicate?

To me, yeah, it's a pass out of daw #1 d/a...via patch bay and cables...in to daw #2 and a/d....

but it always sounds pretty good to me (though I hardly ever mix my own stuff any more).

Src gets a bad rap. Fine. I don't see damage in the alternate choice to get to 48 or 44.1 or whatever....a/d for tracking.....d/a and a/d ti get to the mixdown daw.

Opinions? Math?
Old 2nd March 2020
  #1458
Quote:
Originally Posted by Govier966 View Post
Hypothetical sample rate related question:
Say you record a project at 96K and artists want to add their own tracks recorded with their own interfaces. Most consumer interfaces do not support 96K.
How do you proceed?
Send them stems at 48k. Then up-sample their overdub tracks to 96k with the best SRC tool you have available. (NB: This doesn't make them any better than they were at 48k. It just lets them live in your project without dodgy real-time SRC.)

There are relatively few interfaces that can't do 96k today, but some people are running "free" or "lite" DAW versions that won't. If that's the case, all you can do is send them 48k stems to discourage them from recording at 44.1k.

David L. Rick
Seventh String Recording
Old 2nd March 2020
  #1459
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Quote:
Originally Posted by thenoodle View Post
A bricked mix doesn't mean the multitracks aren't pristine. Marketing 101 indicates.....if "history" is what that recording is....it can be remixed 30 years from now for a more sedate crowd.
Sure. But I'd like to listen to the non clipped mix now, instead of hoping that in 30 years somebody will take the pristine tracks and make a decent mix out of them.
Old 2nd March 2020
  #1460
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Quote:
Originally Posted by sax512 View Post
What do you use?
ADC side Gracedesign M802 with AD option, plus older Grace units into Soundscape/Apogee co-designed AD converters. Don't know which chips are inside. They predate the SSL units, which sound problematic in comparison. I know, because I own and use one constantly, mainly for AES/Madi conversion.
I do remember the M802 was not using standard chip built-in filters.

DAC is the Benchmark DAC2, which won clearly at 44.1 against the big names like Weiss, Bricasti, etc ... and uses oversampling to get rid of the filter conundrum.
The ADC was tested against other big names. Years later against newer more integrated stuff, one of the manufactureres even admitted they deviated from their older custom filters towards the use of the standard on-chip filtering. Needles to say, the older ADCs we use still hold up against the new generation.

Why ?

If, on the other hand, my ADC and DAC chain is so utterly bad, as to make 96K sound as bad as 44.1K - which could be entirely possible - this would implicate almost all the rest of the available ADDA the last 15 years would be even worse ...

Admittedly, most of my tests were at 44.1K. As this involved existing cd recordings, and I followed the logic if it sounds better at 44.1K, it should be designed better overall.
Old 2nd March 2020
  #1461
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Fourier series representations.
https://www.youtube.com/watch?v=r6sGWTCMz2k
Old 3rd March 2020
  #1462
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cjogo's Avatar
Next someone will say :::: 96 sounds better than that analog tape >>> I listened to for hours on my 8 tracks >>> back in the 60's
Old 3rd March 2020
  #1463
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Quote:
Originally Posted by Yannick View Post
DAC is the Benchmark DAC2, which won clearly at 44.1 against the big names like Weiss, Bricasti, etc ... and uses oversampling to get rid of the filter conundrum.
The Benchmark also uses ASRC to vastly reduce jitter, which is yet another factor during playback.

I personally have no problem with good quality ASRC - it fixes more problems than it creates.
Old 3rd March 2020
  #1464
M2E
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I'm not sure if this has been posted as it's really new but, this ends the debate and exactly what I thought from the beginning.
Aliasing, sample rates etc, is all mumbo jumbo.
Just record and mix at 48k. You'll be fine.
Check it out. An incredible video by Dan Worrall from Fabfilter.
He goes in depth but also keeps it extremely simple fro everyone to understand.
I love it.

https://www.youtube.com/watch?v=-jCwIsT0X8M
Old 3rd March 2020
  #1465
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sax512's Avatar
 

Quote:
Originally Posted by M2E View Post
.... this ends the debate...

https://www.youtube.com/watch?v=-jCwIsT0X8M
Not if one is convinced that the ultrasonic content is somehow audible, as some people here seem to be.
Old 3rd March 2020
  #1466
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Quote:
Originally Posted by afa View Post
Excellent video from Fabfilter regarding sampling rate, aliasing, intermodulation...

This video is amazing. This parallels with my findings and why I stick with recording at 44.1-48kHz. There are pros and cons to different sample rates. But this video really confirms my stance on what sample rate I record at.
Old 3rd March 2020
  #1467
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Quote:
Originally Posted by Flippy Floppy View Post
This video is amazing. This parallels with my findings and why I stick with recording at 44.1-48kHz. There are pros and cons to different sample rates. But this video really confirms my stance on what sample rate I record at.
Yes, indeed the video is an eye opener. BUT it also shows that you cannot just use a myriad of plugins, unless you know exactly if/how upsampling is applied. Its a good opportunity to go through the VST folder, do critical testing and drag a lot of stuff into the trash folder. So this is my conclusion to keep stuff I can trust from 1-2 developers only and to the rest OTB.
Old 3rd March 2020
  #1468
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Analogue Mastering's Avatar
Quote:
Originally Posted by Flippy Floppy View Post
This video is amazing. This parallels with my findings and why I stick with recording at 44.1-48kHz. There are pros and cons to different sample rates. But this video really confirms my stance on what sample rate I record at.
Ditto!
Old 3rd March 2020
  #1469
Gear Guru
 

Quote:
Originally Posted by sax512 View Post
Not if one is convinced that the ultrasonic content is somehow audible, as some people here seem to be.

those who have used the right body wash and opened up their skin pores
Old 3rd March 2020
  #1470
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Quote:
Originally Posted by joeq View Post
those who have used the right body wash and opened up their skin pores
You stinky 44.1 kHz apologists. Wash yourself!
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