Do Sample Rates Really Matter? How to Make the Choice (With Science!) [SSP  

After 19 years of producing and engineering this is the I say THE best explanation of sample rate I've ever heard. Not even when I was in engineering school was it broken down so simply. Thank you very much for this dope video.

This video, and the one on Bit Depth, are an absolute treasure. I'd done several courses in person and online focused on recording audio for classical music, and I'd read a couple of books (Richard King, John Eargle), and I had never understood some very important stuff until I watched these two videos. Huge thanks for the time, effort and generosity you put into sharing your knowledge. You really are a great teacher.

Really helpful! Love your pauses at the end of certain explanations, that's something you don't see much in RU-vid nowadays...

I tell my assistant at the studio " The more someone is worried about sample rates or what kind of converters/clocks we have - the WORSE their music is." It's true. These are the people obsessed with audio quality (even engineers) who don't realize the lyrics suck, the melody is boring, or the track is generic. They worry about 18kHz and dither. They have no clue and point it out by worrying about nuanced minimal things like tech stuff.

Our man out here has such a smooth and articulated speaking voice. A pleasure to listen to.

Thanks Justin, this was really helpful. In the future I'd love to see a similar video about bit depth.

Important to remember that the waves created in air are more like circles / rings in 3 dimensional space, so the the wave on a timeline is a 2D representation of a kind of 3D spiral. Great vid!

I find running plugins that don't do oversampling and create harmonics like Soundtoys Decapitator, there's an obvious difference in sound at 96k vs 44.1k. And when I do a null test there's a big difference in the lows and highs. The differences are hard to hear otherwise.

I don’t understand why this video isn’t getting more likes and views! You got Gold sitting on your channel mate! I’ve been blessed just by watching it Thanks for sharing!

At 96k, every VST synth sounds the best I've ever tried (and there have been quite a few). Even the Omnisphere, in the manual of which it is written that there is no point in using 96k, because all the samples in it are recorded in 44.1k or 48k - sounds better in 96k. Obviously. This is immediately audible, perhaps because all the fx processing in the synthesizer starts to sound better when the sampling rate is higher. The same patches sound different, and not in favor of 48k. This is not fiction, but real long-term experience. When I mix in 96k, I also make fewer mistakes (especially in the high-frequency ranges), the work of the saturators is more audible. Also in 96k latency decreases, and recently in my studio, even guitarists monitored themselves with effects (FF Saturn 2) through the Logic Pro, and did not feel any discomfort! It would be impossible at 48k - fact. Yes, it requires more processing power and disk space, but it's worth it. IMHO.

I’ve heard this info 1000 times but never explained this thoroughly. Like…I understood it theoretically. But now it makes intuitive sense.

Sample rate is not about what you can or cannot hear, but the way you may process the sound.

Absolutely brilliant if you can explain it in a simple way it makes you smile then you know this man knows what he is talking about. Well done thank you

One of the most underrated podcasts when it comes to audio!

Thanks for a great video, and wonderful explanation about this subject. Must say its the best explanantion and breakdown i have seen and heard! :) Thanks man! :)

48kHz gang we out here

In summary: higher sample rates are useful for when you're using plugins that make a lot of harmonics (like limiters, clippers and saturators) and don't use Oversampling or any other anti-aliasing technique inside. Because of this, delivering a finished song at higher sample rates than 44.1/48 is useless.

Amen, brother! It's refreshing to see true understanding on this subject and not the typical ignorance towing industry hysteria.

I'm half way in the video but I had to write this: Thank you - I'm blind so I can't understand visuals. But in your presentation now I got how soundwave and digital representation of it really works behind the scene. Especially the part where you're having fun explaining the moveing forward and back of everything.

wow. listen. you should have been my teacher in high school. i just listened to 39 min of the most informative sonic education non-stop, and even revisited several parts.. and it didn't even feel like learning. it was just consistently interesting. i have watched _A LOT_ of videos, and you always offer so much more than what is commonly regurgitated, which is why i can't stop watching. i want all of those bits that no one else explains... and this video was the straw that broke the camel's back. this video did it for me. i'm ready to put a ring on it. lol awkward. sorry. #alittlesorry seriously, thank you for making all of these videos!!!!!

Superb information. This video coupled with your bit depth video completely explains digital audio capture. Thank you.

A great video! One thing that's worth a mention, though: if you're really into extreme sound design, that's the one justification for massive sample rates. If you're going to stretch sounds dramatically, it might be subtly advantageous to use giant numbers. I should add, I do that kind of extreme sound design but I still don't use huge sample rates (I use 48k because that's what the games/tv/film industry demands, and as you say in your video there's a possibility that 48k *might* be better than 44.1k so I'll go that far).

Great. I really appreciate your effort in making these podcasts.

Excellent video and brilliantly explained Justin.

Looking fly as always Justin!

My mixer made the decision for me. It can only use 48k, so I just use 48k.

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I am surprised no one mentions sample rate importance when audio gets stretched. Like with quantizing drums, or stretching loops, etc. The higher the rate is, the more you get to stretch audio without loss ;)

As someone below me has already said: sample rates matter a lot more with regards to processing than reproduction. I probably couldn't tell the difference between a sample rate of 32kHz and 96kHz just listening to a record. But I certainly can hear a difference when rendering a mix with a bunch of nonlinear processing at 44.1kHz vs 96kHz. It's a subtle difference, but it costs me virtually nothing so I do the latter. Aliasing artefact might be quiet, but I strongly believe that they're responsible for giving digital audio a bad rep.

This was very helpful thank you!

Bottom line. I don’t care if it’s easier for people to work with lower sample rates than 88 KHz-that should be the minimum-or how exacting you have to be above 44.1 KHz. I don’t care if computers are pushed harder. I don’t care about the hearing limitations of my next door neighbor or the engineer who mixed and mastered something. I don’t care about CD’s low standard. I don’t care about a studio’s economics. I value music and sound quality. No one would tell a great painter to save money by watering down his paint because the average person just doesn’t notice the difference. I don’t care about excuses. I know from personal experience that 88 KHz -192 KHz, when down correctly, sounds noticeably superior and reveals much more information. I know it from SACDs, from DVD-Audios, from Apple Hi-Res No Loss tracks (even when listened to through Bluetooth which dumbs down the signal). Anyone claiming it doesn’t make a difference is either hearing-impaired or a liar.

Nyquist Sampling Theorem translated into lame man’s terms: 44.1 kHz of Sampling Rate creates a perfectly recreatable digital version of the full frequency spectrum a human is capable of hearing, actually, even below and above it. 1-22050 Hz. If you play it back, there is no measurable difference, let alone an audible one. If you can hear a difference on a higher sampling rate, your audio interface cannot capture at that sampling rate properly anymore. Or maybe your playback device cannot play it back correctly. Building a ADC or DAC of higher frequency ranges as perfect as lower ones is very hard to do, especially on the hunt for ever cheaper production cost or selling price tags. That’s as easy as explaining all of this is. But yea... I get it: It sounds too simple to be true with all of that esoteric offerings for better sound out there.

Brilliant video as always. Thank you.

The best explanation I’ve ever heard! Very well explained without trying to sell us on industry dogma. Loved it!!!

Good points, all true. Many audiophiles want to believe that that next piece of gear is sounding better. I noticed recently that my Focusrite Clarett 4 pre, though it samples at very Hi Res rates, on the spec sheet it says that its analog inputs are all limited to 20kHz! So you are sampling non existent frequencies anyway.

Thanks Justin. I've wrestled with that question a lot. For me, it comes down to how much processing power I need to do what I want to do with a mix quickly. I'd rather stay at 44.1 or 48 and not have to offline process or freeze tracks. That slows down the mix and gives me way too much time to second guess everything.

listening to you explaining sounds physics is the best ASMR I have ever experienced.

The 18 dislikes for this video were obviously clicked by disgruntled bats.

A clarification- When you talk about Nyquist, you imply that digital didn't start happening till around the 80's. AT&T began using Pulse Code Modulation on T1 carrier in 1959 as a method to get 24 voice channels on to one line, for long distance service.

So useful and beautifully articulated!

Thank you, I learned a lot watching this video.

I needed to hear this, thanks!

Beautifully explained.

Once you mentioned Nyquist I know you are on the right track, thanks for clearing up the air.

Hey, really nice that you speak about it, I spent some hours on that subject (and read the full Lavry paper also ^^). Just to be clear, we do agree that you need slightly more than double the maximum sampled frequency to get sampled, right ? I am aware of the way converter work with sincs, but if you would try to record a perfect 20kHz sine wave with exactly 40kHz sample rate, you would end up with constant (DC) signal, right ? so you need that extra kHz or so to get the information needed to recreate the 20 kHz ? Also, why would the anti aliasing filters be close to half the sampling theory, and not simply cut above the hearing limit ? why not for example sample at 88.2kHz, but have the filter at 21, 22 or 23 kHz to get rid of all the "sampled artifacts" ? You still get more samples for time-stretching / working with plugins that do not oversample, but without having potential intermodulation or so, I would guess ? Hope my questions were clear. Have a good day !

You are right. 44.1 is really fine.

Great explanation. Thanks.

10:25 Beautiful analogy.

From my experience, stretching audio recorded at higher sample rate, the transients are more natural. Pitch tunning also may benefit from recording at higher than 44k sample rate; i'm talking about processing after recording.

Hi Justin. Thanks so much for your channel. It’s really helpful and really accurate. I have a question. So what about these really expensive DAC’s able to reproduce 286 kHz at 32 bits, and these platforms like Tidal where you can stream Hi Resolution files? Do I understand by this videos that I’m not gonna hear any difference at all? Does it all comes down to 24 bits better then 16?

I listen to a lot of 192/24 albums on Qobuz and do a bit of Hi-Res mastering. While there isn't much musical information outside of the 20k range per-se, it is my perception, there is less "PCM sound" at at 192kHz. Filter ringing (what you referred to as resonant peaks) in the audio band inevitably interacts with the musical signal. Even 96k doesn't quite get there to my ear. It's not about resolution as much as transparency of the medium. For real-world production level work, there are a thousand more important decisions than sample rate, but for the audiophile remastering world, I welcome the 192/24 releases.

I have been recording in 24b/96kHz but I have watched this video twice (I "double sampled" it, lol) and am wondering if I should be using 24b/48kHz instead. I am thinking this is going to be the route I take, but one question: When dithering down 24b recordings to master to 16b/44.1kHz for CD... is there ANY sonic advantage in tracking the mix in 96k vs. 48k when you know its going to be dithered down to 44.1kHz? Thanks for what you do, Justin!

one main thing people should realize, if your waveform is at 44.1, it's exactly the same sinusoidal waveform as 192. You can not "add" quality by converting that to 192. like he says in the vid. you only need a peak and a trough. dac's produce the perfect waveform from those datapoints.

I have an inquiry. I was thinking, if you have a sample rate of 2 samples per second (2hz), and a bit depth of 10 bits, and let's say for a second you draw out a 1 hz sine wave at it's max amplitude on a graph that has samples mapped on the X-Axis and Bits mapped on the Y-Axis. And let's say the sample points are exactly on the peaks and troughs. You would have one sample at 10 bits and one sample at -10 bits And let's say what you draw is the signal getting sent through your DAW to the speakers. The DAW will take the original sine wave signal, and capture it as 2 points, a peak at 10 bits (at the first sample), followed immediately by a trough of -10 bits (at the second sample). Your Digital to audio converter will perfectly send those 2 signals to the speaker in sequence over a 1 second time period. 1 signal telling the speaker to move forward because of the signal point at 10 bits, the other telling the speaker to move backward because of the signal point at -10 bits. In my mind, this means that no matter what kind of wave you draw, be it a square wave, a sine wave, a triangle wave, or whatever wave the actual signal is. At a sample rate of twice the frequency, no matter what wave you draw, you will have the exact same signal being sent from the DAW to the speakers. Two points, one at 10 bits, one at -10 bits. The benefit to increasing the sample rate beyond twice the frequency of the original signal wave, is that you will be able to replicate the specific shape of the original signal wave drawn. If you had more samples in between, you could connect the dots in a fashion that looks like a sine/triangle/square wave or whatever, and have the speaker accurately represent the signal that you drew. So increasing the sample rate beyond twice the frequency would yield a more accurate wave because it would be able to more accurately differentiate which kind of wave the original signal is. Just learning this concept so I might be wrong but this is just what I was thinking. Please tell me if I am. Idk. Big fan of your stuff btw. (oh, I think I just figured out why you're right and I'm wrong, because our ears only capture audio at 20,000 hz. meaning we don't even have the ability to shape the wave with extra samples in between. If our ears are capturing a wave and turning it into data points that our brain can interpret, we wouldn't even be able to shape a wave at 20,000 hz because we wouldn't be physically able to map out more than 40,000 data points per second in our ear canal. We would take any wave at that frequency, and interpret it the same way, just as the DAW did in this example, as 2 data points to be sent to our brain to interpret as sound. I do wonder though, if even though our ears can only hear up to 20khz, if we can still differentiate between certain types of waves up until that point. You could test it out with a synthesizer in a super high sample rate session But I think 44.1khz is good enough, lol, I listen on crappy headphones anyway.)

Age doesn't correlate to lack of hearing frequencies. It would if some guy was a rocker and constantly around loud noises. My mastering engineers are 55+ and they can hear amazingly well. They've taken really good care of themselves throughout their lives and protected their hearing. It's like saying some Oreo eating 22 year old being out of shape vs Tom Cruise, and that Tom Cruise must be fat because he's older. No. It's habits. Also genetics. Higher frequencies do matter if we're talking about pitch shifting, sound design, frequency foldback etc. The filter being higher is better. One thing many people don't mention is the Gibbs Phenomenon. When a signal is digitized it picks up a vibration or ringing when truncating. It can create a ringing at step transition in the time domain. Higher frequencies lessen this effect and put the filter far outside the hearing range of humans.

THANK YOU.

I have a question about a very specific use case. Guitar Amp VSTs. Some can be quite noisy. I know improved bit depths have a reduced noise floor. Is there any benefit in going from say 48 to 96 in that regard? I assume no but am curious if I'm wrong.

Loved this thank you

Wonderful video. I'm at 96/24, much because I stretch and pitch down a lot for ambient music. Also some people won't buy my vinyls unless the resolution is higher than on the CDs. I did learn a lot here. Everything you're saying sounds right. Big thanks.

Great vid and I'm only 5 mins in. There is a limit to our hearing when it comes to certain freqs and I think that rule has ruined modern production. People scoop out the bass etc automatically in guitars and bass to make the perceived volume louder (because radio is still a thing aparently and the volume wars rage on.. Sarcasm). They do this because the human ear can't hear those sub freqs so why have them there? Producers don't even listen to the music, they scoop out the lows via a parametric eq AUTOMATICALLY on bass and guitars because of that perceived freq rule. Why do I think this is shortsighted? SPEAKERS can hear those sub freqs. Even earbuds react different to these more lopsided mixes. You can see monitors pop and flex when you let that bass fly. What does that do? Creates thump and waves to your chest and creates analog speaker distortion which can be pleasing (similar to the harmonic distortion of pushing tape being more ear syrup vs digital limitations/cracks and pops) . This modern mixing rule is my number one qualm and I think filling the gaps with white noise and perceived volume is why people switch off modern guitar music after the initial impact wears off. It's just not moorish. Bass flub should not be a dirty word IMO. Every damn mix I see on RU-vid shows people sucking out low freqs regardless of a 4 string standard or a 6 string bass. Is anyone questioning that these producers have not had guitar based hits for decades? Since (sic) DAW replaced tape? Yes I just had coffee. When it comes to sample rates I do want to go to town with cymbals and see what's what. In my experience cymbals are the best medium to really hear that digital scree when something is lossy/glitchy.

Great video!! echoes what my friend has told me about using 44.1. CD quality is 44.1 16 bit so if its good enough for the final product its good enough to do the initial recording in.

Best explanation around the internet about this misunderstood problem :) But I wonder, because you mentioned the blind test and Placebo, can this be the case of some audio interfaces? I didn't hear someone actually doing a blind test with few interfaces to hear if the very expensive ones are in fact 5 times better than the average, when in fact can be the Placebo effect also, if someone knows which interface they are listening :)

Very interesting!

This is all very profound. If we can concentrate on bringing the 16 to 20K (high bound) the oversampling that we do engage in will be more valuable in the spectrums (and fucking harmonics) that we actually interpret. Fucking science!

A lot of good points in the video, but I still keep hearing people talk about so much that has absolutely nothing to actually do with the concept of why sample rates are actually considered important (or not), which really confuses a lot of people, which is apparent by some people's comments. Understandable because the idea is a bit counter-intuitive: what's happening in those non audible ranges is potentially affecting what is in the audible range... It's to do with the processing... it has nothing really relevant to do with human/hearing ranges AT ALL, that's not why sample rates are important, it's really to do with the aliasing and artifacts being incorrectly produced in the audible spectrum that should not be being produced in AD PROCESSING in the FIRST place... then on top of that you have the potential for additional aliasing... depends what you are doing... and that's not even touching bit depth. Or dithering... yuk. In practical terms think about this: You plug a condenser mic into the preamp of your standard entry level interface, it IS receiving freqs way above 22k, these are hitting the (likely) cheap converters that are not recognizing these freqs properly at all, (or using various digital FX etc creating ultra high freqs) and instantly you've got aliasing... subtle artifacts creating very cheap digital sounding mixes... Of course this is a gross over-simplification and not always the case, but it shouldn't really be that surprising. Whilst gear is certainly not everything, Invest in a good preamp/interface ( - if you are serious, and making good music... if not... why bother reading this? ) , it really is one of the most important pieces of digital kit you have, use a grass roots-ground up philosophy, this is well known by producers young and OLD, it's not some stupid gear myth like pretty much everything else, and is worth your money. I mean, while they're not bad, do you really think a shitty 200 dollar multichannel interface can even remotely compare to a state-of-the-art preamp and AD converters? No... of course not... but that's a whole other can of ear worms! Do you need it? Of course not!... Does it still sound good? yeah sure... kinda... but does it sound like good ol' rock n roll? Fucking hell no! My 2 cents is basically this: ultimately it's not what's important and the MUSIC is always #1, but it does matter... kind of... just because most people don't care doesnt mean we should just lower the standards of the Earth... (laughs) that's exactly what's wrong with the world! But I digress... A good song is a good song! If the music's good.... it's good! Always focus on that.... HOWEVER... It is unfortunately an unavoidable aspect of the digital era and medium, and worth understanding properly as an engineer/producer or person who wants to take any sort of professional level recording seriously... Personally i tend to use 88.2k, - because i'm exporting to 44.1k... the freq cutoff of 88.2k is not so close as to create excess audible artifacts, and seems to make a noticeable difference, it's also perfectly double to the out rate of 44.1 so you don't run into other problems (or it used to be, 48k is actually the common standard now because of the advent of dvds and digital media, also a lot of modern music hardware runs at 48k, if that is you're target out rate then choose 96k). I don't advocate higher than those, more is not better, but hitting the right mark based on your output sample rate IS. You won't harm things by down sampling to 44.1 or 48 from 88.2 or 96k respectively, the damage was either done in the first place or not by that point. Processing power shouldn't be an issue, you've got nothing to lose. BETTER STILL... Do some proper tests for yourself at different rates at least with what you've got, it really depends on what you are doing, you can certainly notice a difference, without a doubt, but again it all depends on what you're doing and what you're dealing with. That's a lot of rambling, but It's very late here and im also a bit drunk. Basically, IMO choose either 88.2 or 96k depending on whether you are exporting to 44.1 or 48k... (double) think about it... Peace.

Does recording (or just rendering) at higher rates make a difference with fx like delay and reverb?

Love it🎼💯

I’m not sure where I heard it but I have in my head that because video usually has its audio at 48k, it’s better have your music at 48k to avoid any additional conversion if it gets used in a music video or commersial, etc, and that multiples (96, 192) will convert cleaner than non-multiples (44.1, 88.2, 176.4). Any truth to that?

Thanks for another great video! Any chance you can dedicate some time discussing whether you’d mix and/or master 100% in the box or go with a hybrid 2-Bus setup (including Comp/EQ/Color) that can also be used for single track/bus processing or maybe even a hybrid setup that starts off with analog summing then to Comp/EQ/Color? I’m less interested in the controversial summing piece, but thought I’d include the option for completeness. All this assumes great hardware, great converters, a treated studio, and quality monitoring. The debate would be whether there is enough of a tangible benefit for a hybrid 2-Bus setup (Comp/EQ/Color) to justify a higher noise floor. Your thoughts??

Thankyou God bless!

In Reaper when I upsample with the option "Use project sample rate for mixing.... " unchecked, my 48khz mix will be processed at 96khz and saved to a 96khz file. The processing is different , the aliasing is different, and even my reverbs sound a bit more 3D! So why people say there is no difference?

Hello Justin, thank you for all infos, really liked it! I have been testing this and I found that the 96k works better for me, simply because if you use any plugin that creates harmonics close to the barrier of 20k, in 44, the aliasing effect will reflect that harmonic back to the listenable frequency range. Even when you're using sounds that goes over 20k (like virtual synths, without the antialiasing) starts to reflect back, sometimes to 600hz. In 96k aliasing exists as well, I tested with some plugins and I could find some frequencies coming back to the audible range, but it was an extreme case. What do you think about that? For instance, CLA compressor creates harmonics to try to emulate the hardware, if you compress cymbals that goes to that airy highs you could get some aliasing right? Is there anything, besides the oversampling in some plugins, that we could prevent this to happen in 44? thank you best

I hear that my studio monithors sound much better in low frequencies and highs are better to listen with higher sample rate on soundcard

I trust my ears, work, apply plugins and specially use external peripherals at 96k or better will make a drastic difference on the final mix. Try it! Even if you have to deliver a final 44.1k mix. Logic X has the tools for that. I'm not alone, research more and will see the importance of higher sample rate. More important, when you pass your audio through an external hardware at 96 maintain the track intact, without any plugin. Summing with AD tracks will result in a better mix than DD tracks.

What if you take a 96khz file and and cut out the 0 - 20~ khz, then play it on a system with speakers that claim to produce up to 50khz? That way you'll only play 20 - 48~ khz.

I have an apollo x16 and I mix a 88.2 and I can't say it sounds better...but to me, I can hear what I'm doing much better as far as eqing & compression etc etc.

I swear that I can her a different tamber of the whole track at different bit rate

I was more interested in if 48000 is better than 44.1k because it's more divisible by 2 or because of some other conventional reason that keeps you from having to split a waveform in half or something. We already knew it didn't make a difference in audio quality before this 39 minute video

great video thanks so much! But sample rates help with time stretching, warping, and pitch shifting, no? retains the quality more.

Tchad Blake is 44.1k and I know Serban Ghenea WILL downsample to 48k/44k WHATEVER sample rate your recorded in. perspective :)

24/48, FTW (now and into the foreseeable tech future; for just music AND film purposes) ………. ?!?!?! 😬 I remember some people saying they used 88.2 (over 48 or 96), because it was exactly double 44.1, so when they exported at 16/44.1, ‘rounding errors’ that might/would otherwise occur when exporting from other sample rates were avoided (they may well have recorded in 32-bit, as well, rather than 24 … ‘because it’s easier for a computer to just halve data, rather than multiply by 2/3’ … 🙄😏 … and/or ‘because it somehow avoids rounding errors’). … No doubt, they were concerned about quantizing … but, given how good computers are at math, that sounded like nonsense then, and it still does now … ?!

Which sample rate should I choose to streaming with obs ? 44.1k or 48 k ?

Does bit depth make a huge difference? Basically as a beginner in this, I want to know the best to use for general all genres and styles of music in a home studio.

what if you do sound design, you might get extra fidelity when you pitch down the audio a few octaves?

One issue you did not cover has to do with the difference of sample rates as they pertain to virtual instruments. Some virtual instrument developers use the 44.1 kHz sampling rate in their delivered products, while other developers use the 48 kHz sampling rate. This poses a problem for those of us who are media composers. Doing projects with VI's that use the 44.1 kHz while having to deliver finished cues (or stems) at 48 kHz requires sample rate conversion. This, to my ears, causes a degradation in the sound quality of the resulting music. Also, some projects have a requirement to deliver the music at 96 kHz. It is simpler to just double a project which used VI's at 48 kHz rather than having to employ sample rate conversion because of a VI at 44.1 kHz. No client has asked me to deliver a project at 44.1 kHz or 82.2 kHz. Therefore, I have made the decision to use only virtual instruments which are delivered at 48 kHz. Any thoughts on this aspect of sampling rates?

Yes, higher sample rates equal lower latency, and less ailiasing in the audible range. One must manage imd however above 44.1k sample rates.

But lots study shows us that ultrasound also can impact on visualising the sound in different way...In nature, we don't hear sound as pure sine wave, we take it as a harmonics ... lots of musical instruments can produce ultrasonic sound that can give us different kind of musical test(may we can't hear ultrasonic as a sine wave , but in harmonics , it might give different colorization to the preceiving sound)..

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Can you make a video on mqa music ? Is it bullshit like hi-res audio

This is great, i've always thought all this. Science b!7£h! The snake oil that we are fed as producers is surely the highest in any industry other than moisturising creams and make up!

I've never heard anyone make a simple mathematical calculation so damned complex and black-magic sounding. The 44.1 standard was created assuming an insanely steep high-cut filter to eliminate the harmonics. Like 18+ db per octave. You can't have one without the other. They assumed the current-day economics of the 1970's and the least expensive way to meet the minimum spec of reproduction, so 44,100. We now have 192 khz all day long, and the hardware filters to match. Is it better? Well, I record at 96k, but only when high freq is present, and I'm pretty picky with my mix. Could I record at 44.1 or 48k? Yup, but at 48k, I *might* detect some issues with harmonics, and the integer conversion from 48k to 44.1 is problematic in that case. If I know there are no cymbals or high-freq percussion, then I just track at 48K to save some space. I prefer to just side-step all the edge-conditions and use 96k when transients are expected (yeah, my SSD hates me). If we assume 30 kHz to include all the 2nd order harmonics from a cymbal emanating 15K, then 60 k samples/S would be enough, using Nyquist's math. I use 96K, to avoid any "rounding errors" found in real life, because 48K is a wee-bit shy of my personal comfort zone for harmonic overhead. Once mixed, then the output sampled at 44.1 kHz is fine, but inside the mix, I prefer to track at 96k for high freqs.

Finally the full truth about sample rate. And given to us so clearly. Thank you a lot! Can you make a video for "BitDepth" as well?

It’s funny but I just turned 40 and can still hear up to 20k in my left ear and as high as 23k in my right. True story. Had my hearing tested a couple of years ago and the tester was very shocked!!! It may be down ever so slightly - but consider my a rarity!

A little background, 48k comes from the time with DAT recorders, and it is to this day relevant in studio's. I worked in a studio, recording masters to DAT, very annoying work. I did it one year and threw the towel. Enough was enough, 75% were rubbish songs. OMG.

Is this man saying that a squarewave and a sinewave is the same thing? kinda getting that vibe 13:00

Sonicscoop . What sample rate do you use?

@30:10 I actually make music for bats -- Batronica Dance Music

Hi Justin, are you still using these Focal monitors?

ok 😁 thaynx bro💟

Make a video on dsd and pcm