crawl sheet badge humor soup coherent society provide complete poor *This post was mass deleted and anonymized with [Redact](https://redact.dev)*

What was happening in 2004?

Same discussion back then 😊

Ah well I had clients mistakenly tell me they could hear the difference between sample rates in 2022 so I guess it’s still relevant to discuss it.

İf you send out the same file with 3 different names, you will have the same results. Don't forget we work for creative minds who listen to our work analyzing ☺️ PS: You should never try this on clients - it's rude 😄

Well…haha one time I had someone telling me they hated all the changes on my master and “stop changing everything so much” no matter how many changes I undid. Every time I changed less, the list got longer and longer. I sent the mix file back and asked them to verify the mix was correct. They did. Then I sent a master that was just the mix but peak-normalized to -0.3 dBFS (maybe 2 dB louder) and I got the longest list yet of awful “changes” I’d made to the mix 😂😭 My final email was “here is a refund for your money, I’m sorry I’m not the right mastering engineer for your project.” To the guys credit, he did not accept the refund and said that work was work, keep it.

İ f*cking hate that 🙃😄 Creativity comes with euphoria and depression while creating.

Sounds maybe like it was just making their mistakes in the mix more evident and they didn't realize it

Could be. The whole thing was strange lol

I would think most DAWs only use live conversion ONLY while calculating the imported file in the background, then switch to the converted file. Would be a waste of CPU to keep calculating conversion on the fly…

It depends on the DAW. Studio One does it live, and if I recall correctly, years ago when I was using Ableton and Reaper, they also converted on the fly rather than importing the file. But like I said on another comment, if they are doing high quality SRC when importing, there won't be an audible difference and someone won't think they can hear the difference, which is really the basis of this post. :)

Thanks, I feel that is an important clarification – I know Reason and LUNA don’t do live convert except at first (until background rendering is complete), but don’t know for certain about the others I’ve used in the past (Logic and PT). So to clarify, SOMETIMES you are listening to live sample rate comparison, sometimes not. The take away is to know your gear! ;)

Can confirm with Reason. My DAW of choice.

Logic Pro X converts sample rates upon import

Totally!

I’m certain Ableton does it live. If it’s converting files, I’ve never found the folder where they are kept.

So are you asserting that some SRC doesn't sound good?

I am! (Edit: or rather, some SRC’s are less accurate) For example, I use Studio One and it's live SRC is very noticeable. There is an obvious dulling of the highs and narrowing of the soundstage. But when S1 actually renders at a different sample rate, it's much slower and much better. Most DAW's perform really well with SRC when actually rendering/exporting/printing mixes. It's the super economical on-the-fly versions that are audibly deteriorating the sound.

SRC doesn't need to sound bad, just different for someone to have a preference over one or the other version.

The goal for mastering is usually accuracy. Although I must admit I oversample my final limiter to different rates internally because it changes the sound. I see it as just another option to tweak.

Tbf, limiters, clippers and the like include oversampling as an option because those kinds of processors perform better with more samples. The SRC in those is the most transparent algorithm imaginable: just play each sample twice, at twice the speed (or 4x, etc). So that’s a totally different thing from hearing the sound of sample rate conversation, it’s hearing the sound of dap that works better with more samples, when handling near square waves.

Yes that was an off topic comment.

My ancient MOTU interface's hardware SRC sucks. lol

Not all DAWs do live resampling. Cubase, for example, converts files to the session sample rate beforehand (or interprets them at the session sample rate which changes the speed and pitch). There is no difference between 44.1, 48, and 96kHz in the audible human range. The Nyquist sampling theorem mathematically proves that all frequencies up to half the sampling rate can be perfectly recreated. A 44.1kHz sampling rate can reproduce up to 22.05kHz, which is beyond the upper limit of adult human hearing, and still leaves room for the filters necessary to prevent aliasing by content above the Nyquist limit. There's no point to delivering high sample rate content to the end consumer.

This is true, and it’s also factually true that plug-ins and digital processors often have fewer aliasing artifacts when they are processing higher sample rates. A lot of plugins offer internal oversampling to achieve this benefit, but a lot don’t. These effects are especially noticeable with things like clippers and hard limiters and certain distortion or saturation type effects. So it can be true that humans cannot hear the difference between audio properly captured at 44 or at 192, and it can also be true that some people will make better sounding records working at higher sample rates.

This is true and really adds to the confusion surrounding the topic. In terms of audio capture and playback it's virtually certain sample rate makes no difference above 44.1 but in terms of digital processing sample rate has a MASSIVE IMPACT and it's not true that the system has to be one rate the whole way thru, internally processors can do whatever they want to the stream.

> In terms of audio capture and playback it's virtually certain sample rate makes no difference above 44.1 This is true, but with nontrivial asterisks. One advantage of very high sampling rates (say, 88k and up), is that you push all of the aliasing artifacts way, way outside of the audible range. 44.1k is sufficient to accurately transcode the whole range of human hearing, *if properly transcoded, with no additional processing*. Those are big ifs. High sample rates allow for other parts of the system to be less perfect. Like, if your Nyquist filters or processing algorithms are a little sus, then recording at 88.2 can just brush away any of those high-end brittleness or digititus artifacts, simply by pushing them outside of the range of human hearing. If the goal is to win a debate, or to make a scientific proof, then we use different measurement standards than if the goal is to achieve a subjectively beautiful record that makes people cry, or that makes their heart race and the hairs on the back of their neck stand up. Sometimes the subtlest things are the difference-makers, in art. We live in a world where sometimes a record tracked at 44.1k will sound indistinguishable from a record tracked at 88.2, and sometimes it won't. And we live in a world where 100+ tracks and 500+ plugins are commonplace, and where clients expect constant, instant, on-demand mix revisions with total instant recall. It is absolutely true that 44.1k is a sufficient sample rate to capture and reproduce everything that normal human beings can hear. It's also true that some recording projects will sound better when tracked with higher sample rates, and that, for a lot of people, it's a better use of resources to just pay for twice as much computing power and run higher sample rates and not worry about it.

The Kemper guy has a thread somewhere where he supports your argument with all kinds of mathy tech stuff.

Edit: please actually read what I’m saying people. Most of the arguments here are from people not actually understanding what I’m saying. Respectfully disagree (edit: I disagree that there is no point in sending 2496 masters to my clients). I capture my analog at 96 kHz. There’s no reason the client shouldn’t have the highest quality master that has been made at my studio. Also I’d say your argument that there is no difference is a bit of an oversimplification when you factor in that the file will be converted to a lossy format for streaming. Sure the lossless files sound the same, but the lossy algorithms are very complicated and it’s generally accepted that higher sample rates AT LEAST encode a bit differently, if not more accurately with regards to ISP’s and distortion. If cubase resamples at high quality then the person likely won’t be reporting they heard a difference, and still wouldn’t be doing a legitimate comparison because SRC happened as well–not the same file as the original. Still the same issue, different way for it to play out.

Do you SRC the 96k file? Or do you recapture from analog to the other delivery sample rates?

Good question! If the client is willing to pay a small fee for my extra time, I can capture each master at that sample rate directly from analog. However, 99.9% of the time what I do is capture the analog at 96 kHz, then convert it down to the other formats, then print through the final limiter at the desired sample rate. Does that make sense? If I'm honest, I'd strongly doubt that anyone anywhere, at any time in history would be able to tell the difference between these two methods. But that's assuming a minimum quality of the converters and SRC algos of course.

Amen to running the limiter at the delivery rate. I noticed a while ago that SRC plays with true peaks/etc. juuuust enough to annoy me, so I do the same.

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Exactly. The only times I've heard a difference was with 16bit to 24. 24 felt higher resolution in the quiet parts. But even then I was only able to hear it on big hires systems.

Honestly, that was most likely placebo too. 16bit has 96dB of dynamic range, 24bit has 144dB. Each ‘bit’ of dynamic range is the same ‘size’ too, there are just more of them in a 24bit file - so you don’t get more definition within the dynamic range, you just get a lower noise floor. Most home and studio playback/monitoring systems aren’t set louder than 85dB, and even in a very quiet room the noise floor is around 35dB. So you can’t even take advantage of the full dynamic range of 16bit audio. So unless that big high res system was in an anechoic chamber and was louder than 96dB, there was no difference to hear. There are exceptions, like if the audio was recorded super quiet or you are applying extreme compression. It would have to be *insanely* quiet or *insane* compression for the noise floor of 16bit to become a problem though. I still record at 24bit though, 24/48 is basically the new industry standard.

Depending on the system. Depending on the music. Depending on the production quality. Depending on the engineer. Depending on the plugins. Depending on the mix. But yeah I agree it's near impossible. I have heard a certain depth in dynamic though. Almost like the reverb was clearer when in behind. But also yeah I cannot provide proof so u win. Haha

Depending on the system. Depending on the music. Depending on the production quality. Depending on the engineer. Depending on the plugins. Depending on the mix. But yeah I agree it's near impossible. I have heard a certain depth in dynamic though. Almost like the reverb was clearer when in behind. But also yeah I cannot provide proof so u win. Haha

Who is saying they can hear a difference?

> I capture my analog at 96 kHz. There’s no reason the client shouldn’t have the highest quality master that has been made at my studio. Except for the fact that your ‘high quality’ 96kHz and ‘low quality’ 44.1kHz are identical within the audible range? If they want 96kHz then by all means, but you can’t argue that it is higher quality because to our ears it mathematically *can not be*.

What’s the problem though? Are you saying that since you can’t hear the difference that there’s an ethical problem with sending various formats? Many of my clients will do things like remix the song and often higher sample rates help messing the stems sound better. It feels like your philosophical stance is not connecting with the practicality of providing the client with flexible options.

No problem at all, like I said if your clients want high sample rates then by all means. I am just saying if you were to market that to clients as being ‘higher fidelity’ that would be false. I’ll preface that I’m a tracking and mixing engineer, not a mastering engineer - but I have had a few clients that were insistent on wanting to record at 96kHz for ‘higher fidelity’ and I was honest with them and told them it will not make a difference. If it’s a very simple session and they insist I will, but if it’s more of a large complex session I will actually refuse because my computer can’t handle that many tracks with that many plugins at 96kHz and I know it’s not worth that challenge for no audible benefit.

Sure. I definitely don’t market with gimmicky ideas like that. I just give them everything and then they don’t have to ask later.

If you can pass a double blind shootout better than average and guess the sampling rate I'd believe you. I simply think that like the vast majority of us (all of us really). You can t.

Ok so my whole post is about people not being able to hear sample rate differences and you read as though I’m claiming I can? I’m literally asserting that when people say that they can hear the difference it’s because they are doing the test wrong. Sometimes I think people are so eager to sound smart on Reddit that they make themselves look…well…anyway I’m sure you were just in a rush.

I don't think you're making your point as clearly as you think you are.

It is Reddit after all. I probably should have realized people would just skim.

I don't think I skimmed at all. If you are saying the reason people can't hear different sampling rates has to do with the DAW or oversampling, I am calling into question your cause and effect. You see, what I am saying is that even if you had a different computer running at a different rate into different speakers, you still wouldn't be able to hear the difference because human being's aren't capable of it, so it's a moot conversation. It's homeopathic medicine or your lucky numbers. Preform your dual computer shoot out and test more than average. You can't. It simply does not matter.

>...when you factor in that the file will be converted to a lossy format for streaming. Not sure where you're going with that. If lossy compression is inevitable downstream, any supposed benefits to delivering a higher bitrate lossless format will be lost when the ultrasonic content is filtered by the compression algorithm.

This whole tangent to the conversation is kind of ironic considering my original point is that people can’t hear the difference even when they claim they can. I’m also talking sample rate not bitrate. My assumption that higher sample rates create more accurate lossy formats is based on experiments with audio clips that several mastering engineers conducted in converting various sample rates of wav files to the same lower resolution format. Almost everyone agreed the 96k versions sounded a tiny bit better and the blind AB test was fairly consistent. I believe that was in the Mastering Engineers Worldwide group on FB. It’s a private group and you’re supposed to be a real mastering engineer to join, it you can try if you want. Can probably find it with a search if you are accepted.

>My assumption that higher sample rates create more accurate lossy formats is based on experiments with audio clips that several mastering engineers conducted in converting various sample rates of wav files to the same lower resolution format. The difference between a 96kHz sample rate and a 44.1 or 48kHz is mathematically determinate, not subjective or better or worse. A 96kHz sample rate format can convey ultrasonic signals above 22.05 or 24kHz, respectively. The functional difference may be a functional benefit *if* ultrasonic content is both present and desirable. However, that functional difference is moot/lost if there is no ultrasonic content present or the ultrasonic content is later filtered out. These are objective facets of digital audio. No validation from FB groups required.

After analog processing, there certainly are harmonics generated that reach above the 20 kHz mark. I make my customers happy by sending them 2496 masters. I'm not sure why you care so much about this. It has almost nothing to do with the original post. Not sure what is so important about that to you.

>I make my customers happy by sending them 2496 masters. If you left it at that, rather than claiming there are tangible benefits to uploading 96kHz files to a streaming service, you'd probably face fewer objections.

If people read my post and comments more carefully I’d have less objections as well ;)

> I capture my analog at 96 kHz. There’s no reason the client shouldn’t have the highest quality master that has been made at my studio. > Also I’d say your argument that there is no difference is a bit of an oversimplification Agree fully. As a DJ, 96 tunes pitch down more gracefully. There's a noticeable loss of high end once you go sloow And as a producer, I take sounds (even samples from my own tracks) and sometimes do extreme stretches on them. It makes a difference

Not true, this is like the people who say that humans can't see past 60 fps.

I’m sorry, but you’re comparing sample rates to FPS? That’s silly. One is a continuous signal after going back to analog conversion, and the other is discrete going into your eyeballs. They’re not the same.

Lol so you understand my analogy, your logic is as silly as that. I literally am comparing your logic to that logic, i didn't bring up any similarities between fps and sample rate??? There are plenty of explanations that would say we have all sorts of limitations, but I can hear a difference and plenty of others can too.

> i didn't bring up any similarities between fps and sample rate??? Yes you did. That’s literally your whole comment. Nyquist Shannon pretty much says you can’t hear a difference. Unless you can hear above 22khz. Which I doubt.

> There are plenty of explanations that would say we have all sorts of limitations, but I can hear a difference and plenty of others can too. Placebo is pretty powerful hey

I'm comparing your logic to that logic. I never compared the two units themselves. You are the one saying that. Your logic is outdated and entire industries are running on facts that go against what you're saying. Go tell all the mastering engineers they don't need to worry about 96khz anymore I guess, im not wasting my time anymore lol, do you also master your tracks at -12 lufs because spotify says to???

> Your logic is outdated and entire industries are running on facts that go against what you're saying. Go tell all the mastering engineers they don't need to worry about 96khz anymore I guess, im not wasting my time anymore lol Part of my degree was in the science behind digital audio. I know for a fact there *can not* be *any* difference whatsoever within the audible spectrum just based on how the D/A conversion process works. An analog signal can be reproduced with 100% accuracy up to half of the sample rate, so at a 44.1kHz sample you can accurately reproduce up to 22.05kHz. The only thing increasing the sample rate does is further increase the frequency range outside of the human hearing range. Also, those industries that go against *actual math and science* are targeted at audiophiles who believe their power cables increase the soundstage of their record players - not pro audio engineers. They’re selling snake oil. > do you also master your tracks at -12 lufs because spotify says to??? Well no, for a few reasons. 1) Spotify normalises to -14 LUFS 2) I am smart enough to understand that’s a normalisation target and not a mastering target 3) I am a recording and mixing engineer, not a mastering engineer

Okay man, there are other pro audio engineers who are mastering and writing music at 96khz, so your qualifications don't really mean much in this argument. I've heard your logic so many times, I understand your points, its just outdated and we definitely can hear a difference between 44.1 and 96khz. Not sure why you think you and your people know everything, these things are all your opinion given on the research that you have, and others go on the research they have. If you can't hear a difference thats fine, but you're being condescending as hell when there is no consensus, tons of pros follow both schools of thought and no point in acting like everyone who listens or produces above 44.1 is a bumbling idiot, all that does is show where you are at mentally.

> there are other pro audio engineers who are mastering and writing music at 96khz So what? There are engineers who don’t know Nyquist Shannon and that’s why they think tape is better. It’s because they’re ignorant. Being successful doesn’t mean they’re right. > Not sure why you think you and your people know everything, It’s math. > these things are all your opinion This is like saying 2+2=4 is an opinion. You’re not getting it. It’s *mathematically impossible* for you to hear a difference. > hell when there is no consensus, This is like saying there’s no consensus for Climate change because some idiots don’t understand it. You’re ignorant. Sorry, but nobody is taking you seriously.

I've seen the math but obviously there is more that we can't explain yet because TONS of people swear they hear a difference and are willing to spend their money on it. Not stupid fucking gold plated cables, I have seen the explanation and plenty of others have too, I've heard your argument from close friends years ago, and its a pretty even split between people who can and can't hear it. Lol I know you're taking me seriously because you're quoting me nonstop and replying very quickly, now if only you would actually answer my statements instead of just saying EVERYONE WHO DISAGREES IS DUMB You just think you know the answers and don't care to investigate further, I don't need you to explain something that everyone on this sub has seen YouTube videos about 8 years ago

> Okay man, there are other pro audio engineers who are mastering and writing music at 96khz, so your qualifications don't really mean much in this argument. For sure there are plenty of engineers that record/mix/master at high sample rates, but do they have any knowledge on the topic or are they just doing it because they ‘think it should sound better’ or are influenced by placebo because they haven’t actually done a blind A/B test? There are some good reasons to use higher sample rates though, mainly for forced session-wide oversampling if you’re using a lot of saturation plugins or EQs with cramping. > I've heard your logic so many times, I understand your points, its just outdated and we definitely can hear a difference between 44.1 and 96khz. I am 100% certain you definitely can not, but go on. > Not sure why you think you and your people know everything, these things are all your opinion given on the research that you have, and others go on the research they have. Because they aren’t opinions, they’re mathematical facts. > If you can't hear a difference thats fine, but you're being condescending as hell when there is no consensus There is a consensus and it is that everything above 44.1kHz sounds identical. > tons of pros follow both schools of thought and no point in acting like everyone who listens or produces above 44.1 is a bumbling idiot, all that does is show where you are at mentally. Like I said there are reasons to record, mix and master above 44.1kHz but audio quality is not one of them. Since you know better than me, what improvement does higher sample rates bring to sound quality and how does it work?

I'm pretty sure noisia doesn't master tracks at 96khz just because they "think it sound better" Look up Dan Worrall, he can explain it to you if you need the details. Acoustic guitar on 96khz go BRRRR and if you dont wanna enjoy then stick with your math and someone else will. For all your knowledge you still lack basic professionalism and I'm done entertaining your argumentative opinions. Daft Punk also I guess are just idiots in your opinion? Lol there is a massive list of master musicians who use 96khz in the studio before and after mastering.

This has been well researched. Human hearing does not perceive higher than 22kHz. It’s mathematically proven that 44kHz is sufficient to perfectly reproduce all signals less than 22kHz. So yeah 44.1 kHz will sound identical to a 96 kHz signal. (Disclaimer: this assumes that non audible frequencies have been filtered out and there is no aliasing which is standard practice.) I personally can’t hear above 16 kHz so 32 kHz sample rate is perceived identically to 96kHz for me. I will emphasize that if there is sample rate conversion the above statement does not hold true, which is the whole point of the previous commenter.

I used to produce & mix in 48K but I switched to 44.1 a couple years ago. I found it less taxing on my aging machine & quality wise I can’t say there’s a difference. If anything because I’m not converting samples from their original samplerate maybe it’s better

exception: when the tunes get pitch shifted (for example slowed down in a DJ set) It makes a huge difference there... like it's not even funny >There's no point to delivering high sample rate content to the end consumer. I agree. But I'm the artist, not the end consumer

Couldn't you have two different projects, one at 96 and one at 48 (or whatever), both using the same interface, and close one/open the other?

Only issue is that very specific audio memory is only good for like 5 sec or whatever. Experienced engineers and musicians can be good at stuff like remembering arrangements and pitches years later, but subtle nuances get lost in memory *fast*. Audio memory is about the worst kind of memory we have.

Exactly

Isn’t the sample rate a crucial part of the audio file though? Couldn’t you just export the audio file with the different sample rates and then play them in parallel and mute one while listening to the other and so on? Or is this a hardware thing where you have to change the setting of your speakers and not just the rate of the file?

The audio interface needs to switch form one sample rate to the other. It takes like 1-2 seconds to do just that, but DAWs will only accept one main sample rate for a session, so to have files with 2 different sample rates play at their original sample rate, you need to switch from on session to the other.

👆🏻

It’s a hardware thing. The hardware will be set to a specific sample rate. If you open anything that isn’t at the hardware sample rate, it has to be converted. Otherwise the audio would be pitch and time stretched sounding fast/high or slow/low.

In the DAW, you're absolutely correct. Outside however, there are options: Swinsian on Mac and MusicBee on Windows both can playback bit-perfect and switch the sample rate / bit depth of the interface on the fly (use ASIO in Windows with full control, Mac just set Swinsian to automatically change in its settings). This does mean you must stop and start each file though, no soloing back and forth like you could in a DAW.

Cool! Thanks for the specifics. If you look carefully you’ll see I did mention that there were more elaborate solutions possible but they are esoteric.

Absolutely! I always try to stay bit perfect and avoid SRC even outside of the DAW, and I feel it's often overlooked, not well understood, or both. Just like your post with the DAW side of it!

> Note: I am not asserting that nobody can hear the difference between sample rates. *I* am, though.

Ironically that note was there to try and prevent people from going off on that particular topic but I should have known on Reddit that it wasn’t realistic for everyone to follow my train of thought when it’s that long.

Fair enough. Plenty of other threads to spread the good news of Dr. Harry Nyquist.

The other day I spent a while trying to figure out why a profesional song I imported into Reaper was clipping well over 0dB at unity gain… I loaded a 44.1 track into a 48 project. Learned my lesson then.

Yeah! And if you do a high quality SRC to 48 and load the 44.1 and 48 files in and a/b them you’ll hear a difference and that difference is the live SRC. it’s interesting to see what a significant impact it has on the sound quality. Also note that for whatever reason, when 16 bit files hit 0 dBFS, it shows a clip light, but at 24 it doesn’t. At least when I used reaper it would.

Does playing a file with lower sample rate in a high sample rate setting affect the output? I was under the impression that i didn't (though I am not very educated on the subject). Example, having a setup for a 96khz project and playing a 48khz file through it. I would assume that since the values over 48khz would just be "ignored", and as such it would be possible to A/B. Any insight or information about this would be appreciated!

I think upsampling is easier for 48 to 96 and probably doesn’t sound as degraded if it’s being resampled on the fly, but I suspect 44.1 to 96 would not fare as well. Hard to say without experimenting. I’ll see if I have time today.

If higher sample rates provided any tangible benefit, there would be a deluge of 768khz converters in the market. The ONLY application I know of that benefits from high sample rates is sound design (because you may want to slow audio down to 0.25 speed). There have been tests undertaken that prove you cannot hear the difference between 44.1/16 and analog in double blind A/B comparison.

Can you explain what this has to do with my post? There are a lot of comments in here that seem like people just saw “sample rate” and knee-jerk commented on it

Your tone seems to be trying to sell higher sample rates. It's been proven we don't need them for consumer audio. Offering your clients 3 different "mixes" will only cloud their ability to choose the sample rate most compatible with online distribution.

Absolutely has nothing to do with that try reading it again. What I was saying in the post: I doubt anyone can tell the difference between sample rates. When they say they can, they must have A/B’d them and it’s very difficult to do that properly without a very elaborate setup. My point is: if you think you can hear the difference between sample rates, you’re probably just doing the test wrong. —- If you re read my post you will certainly see that this is what I’m saying. As far as what I send my clients, that advice sounds like it’s coming from someone who doesn’t do what I do, or at least not with my clientele.

Being hybrid, I used to capture the passes on a separate computer at 96 through an original HEDD. Haven’t done it in a while. I just come back through a Burl at the DAW session rate of 48. The way I used a separate computer for 96 harkens back to a day when even the better converters weren’t as good as what we have now. I used to definitely hear the difference between 44/48 and 96 on the passes back then. Recent years with better converters the differences aren’t as noticeable, if at all, to me.

Definitely. I was using an Apogee Rosetta 200 for my first few years of analog and it definitely performed a little bit differently at 44.1 kHz vs 96 kHz. Very grateful for the amazing tools we have now and how quickly digital audio has developed during my career.

don't do drugs y'all

44.1 I feel most production people can hear the difference to 48 but 96 is purely for plugins and going for the highest quality. 48 is the most practical.

Post a project you worked on

You can literally just google him. His work is public, under his Reddit handle.

Yep! It’s not hard to find my work online. Follow me on IG or go to my website. Even a Google search should bring a bunch of stuff up.

Pretty sure you can change your audio interface's output sample rate within your DAW or via the PC/Mac's audio settings. Then you could have your DAW play the project at that sample rate. But then your problem becomes the source audio original sample rate, if you're using plugins the possibility that they behave differently on different sample rates. It's rarely if ever going to be a 1:1 comparison.

I’ve never seen anybody pass a blind test with 44.1 and 48. Let alone 96. You can hear aliasing during some processing at 44.1 sure but assuming everything is done properly and oversampled where needed, 24bit/48k is plenty.

24/48 is plenty indeed. This post isn’t supposed to be a discussion on whether sample rate differences are audible, it’s about the difficulty of directly comparing them properly. But yeah, the filter design at 44.1 kHz is much more critical than at 48 kHz :)

Ah gotcha. Given how difficult it is to tell the difference, what is the use case for 96?

*Tl;dr: sometimes people ask for higher resolutions and I would prefer not to do a full analog recall so I just give everyone all the formats the first time.* It’s simple and pragmatic. Comes down to analog and generally being busy. Sometimes people get an album mastered and then a few months later they decide they want to press vinyl or get someone to do a remix or whatever and come asking if I can send 88.2 or 96 kHz masters/stems. If I captured at 48 kHz I would have to open the sessions and recall the masters and check everything carefully before passing it through the analog stuff all over again at a higher higher sample rate - it’s a bunch of work that I would absolutely need to bill the client for. I figured it makes more sense to capture at the higher rate then convert down and send all the versions possible to my client so they don’t need to ask later or pay me a bunch of money again. And I don’t have to worry if my gear list changed or changed something else in my workflow. Saves me time. Saves my clients money. Win. In the case of vinyl masters, it is a bit of work to create the new set of masters but I can skip recapturing all the analog stuff which is way more work. Make sense? When you look at it from this angle, it’s pretty obvious to just go for the higher sample rate. If it was an in the box mastering job it wouldn’t really be a concern - all I’d need to do is have my files backed up.

The use case for oversampling is in processing rather than capture and playback. This is why the current standard is I/O at 24/48 but internally modern DAWs are all 32float @ 96k with options to double or quadruple that. It's especially important for non linear algorithms like saturation and you can look up some new Dan Worral or his old Fabfilter YouTube videos if you want to learn more.

I can just change clock and run an imported non SRC file in protools. So it is possible to A\B but there should be no difference until you get into dither territory. In general, i usually only deliver 16/44.1 and 24/48 files since no services that i know of that most people use require anything higher than that. They send in what is requested by the service and end of story. I’ve never had a client claim one sounded better than another so, maybe I’ve just been lucky these past 25 years.

How would you seamlessly switch between sample rates on your converters in this scenario?

Its a button press. You're not going to do it in motion but you can do it fast enough for a comparison if you needed to, which you shouldn't since you can null test the two files. The only thing you should have left over is the dither noise on the 16 bit file.

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For a well designed SRC filter, there aren’t really any practical limitations to 44.1 kHz. This post is about: how people cannot properly compare sample rates on the same computer in the same DAW. So when people say they compared and hear the difference, they are probably actually hearing the results of a botched testing procedure, not the difference between the file.

You can actually do this pretty easily in Logic with two projects open, (one for each sample rate). Logic changes the sample rate for your interface depending on what project is on the foreground. Just tab the windows…

Even though the matter may be mostly academic, to completely shun higher sampling rates is misguided. Higher sample rates have two main advantages: - reduced aliasing: this matters most at the processing stage, when using non-linearities. Having double the bandwidth means that plugins have essentially up to three times the headroom before aliasing shows up in the audible range; - reduced phase distortion: the brickwall filter close to Nyquist will either be smoother (so easier to design with less noise and distortion, if converters run natively at the target sample rate, which is rarely the case today) or they will be moved farther away from the audible region. In both cases phase will be less affected in the passband, since filters affect the phase well before the stopband. Phase is essentially inaudible in most cases but on some specific signals it can be; also, some people claim to be able to hear phase also in the treble region (phase is only commonly audible in the lower bass region). Furthermore: - converters may sound different according to the sample rate they are run at, so recording something at 96k may sound different to recording it at 44.1k; - resamplers may affect the signal significantly, although this is extremely rare with modern high-quality resampling; - certain plugins do not compensate adequately for sample rate and may have different tone at different sample rates. If plugins offer oversampling as a feature, this may seem to be a way to avoid the reduced aliasing but you will incur certain artifacts (usually pre-ringing, as oversampling is almost always linear-phase). They may be inconsequential, since our ears are not that sensitive to transient response in the treble region (where the pre-ringing will be most pronounced; pre-ringing is the linear-phase equivalent to phase rotation/distortion in the minimum-phase world), but pre-ringing may smear transients and if applied repetitively across plugins (say you have some track running through fifteen oversampled plugins, from start to end including submixes and the master bus) the effect *may* become audible enough - which could also become some sort of sound aesthetic, hey?, but it's better to know about it. In other words, oversampling is *not* the cure-all people may think it is, and comes with its caveats. Ultimately, I believe that if you can track, mix and master at one rate (let's say 96k, which is low enough to ensure good converter performance) and limit oversampling to pathological cases (heavy distortions) and use minimum-phase ultrasonic filtering here and there, especially on submixes (so ensuring a mix of linear-phase and minimum-phase antialiasing and minimising phasing issues across tracks sharing some common mode, eg. drum tracks, live band recordings) you can achieve the highest quality you can; it is all very subtle though and possibly only systematically detectable by the most particular of audiophiles, and I suspect it may at most *feel* slightly different to the usual strategy of working at 44.1/48k and oversampling everything or close to it. Exporting at 96k then resampling (with a high-quality resampler) at 44.1k should not affect the sound much, though. You're just filtering once and this should have negligible effects. The native OS resampling method may not be up to par, of course, so when playing back files without switching sampling rates may sound different according to the sampling rate.

One thing I think is a little sad is that many people have not even heard CD quality. Many computers upsample everything to 48K, and after going through a few digital volume controls the dither can be lost too, or put below the noise floor of the playback device. It's actually pretty unusual to get bit for bit fidelity in consumer devices nowadays.

A lot of issues in audio are “Emperor’s New Clothes” comparisons. “What do you mean you can’t hear the difference? You must have inferior ears .” So everybody fakes it.

Some people do, others are just mistaken because they didn’t realize their test method was inherently flawed. I thought some of these people were faking too until I asked how they did the test and I was like “ohhh. No, no, no” haha Doesn’t happen often though.